I have already subscribed to the alerts but updates don't mean much when the proposed time frames keep getting pushed out � your 3 to 5 day time frame for resolution should not now be pushing 18 days.
You might like to consider also including an "unsubscribe" option for anyone wishing to opt out at a later stage.
Every email will have an unsubscribe link
Hi John, i need your help
I'm trying to setup a TP-LINK VG3631
The settings I'm being asked for are
Profile Name: Telecube
Phone Number: 1029XXX
Display Name: 1029XXXX
Authentication Realm: sip.telecube.com.au
Authentication ID: 1029XXXX
Password: XXXXXX
Registrar Address: sip.telecube.com.au
Registrar Port: 5060
SIP Proxy: sip.telecube.com.au
SIP Proxy Port: 5060
Outbound Proxy: sip.telecube.com.au
Outbound Proxy Port: 5060
Register Via Outbound Proxy: Enabled
Are these settings correct? Because it is not registering
ps. In the advanced settings of the router, register expiry timeout values go from 300-3600
I can't specify a timeout of 180, is that a problem
This is more of an FYI for all: [#349-443-800]
Over the last week or so I have been unable to make any calls to Telecube but I was able to receive calls with no problems. I notice the problems first occurred around the time the DDoS attacks started so I didn't worry much about it and waited for everything to settle, thinking that was my problem.
I was receiving:
Got SIP response 603 "Decline" back from 103.193.167.162:5060
I have Vanilla Asterisk 13.9.1 and to finally get it working I had to add the fromuser. Trying to remove defaultuser and just have fromuser did not work and as soon as I removed fromuser the SIP 603 reoccurred.
[Telecube](ulaw)
host=sip.telecube.net.au
fromuser=102XXXX
defaultuser=102XXXX
secret=somepassword
qualify=yes
type=peer
Authentication Realm: sip.telecube.com.au
This is sent to your ata with the request to authenticate so leave it blank or try telecube.com.au
Registrar Address: sip.telecube.com.au
Try sip.vic.001.telecube.com.au and same for sip proxy
Outbound Proxy: sip.telecube.com.au
Outbound Proxy Port: 5060
Register Via Outbound Proxy: Enabled
Should not be needed
In the advanced settings of the router, register expiry timeout values go from 300-3600
TC likes anything between 120 -600 if you can.
Hi Robnll
Thanks for the settings
I've tried sip.vic.001.telecube.com.au as the registrar/proxy address
and I've also tried telecube.com.au as the realm and leaving it blank
Expiry is set to 600
Registration status is still: DOWN
to finally get it working I had to add the fromuser.
Asterisk 12.1.1 (Vanilla) uses fromuser in the From and Contact hdgs and defaultuser in Digest Username when authenticating and this works with all my vsps.
[Telecube](ulaw)
Just out of interest is (ulaw) a vsp template
secret=
I'm sorry .. I found your extension and deactivated it. You shouldn't post your passwords here.
Registration status is still: DOWN
Are you sure of your password . You could always get a new one and try that. I dont have your model ata but I use asterisk and a gigaset and these details work with them. Can you disable sip alg in your router. it shouldnt matter because I have sip alg enabled and TC still works ok.
Are these settings correct? Because it is not registering
Is there a username field in there too somewhere?
Just out of interest is (ulaw) a vsp template
It is a template not specifically for a VSP but for anything. It contains:
disallow=all
allow=ulaw
I usually have the following in my installs and just reference the templates that I need. This limits the lines for friends, users and peers. More so for friends i.e. extensions.
[exten-set](!)
type=friend
qu
port=5060
callgrou
pickupgroup=1
host=dynam
dtmfmode=rfc2833
canreinvite=no
context=internal
[allow-lan](!)
deny=0.0.0.0/0
permit=192.168.1.0/255.255.255.0
[g729](!)
disallow=all
allow=g729
[ulaw](!)
disallow=all
allow=ulaw
[g722](!)
disallow=all
allow=g722
A typical extensions for me looks like this:
[1234](exten-set,g722,allow-lan)
defaultuser=1234
secret=somepassword
mailbox=1234@default
callerid=Person 1 <1234>
description=Person1 Phone Polycom 560
Are you sure of your password . You could always get a new one and try that. I dont have your model ata but I use asterisk and a gigaset and these details work with them. Can you disable sip alg in your router. it shouldnt matter because I have sip alg enabled and TC still works ok.
I've tried resetting the password twice
I have different extensions registered on a gigabit, a netcomm vp200 and a os x soft phone and they all work
Is there a username field in there too somewhere?
No filed labeled username, I assume that would be (Authentication ID: 1029XXXX)?
ps. there is a interface emulator for this router
http://www.tp-link.com.au/re
If you would like to look through the voice section of the router
I usually have the following in my installs
Many thanks for all the info. It is one of your raspi installs I started with and am gradually adding features whenever I get a chance. A valuable learning experience.
Here are the settings that finally worked for the TP-LINK VG3631
Password resets didn't work, however de-activating the extension and re-activating generated a password that worked
Profile Name: Telecube
Phone Number: EXTENSION
Display Name: BLANK
Authentication Realm: BLANK
Authentication ID: EXTENSION
Password: XXXXXX
Registrar Address: sip.telecube.com.au
Registrar Port: 5060
SIP Proxy: sip.telecube.com.au
SIP Proxy Port: 5060
Outbound Proxy: BLANK
Outbound Proxy Port: 5060
Register Via Outbound Proxy: DISABLED
Phone Number: EXTENSION
Just looked at simulator and this is most likely what others call username. Trouble is there are plenty of standards for sip but no standard terminology for ata field hdgs.
Anyway good to hear it is ok now.
"Just wondering if anybody still has issues with DTMF tones when using Telecube?"
Hi jdd � I have a Frtiz 7390 and the DTMF should be set to inband to recognze keypad response. I have had no problems with telecube and several teleconference providers (but admit I have not used AT&T).
As suggested earlier also try the different codec settings...
ALI G... re your TP-Link 3631 router issues.... I have a carton fresh TP-Link 3531 ( basically the same ) and have tried all setting that you have mentioned, and like you, no luck... new passwords, Alternate sip addressing, the lot....
The router will connect and works with two other service providers instantly, so I'm sure it's not the router...
I have reported (two days ago) a possible system fault as well as our mutual router problems to TC in an email.... we may be getting locked out via a simple system fault at TC's end.... If / WHEN an answer for our router issues is forthcoming, I will whim you the info... I won't highlight the (possible) system fault publicly, after recent issues...
Cheers,
Glenn
Hi mate
Did you see the settings that worked for me a couple of posts back?
Set your register expiry to 600 as robnll suggested in advanced settings
If you have access to a soft phone on your computer try to authenticate the login details to see if they work before trying them on the modem
Was interstate this week and my partner called me to say that the Telecube VOIP is down.
Just got back home and for some reason my Gigaset has Telecube as "registration failed"
Has anything changed recently?
What do I need to do to get it back up?
What do I need to do to get it back up?
Power cycle your modem/router.
Restart the Gigaset base.
Edit. Already answered.
eh, before you guys got to me, I deactivated the connection and reactivated with new password.
Then I power cycled but unfortunately it is still down.
I will contact Telecube tomorrow and hopefully it can be fixed.
I probably messed up by deactivating too. I should have been patient.
Asssuming you can access the internet by posting here.
Can you open a command prompt and "ping sip.telecube.com.au"
Hi Ali G, robnll and others who contributed to my TP-LINK 3511 router problem solving.... the combined suggestions have at least got me registered, which is MUCH further than before :-) ....so many thanks for the help and pointers, it's appreciated.
Too late to test call quality tonight without risking sleeping outside, so that will wait for tomorrow... along with DID testing...
Cheers
Glenn
eh, before you guys got to me, I deactivated the connection and reactivated with new password.
if you're still on sip.telecube.net.au try sip.telecube.com.au
no idea but i thought had some set to 'net' 2 wks ago and lately they've gone 'com' afaik
Hi Ali G, robnll and others who contributed to my TP-LINK 3511 router problem solving....
Great stuff glad you got it working
Since the last problem with Telecube, when I get an incoming call on my primary DID (07412XXXXX) the call disconnects after only a few seconds. When the caller calls back the call proceeds normally.
I will do some more testing today from my mobile and another VSP to see if I can get more handle on the problem.
The device is a Siemens A580IP with A58H handsets.
Since the last problem with Telecube, when I get an incoming call on my primary DID (07412XXXXX) the call disconnects after only a few seconds. When the caller calls back the call proceeds normally.
Sounds familiar. I posted about that yesterday.
But I am going on what my colleagues tell me, i havent seen it myself.
Is anyone having issues registering on telecube this morning? Rebooted my Gigaset c530AIP and still not registering
Asssuming you can access the internet by posting here.
Can you open a command prompt and "ping sip.telecube.com.au"
Yes I can and these are the results I got:
Microsoft Windows [Version 6.1.7601]
Copyright � 2009 Microsoft Corporation. All rights reserved.
C:\Users\Andrea>ping sip.telecube.com.au
Pinging sip.telecube.com.au [103.193.167.161] with 32 bytes of data:
Reply from 103.193.167.161: bytes=32 time=12ms TTL=55
Reply from 103.193.167.161: bytes=32 time=11ms TTL=55
Reply from 103.193.167.161: bytes=32 time=11ms TTL=55
Reply from 103.193.167.161: bytes=32 time=13ms TTL=55
Ping statistics for 103.193.167.161:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 11ms, Maximum = 13ms, Average = 11ms
C:\Users\Andrea>
if you're still on sip.telecube.net.au try sip.telecube.com.au
no idea but i thought had some set to 'net' 2 wks ago and lately they've gone 'com' afaik
I just took a look in Advance Settings, and everything is set to sip.telecube.net.au
Please subscribe to the status alerts, we will have an update tonight.
Did anyone else who subscribed to the alerts not receive anything overnight?
Please subscribe to the status alerts, we will have an update tonight.
still waiting on an update....
Sorry guys .. getting there .. at this stage Optus have committed to a transfer date of 20th June. We're pushing for an earlier time but it's not looking like they will come to the party.
Full details will be in a notice going out shortly.
Is anyone having issues registering on telecube this morning? Rebooted my Gigaset c530AIP and still not registering
Yes. me too. I moved my phone over to another provider yesterday at lunchtime. Just tried to change back to Telecube but won't register. So back to World Dial Point for a while. I don't know what's going on.
My issue got resolved. Pretty much had a wrong Domain address due to recent changes.
All fixed with a simple phone call.
Hasn't been able to register for over a week.
You need to be a customer first, before you can register.
You can signup here. https://www.telecube.com.au/signup/?fr=home
You need to be a customer first, before you can register.
I am a customer, and I used Telecube successfully until they started going downhill.
Same story for other people I know (and it irritates me because I recommended Telecube to them originally).
You can signup here. https://www.telecube.com.au/signup/?fr=home
Can't recommend them these days.
I am a customer, and I used Telecube successfully until they started going downhill.
Amazing.
You have been posting in several other companies VOIP threads all month.
Not one post in a Telecube thread ever until today, yet you have had total loss of service for a week.
Apparently.
I'm expecting to get the first alert out this evening or over the weekend .. the new status system is taking subscribers but can't actually send messages just yet .. apologies for the delay.
A major criticism by many through all this has been that we haven't communicated the issues to the end users and I agree completely. At the moment it's a fine balance between resolving the remaining customer issues, focusing on the overall platform and announcing the current state of play.
Once the status alert system is functional I will be able to easily post messages to subscribers and will do so.
Please subscribe at http://telecube.io
You have been posting in several other companies VOIP threads all month.
Only because MNF are migrating Pennytel customers to MNF's platform.
And surprise surprise, I have accounts with more than one VOIP provider. That way I have back-up just in case.
And surprise surprise, I have accounts with more than one VOIP provider.
And so do I.
Can I suggest that emailing alerts is a little old school. Instead most of my service providers sent alert through a twitter feed designed solely for this purpose.. Hopefully you can do the same.
Can I suggest that emailing alerts is a little old school. Instead most of my service providers sent alert through a twitter feed designed solely for this purpose.. Hopefully you can do the same.
Wouldn't the email alert be quicker?
One less step to follow not having to click through the twitter email and find the announcement.
Wouldn't the email alert be quicker?
Email or sms suits me. I have nothing to do with Twitter, Facebook etc and don't intend to.
Email or sms suits me. I have nothing to do with Twitter, Facebook etc and don't intend to.
+1
Email or sms suits me. I have nothing to do with Twitter, Facebook etc and don't intend to.
+1 exactly my sentiments.
(BTW I have signed up to http://telecube.io)
A benefit too is that I can integrate monitors and display statuses on network segments on the site as well.
Email or sms suits me. I have nothing to do with Twitter, Facebook etc and don't intend to.
I'm with Imn on this one. G20.
John actually no clicks required, I see critical feeds on my home screen and it works very well. I actually only use twitter for this reason :-)
Hi John,
I know things have been unpleasant for a month or so for TC and its customers. I also know that you've been working tirelessly to clean up outstanding issues. I feel that you working too much is showing up as undesirable little issues in some areas. I am sure you'd have handled things differently if you had the choice.
For me, our business account has been experiencing missed calls and unexpected call termination among other strange behaviours, all without changing anything on our FusionPBX server. I could easily troubleshoot the issue if calls were indeed hitting my server and then failing to proceed but in this instance I had nothing to work with. I think there is an issue somewhere in your new OpenSIPS proxies handling remote NAT and registration, obviously I am speculating here.....
I think it would be better to have an alert page on the website in addition to the email alerts.
If you need a hand for the short term with some menial work while you are crushing those bugs, you know how to get in touch :)
General Announcement
I can understand peoples frustration and need to lash out at me and my company and I accept it.
I hope you can understand that I'm up to my ears in this at the moment and have been for close to a month now, my staff and I are doing the best we can in a tough situation.
I'm not hiding or running away or trying to mislead anyone, I'm committed to bringing us back to the level of service that you've come to expect over the last several years.
The last 6 months has been troublesome as loads have slowly increased on the old platform and the forced switch to the new platform in the last few weeks has seen some system bugs that are affecting services on a small number of users.
We're working through them individually with each customer so if you are still having service issues please email support so we can assist you.
I sincerely apologise to those affected and thank you for your patience.
Wouldn't the email alert be quicker?
I am over 16yo. Email is good.
I think everyone has an email address these days. so you can reach everyone.
I think everyone has an email address these days. so you can reach everyone.
Also having an email address is a requirement to be a Telecube customer and we send system and accounting alerts that way so it makes sense to send network status alerts by email too.
Email or sms suits me. I have nothing to do with Twitter, Facebook etc and don't intend to.
Me neither. Even though my grandson signed my up to a facebook account and now all I get are almost daily E-mails from facebook with names of people I supposedly should know.
Email or sms suits me. I have nothing to do with Twitter, Facebook etc and don't intend to.
+ 1 . Ditto
that way so it makes sense to send network status alerts by email too.
And there will still be those that will get the E-mail and still come here complaining something isn't working and expect a resolution time, all before you have even identified what the issue is.
I hope you can understand that I'm up to my ears in this at the moment and have been for close to a month now, my staff and I are doing the best we can in a tough situation.
A little frustration showing here. Fact is you can't all things to everyone or even solve everyone's problems instantly all of the time. I'd suggest you concentrate on what you where and what your platform wants to achieve. Forget about the "I want" brigade. In the end no matter what you do there will always those that want more.
I think it is pretty clear cut that there are some out there reselling your services, probably at much more profit than yourself. The contribute very little, except whinging. No one forces them to use TC, but when all is well they have no worries and more than likely make a handsome return.
These are odd ones. As Robnll has pointed out, the best thing is *usually* to send DTMF from you to the VSP as RFC 2833 � which I assume you have been doing !!??.
I currently have RFC 2833 configured on my VOIP Router (Billion BiPAC 7800VDOX).
But that's not working to AT&T so it might be worth breaking with convention and trying "in band" signalling rather that RFC 2833.
To try this log into your ATA/IP phone and wherever you have RFC 2833 selected, re-select "in-band" or "audio" or similar name for DTMF signalling.
Thanks, I did try this also with no success.
You should at the same time select only the G.711a codec (sometimes called PCMA or A-law) and have that as the *only* codec selected so that is has to be used.
Ok, tried this with no success.
You can also try the US G.711u (Greek mu, sometimes called PCMU or U-law). Again only the G.711u codec selected to force it.
Finally I have had Success with this one! You are lifesaver Ned!
I can now dial the AT&T conferencing system and it works perfectly again responding to Key Code Dial tones.
Thanks for taking the time to give so many valuable options to try!
My solution has been to remove G.711a from the TC codec order so it's never used.
Thanks JDD, This same advice worked for me as well Switching to just using the G.711u codec has got my dial tones working properly again.
I have cloudsoft on my android phone but over 4G when I call a landline they can hear me but I can't hear them.
Over wifi it works ok both ways . Any ideas.
Even though my grandson signed my up to a facebook account
luckily my youngest grandson is only 2........ but maybe next year???
H.
Over wifi it works ok both ways . Any ideas.
I had similar issues when overseas last year.
Csipsimple worked no problems so I switched to that, despite no g729. (On Android 5.0)
Csipsimple worked no problems so I switched to that
same problem with csipsimple
I have a quick question. I have a VoIP service with another provider at the moment for my mother who makes occasional international phone calls. Will i be able to prepay service with Telecube ?
Hi H?LIO?, I may be wrong but the fact that you are asking suggests to me that you may have a less than perfect understanding about VoIP which is something we can help you with. It would be helpful to know why you are asking and what you are trying to achieve.
The answer is probably, "yes", but if your VoIP equipment is locked it will need to be replaced if it cannot be unlocked. Your equipment will need to be able to support 2 providers if you wish to occasionally use the other provider or if you have a DID with the other provider that you don't wish to port. You don't need to have a DID with a provider to make calls, but you do if you wish to receive calls from the PSTN.
For a Whirlpool discount, click on the relevant offer link in the Telecube Offerthread.
No I am talking about the Optus router failure
That was suppose to be an inbound DID problem.
What happens when you ring your DID?
Switching to just using the G.711u codec has got my dial tones working properly again.
The USA primarily uses ulaw, while Europe (and Australia) primarily use alaw. Though many Australian businesses buy US equipment and leave the codecs set to the default (ulaw).
So since you're connecting to a US based system it makes sense to use ulaw.
I wonder if the DTMF RTP events are being stripped out when transcoding happens. That shouldn't happen. Anyway, you typically need PBX functionality to vary codec for specific destinations.
BTW, the last time I tested Telecube balance and/or echo with a range of codecs they gave null audio on some of the codecs they agreed to. I suppose, I better retest and submit a ticket :-).
Can anyone give me an update, are queues working? I changed a number of extensions using Cisco SPA508g handsets to sip.telecube.com.au and can't get them to work on the ring group. They register and can make outgoing calls. Tried deleting the queue and added new queue with just one extension, still not working. Deactivated the extension and set it up again, restarted modem, still not working. Calls just go straight to voicemail, phone does not ring Running out of ideas.
Email or sms suits me. I have nothing to do with Twitter, Facebook etc and don't intend to.
You don't have to sign up to Twitter to make use of it.
For example, https://twitter.com/BOM_Vic
Anyone can click on that link and see tweets by BoM Vic � you don't need to have an account on Twitter.
The advantage of using something like a dedicated website, blog, or Twitter is you aren't bothered unless you think you may have a problem.
I don't know what proportion of outages affect all customers but I know it isn't 100%. For example, if you don't have a DID, you don't need emails about DID outages.
It is possible to have both email and Twitter once John has some spare time. In fact anyone could set this up.
e.g. set up an email address that automatically posts to Twitter via IFTTT, or Tweetymail.
Or a blog set up with email to posting (e.g. blogger.com supports email posts).
I don't know what proportion of outages affect all customers but I know it isn't 100%. For example, if you don't have a DID, you don't need emails about DID outages.
Once the alerts is functional I'll also add an option that will allow you to subscribe to specific service type groups.
Just received an email. Are you saying some DIDs have been down for weeks?
Are you saying some DIDs have been down for weeks?
Yes, some have.
The error message is played twice. Looks like an email is generated each time it's played.
The error message is played twice.
Hehe... and here I was just about to call you a liar :)
I made a few test calls to my DIDs and the message was only played once � so only one email generated.
With test calls from my mobile, MaxoTel and TC accounts, the message was only played once before the call was terminated. But with a test call from my Faktortel account, the message was played twice � generating two emails.
So it seems that the number of times the message is played depends on where the incoming call is from.
Something to note in regards to the Caller ID displayed in the generated email notification is that the CID may not be the correct CID if the caller has barred sending CID.
Having made a few test calls from various accounts, I found that with calls from my mobile and Faktortel account which are enabled for sending CID, the number displayed in the emails is correct.
But test calls from my MaxoTel and TC accounts ( which have sending CID disabled) display a generic number which I suspect is associated with the VSPs underlying carrier service � e.g:
Calls from my TC account to my TC DID (088427xxxx) show as a call received from 087123xxxx.
Calls from my MaxoTel account (which has an associated DID of 087123xxxx) show as a call received from 028203xxxx.
Note � The two 087123xxxx numbers are totally different numbers and just happens to be a coincidence that the prefix is the same albeit with different VSPs.
[Edit] � Corrected DID typo.
Calls from my TC account to my TC DID (08427xxxx)
In reality you don't make calls from your DID. With any PBX you make calls from an extension and the call goes out on a trunk. The trunk is not is not really associated with the number you refer to as a DID.
The VoIP industry has bastardised the term DID in such a way that people misrepresent what it actually stands for. DID stands for Direct In Dial. It is a process that determines where a number is terminated when it is received by the P(A)BX.
When you choose to disable the outgoing call CID associated with your extension, anything could happen in regards to what caller ID is presented to the terminating call. This will depend entirely on how the trunks are setup with the respective carrier as well as how P(A)BX is programmed. hence the reason why you are seeing the numbers displayed.
IMO the VoIP industry should ditch the DID term from their vocabulary as it only serves to confuse end users. Call these numbers public or geographic or even ACMA numbers and be done with it.
should ditch the DID term from their vocabulary
Well it surely did me when I plunged headfirst into this new game a few years ago.
howdy folks! Slight odd issue I can't find resolution to...
Been trying some Panasonic handsets, they've got screens to show CLID... when phone not used, they should show the time.
Supposedly there's no way of manually setting this time, it automatically gets it off the network. Nothing to force refresh afaik.
Since plugging in to TC... the time did appear (I was amazed it was correct time not 12:00 12:00 12:00 haha) a couple times in previous weeks but lately it's been no-show.
Just curios where/when this data gets fed to the phone... do I need to make a call to that handset before it reads? Or is it meant to poll time server occasionally in the background.
Just curios where/when this data gets fed to the phone.
I presume it is the same as my unidens. They get the time from the ata they are connected to and the ata gets it from the gateway router which gets it from the network.
For analog phones, the time is sent with the CLID each time there is a call. iP Phones usually use NTP, so will update more quickly.
So if using an ATA, it needs to have the correct time as Robnil says.
I'm getting registration failures today �
SIP/2.0 503 Service Unavailable
I'm getting registration failures today �
SIP/2.0 503 Service Unavailable
If you subscribe to the Telecube Status Alerts feature, you would have received the following email earlier today:
Status: New
Event: Database maintenance
Affected Service Group: Non Service Affecting � General Announcements
Impact Severity: Non Service Affecting
--------------------------------------
I'm making some changes to the databases that will affect the registration status display in the portal for a few hours this afternoon.
Inbound services won't be affected but there is a very small chance some registrations may fluctuate for a few minutes during the updates.
:)
where/when this data gets fed to the phone...
On my Gigaset 610IP I have an option to synchronise the time with that provided by the VoIP provider, and it works just fine. My F!B 7390 also regularly updates itself with the Internode server... (I just hope they don't get out of step LOL)
If you subscribe to the Telecube Status Alerts feature
Thanks, but I don't see how this notice is relevant to a no SIP registration for a few days fault.
I'm getting registration failures today �
Thanks, but I don't see how this notice is relevant to a no SIP registration for a few days fault.
Your first comment only said today, his answer was totally relevant for the information given.
I'm getting registration failures today �
SIP/2.0 503 Service Unavailable
You'll get this response if the username you're sending isn't valid.
If you are registering with a server make sure you are setting username and fromuser
Calls from my TC account to my TC DID (088427xxxx)
In reality you don't make calls from your DID.
Sorry but I dont get your point as TB1 clearly stated the call was to my TC DID (088427xxxx)
John � the email status notifications are working very well. It's great to be informed of your progress and warned of any possible impending issues. They are nice and concise too. Good job!
John � the email status notifications are working very well. It's great to be informed of your progress and warned of any possible impending issues. They are nice and concise too. Good job!
Ditto.
I also appreciate to see codecs in use and other information under Call Type.
You'll get this response if the username you're sending isn't valid
Thanks. Nothing changed here, but knowing this I disabled then re-enabled the extension on the portal, l got a new sip password and it starting working again.
If you are registering with a server make sure you are setting username and fromuser
Does this page need updating? Telecube VoIP
Does this page need updating?
It's a peer reviewed document. Go ahead and make the changes.
For those not following the email notifications, have just received notice that: "The extension statuses in the portal are displaying the correct state again."
I've just checked and it's all looking good (well, for me anyway!)
Does this page need updating? Telecube VoIP
I've added the status alert website link. What else where you thinking of?
Mine has been online, calls made and receive without issues.
Short registrations @ sip.telecube.com.au sip.vic.001.telecube.com.au but the portal seems a little sluggish today though.
Im getting drop outs after a few seconds and unable to answer incoming calls
I am failing to register an extension
We are getting drop outs on calls still today. Anyone else getting this?
I am failing to register an extension
Im getting drop outs after a few seconds and unable to answer incoming calls
We are getting drop outs on calls still today. Anyone else getting this?
Can you email details to support please so we can look into it?
Hi John, I have been having some difficulties regarding incoming calls for sometime now (ever since the outage started.) Basically, with all my incoming calls, they would automatically cut itself off after exactly 60 minutes. I have used several different PBX systems, they all have the same problem. In fact, it happens to my fritzbox as well.
I understand that you do not set a limit on the duration of any incoming calls (Which I can testify to for nearly a year, until the outage started.) I notice a number of posts here, indicating the same problem. I have contacted support over the phone a few times (#312-191-669), but have not got any response.I wonder if you can look into this for me, because I rely on having the conferencing function-and all of the meetings are over one hour.many thanks
We are getting drop outs on calls still today. Anyone else getting this?
As I mentioned a few times in this thread, this is our experience every day for at least a couple of weeks
Incoming Call drops within a few seconds of the call, if they ring back straight away its always fine.
Can you email details to support please so we can look into it?
With all due respect, I was informed by your staff that this cant be looked into?
A couple of months ago we were experienced some call dropouts and I asked your staff to look at the logs to find out what was the termination reason for the call, and I was told that you dont keep these logs.
Basically, with all my incoming calls, they would automatically cut itself off after exactly 60 minutes.
We had incoming calls dropped at 15 minutes 32 seconds since the outage. Our fix was to add "session-timers=refuse" to general section of sip.conf. Give that a try.
[edit]: we use Elastix
I need to know the services and configuration that this is happening on please, if you send me details I can look into it.
I understand there is some frustration at the delays in responding and I'm sorry but we are catching up now and if you give me specific details I'll be able to sort this bug out.
Thanks for that. Just PMed you.
Thanks for that. I'm trying this out now, and will be able to tell you in an hour's time. Cheers...
My account is active and online but I cant dial anything. Phone says "declined"
Whats going on. I haven't used it since the outage
Whats going on
Without specific information about your account there's no way to know .. please email support with your details
Basically, with all my incoming calls, they would automatically cut itself off after exactly 60 minutes. I have used several different PBX systems, they all have the same problem. In fact, it happens to my fritzbox as well.
Sounds like an issue with the NAT timeouts on your router. Especially with the introduction of proxy IP addresses by Telecube recently there is a greater chance that the NAT feature on your router is expiring the UDP sessions for SIP and/or RTP after 3600 seconds. If you can't see what the settings are on your router or can't change them then your solution may be to get a better router that is more configurable or implements better SIP inspection for NAT.
Sounds like an issue with the NAT timeouts on your router. Especially with the introduction of proxy IP addresses by Telecube recently there is a greater chance that the NAT feature on your router is expiring the UDP sessions for SIP and/or RTP after 3600 seconds.
Oh dear.... Well, it still dropped off after an hour... So, what if I increase that to 7200seconds...?
Can you email details to support please so we can look into it?
Hi John, I've done this. The ticket is #658-277-805
Still waiting on a response for #830-511-708
Without specific information
You may have to write a FAQ on how to ask for support.
Things like...
Origin of call
Destination of call.
DID to extension configuration.
Equipment (PBX or phone or ATA)
Internet connection type.
etc.
Are extensions still failing to register.
My account for my Linksys Pap2 is getting error I get is Registration State: Can't connect to login server
Sip servers tried:
sip.vic.001.telecube.com.au
sip.telecube.net.au
Have checked Pap2 settings and deactivated extension and reactivated
So, what if I increase that to 7200seconds...?
In the response TC will not accept this and will request your device to use a much lower period . Probably 240 and your device will comply even though the setting may show 7200. Nat session closing in your router may block an invite but once the call starts the established connection will keep it open. I dont know if TC use re-invites but if these aren't responded to the media will be stopped by TC or they may have a 1 hr limit on calls I don't know. Usually its 2 hrs.
So, what if I increase that to 7200seconds...?
Have you turned on sip ping options for the extension.
Already submitted a ticket for this but thought I'd post it here so people would be aware of this. I only noticed it by accident.
A few of my extensions are completely scrambled, in that the call to them goes somewhere else, and a few of them even say "Forbidden" when I try to dial out from them.
I've also lost the ability to change call barring option on all extensions.
Has anyone noticed something similar?
Oh dear.... Well, it still dropped off after an hour..
I also have this problem and have raised a ticket but so far no response.
My calls get disconnected at exactly the same call duration of 60 min and 32 secs (according the the TC log).
Nothing has changed in my router/ATA (AVM 7490) and this has only recently become a problem.
My calls get disconnected at exactly the same call duration of 60 min and 32 secs (according the the TC log).
Can you email support with the extension that this is happening on please and I can setup logging on it and see where the BYE is coming from.
A few of my extensions are completely scrambled, in that the call to them goes somewhere else, and a few of them even say "Forbidden" when I try to dial out from them.
I've also lost the ability to change call barring option on all extensions.
Okay so just to update, some of my scrambled extensions had call forwarding applied them for some unknown reason (maybe they were inactive for a long time). The advice given to me worked (dial *21** on the affected extension to cancel the call forward).
As for the "forbidden" extensions, a simple deactivation and reactivation solved that.
The call barring will apparently be fixed soon.
Thanks to John and the team for all the help!
Nice to see a smily face for a change ;-)
Nice to see a smiley face for a change ;-)
There are a lot more of those who never bother posting because like me it is all working to their satisfaction.
Not sure if this is a TC issue a router issue or a SNOM issue but I have had my last few calls cut out at 15 minutes 30 seconds ...which is really annoying when the first 10 minutes of each call is spent on hold ( banks).
Anyone else experiencing similar or can anyone suggest a router or phone setting to check
are there any better android apps to use then Cloud Softphone??
its annoying to switch between two different Telecube accounts.... dunno why it doesn't have multiple accounts/profiles
are there any better android apps to use then Cloud Softphone?
CSipSimple works for me, reasonably easy to change between accounts but doesn't have G.729 in the free version.
are there any better android apps
You can switch between accounts easily in Zoiper
Although bluetooth doesn't work that well with it
You can switch between accounts easily in Zoiper
I have used Zoiper off and on for years with multiple accounts. Seems like everytime I switch to another account the previous one is deleted. I eventually gave up and just have one account, too much trouble for the one phone call a year i make.
Have you turned on sip ping options for the extension.
Thanks for the advice. I have tried that. Sadly, it made no difference at all. I'm now waiting for John to find out what can be done.
setup CSipSimple, moves between multiple accounts nice enough, however is G.729 a must have?
Also quite liked interface in Grandstream Wave app
How do I fix this issue?
I have softphone setup on my phone, computer etc etc.
When someone calls my DID, on my phone it shows up as my own DID. So it does not show who is calling me ? How do i fix this?
What's your Optus account number for 1300 numbers?
however is G.729 a must have?
Not, but its very nice to save a bit of bandwidth when bandwidth is constrained or expensive. if iLBC worked properly it would be a much better choice over a wide range of network conditions. As it is, you need to either choose a poor quality codec, GSM or more bandwidth intensive ones, G.722 and G.711 (and G.722 might also have transcoding(?) issues when calling off-net to the PSTN, not that I've tested it sufficiently.)
Any way, I'd be interested in whether anyone is having good experience with iLBC (and to a lesser extent G.722) when calling the PSTN, particularly with CSipSimple.
I haven't got around to checking what codecs Cloudphone supports so I'm curious whether iLBC or G.722 is better with it or any other client.
What's your Optus account number for 1300 numbers?
If you mean for porting a 1300 number away you just need to quote your Telecube customer id now the same as geo numbers.
You will have set the DID as Caller ID option in the manage screen for the DID
Yeah, but when I make a call from that extension, I want the caller ID to come up as a particular number.
We are talking about incoming calls.
Yeah, but when I make a call from that extension, I want the caller ID to come up as a particular number.
If you want outbound caller id to be a did you need to manage that in the extension screen. A DID isn't an extension
We are talking about incoming calls.
If you set DID as Caller ID to yes in the DID management screen then calls in through that did will show that did as the caller when the calls gets to you, instead of the actual caller id passing through.
Finite State Machine writes...
Any way, I'd be interested in whether anyone is having good experience with iLBC
Going back 12 months or so before the days of unlimited calls on mobile I used iLBC with Zoiper as an IAX client and it worked well. I've just fired it up to see if it still works. It would seem IAX is no longer supported on the new platform at TC, so can't give an up to date assessment now.
A DID isn't an extension
That's why I am of the opinion the VoIP industry should stop calling them a DID. DID is a programmed process that tells the P(A)BX what to do with public/geographic numbers when they are received. Just because someone ports a number from say Telstra to a VSP doesn't change the type of number is, it is still a geographic number in the ACMA numbering scheme.
after i starting using CSipSimple, Cloud Softphone no longer has sound when calling out, however calls coming in (via DID) have sound...
calls out on CSipSimple have sound.
i'm not running both apps at once
i've uninstalled Cloud Softphone and installed it again, however the outbound calls still dont have sound...
anyone have any ideas?
after i starting using CSipSimple, Cloud Softphone no longer has sound when calling out, however calls coming in (via DID) have sound...
calls out on CSipSimple have sound.
i'm not running both apps at once
Sorry I have no idea how to fix this however; I just thought I would say that I have this same combo working on my 3rd gen Moto G running Marshmallow and there is no issue with either app.
I've been using CSiSimple for years on various accounts (and various handsets) but have only recently installed Cloud Softphone, which is a great app but a pain if you have multiple accounts.
having issue registering my 3 extensions via CSipSimple..... sometimes they register, other times they dont, i'm connecting to sip.telecube.com.au
to be honest i'm considering moving now, telecube can't coming across as very reliable
telecube can't coming across as very reliable
They have been great. They are just working through some problems right now. I think they will be great again.
+1
+1 ;-)
Bear with us please .. things have stabilised and we're just ironing out a few bugs.
Zoiper as an IAX client
It would seem IAX is no longer supported
I have same/similar config and yes it doesn't work at the moment. TC support say they don't have an ETA for fixing IAX registrations yet. No doubt they will get around to this when they get on top of the other issues. No doubt they have to prioritise what they work on at the moment.
I have found IAX seems to survive bad NAT implementations better than SIP so in my case is used for backup only and not desperate for a quick fix � would like it to work eventually.
If you want outbound caller id to be a did you need to manage that in the extension screen. A DID isn't an extension
I'm well aware of the difference between an extension and a DID. You may like to clarify the telecube settings page so the difference is clear.
TC support say they don't have an ETA for fixing IAX registrations yet.
With OpenSIPS proxy it will never be fixed because of the fact that OpenSIPS is a pure SIP proxy and doesn't support IAX. Ofcourse John can introduce an IAX only Asterisk server so people can register to it directly. John's decision to deploy OpenSIPS tells me his Asterisk server resources have hit the upper limit and wouldn't scale up any more.
Now that he has his shiny new SIP proxy, I was wondering whether we will see multiple registrations for the same extension.
That's why I am of the opinion the VoIP industry should stop calling them a DID. D
are you serious? The issue isnt anything to do with terminology, it's that the Telecube site isnt clear what all these options are, and settings were changed without informing us of the changes. Nor does it have anything to do with your crusade to change DID terminology with what works well across the world.
It had been working for months, and then with all the changes in the last few weeks, many things were modified, one of them (not user changes) was this incoming DID setting. As you can tell with the number of people asking about this, it's widespread.
I'm happy to stick around with Telecube, but please dont make out it's my problem (John M), when it was a change on the programming side, or user confusion over terminology (waiting for NBN).
but please dont make out it's my problem (John M), when it was a change on the programming side
Actually there hasn't been any programming change in regard to the DID as Caller ID setting, the only issue is that with the forced move to the new platform that feature wasn't working for approximately 2 weeks and it's my expectation that people set it thinking it was something else and when nothing happened they just left it.
Then once I restored the feature, those who had set it and forgotten started getting the DID coming through as caller id and couldn't understand why.
Edit: I should add though that I can see how it can be confusing and will add information on the screen to explain the purpose of the option.
Edit: I should add though that I can see how it can be confusing and will add information on the screen to explain the purpose of the option.
Currently is says DID as Caller ID. Maybe a change to DID as inbound Caller ID all calls.
many things were modified, one of them (not user changes) was this incoming DID setting.
Fortunately it hasn't changed for me and if it does I will no longer use it as I use CID to monitor calls before answering.
once I restored the feature, those who had set it and forgotten started getting the DID coming through as caller id and couldn't understand why.
This is exactly what happened in my case.
I should add though that I can see how it can be confusing and will add information on the screen to explain the purpose of the option.
Good idea. It will probably save you answering a lot of support tickets.
/forum-replies.cfm?t=2531065&p=72#r1426
Thank you, now the caller number is displayed when calling in
Can anyone advise if Caller ID authorisations are working again? I don't get the authorisation call as described in the portal.
Hi John, I have been having some difficulties regarding incoming calls for sometime now (ever since the outage started.) Basically, with all my incoming calls, they would automatically cut itself off after exactly 60 minutes.
I just wanted to thank John from TC for the fixing the problem for me. The service provided is absolutely amazing, and I am so grateful to him for the time and effort he has put into rectifying the problems.
I'm a very happy customer! Thank you so much!
I'm a very happy customer! Thank you so much!
+1
Thanks John, this also fixed my problem of calls being dropped at 60 mins duration,
Yep I found a config issue on our side, it should be resolved now. Apologies
Can anyone advise if Caller ID authorisations are working again? I don't get the authorisation call as described in the portal.
Apologies .. caller id authorisation is working now.
Trying to register FreePBX here on sip.telecube.net.au
Throwing a 403 error on your server, John � but doesn't tell me why.
[2016-06-20 13:49:35] WARNING[13804] res_pjsip_outbound_registration.c: Fatal response '403' received from 'sip:sip.telecube.net.au:5060' on registration attempt to 'sip:<ext>@sip.telecube.net.au:5060', stopping outbound registration
Trying to register FreePBX here on sip.telecube.net.au
Throwing a 403 error on your server, John � but doesn't tell me why.
Did it work previously?
Is it a new install?
Did you try sip.telecube.com.au?
Did you try sip.telecube.com.au?
I'd been previously trying to register to that endpoint as well; with no success.
Is it a new install?
Reasonably; yeah. I've not done something, haven't I? It's something stupid.
Throwing a 403 error
Make sure you have set fromuser= and username=
Also try one of the hostnames below, they are mapped directly to registration servers and not through the proxies.
sip1.nsw.telecube.com.au
sip2
sip1.vic.telecube.com.au
Actually .. only use the static hostnames for registration .. not for IP Auth
I've not done something
The wiki info should work for registration....
General Settings
Trunk Name: Telecube
Outbound Caller ID: <extension ID>
PJSIP Settings
Username: <extension ID>
Secret: <password>
SIP Server: sip.telecube.com.au
Contact User: <extension ID>
Context: from-pstn
A from user field may also have to be set to <extension ID>
Make sure you have set fromuser= and username=
Set all. Same error, just the sip1.nsw address now.
Also do not use from user if you are wanting IP auth with CLID pass through. If you do you your CLID will not be presented (technically it will be your extension user ID but this will not be passed out of telecube.)
Also do not use from user if you are wanting IP auth with CLID passthrough.
I'm using from-pstn. :)
I'm using from-pstn. :)
I'm not talking about the context. I mean the fromuser setting which translates into 12345689@sip.telecube.com.au on all the sip traffic if you have the fromuser setting set.
Same error
Your error snippet shows pjsip.
On a fresh install creating a pjsip trunk.
General Tab...
Trunk Name: Telecube
Outbound CallerID: TC_Extension
PJSIP Settings Tab...
General Tab...
Username: TC_Extension
Secret: TC_Extension_Password
SIP Server: sip.telecube.com.au
Advanced Tab...
Permanent Auth Rejection: Deselected
Expiration: 180
Contact User: TC_Extension
From User: TC_Extension
Submit and Reload
Yay... DID's are now ringing.
Yay... DID's are now ringing.
Yes the DIDs affected by the outage on the Optus link are restored now, we're just checking some last details and I'll send out a status update.
I am still having calls drop out after 15 minutes ..... any clues please.
I am still having calls drop out after 15 minutes ..... any clues please.
Did you email support with your details so I can monitor a call?
No I wasn't sure if it was your problem or my end. I will submit ticket now.
No I wasn't sure if it was your problem or my end. I will submit ticket now.
Yes please, even if it's not on our side I might still be able to see something here that could give a clue.
sip2.nsw.telecube.com.au is NOT working. Handset registers OK, but portal displays offline. Changing it to sip1 has registrations on both handset and portal...
sip2.nsw.telecube.com.au is NOT working. Handset registers OK, but portal displays offline. Changing it to sip1 has registrations on both handset and portal...
Can you switch back to sip2.nsw again please and let me know if it is showing the correct status in the portal now?
sip2.nsw again please and let me know if it is showing the correct status in the portal now?
They do here.
Still offline in portal. Note, customer site was experiencing no DID to call queue and no calls to local extensions. So will keep them on sip1
Hmmm, I have one last remaining extension on sip2.nsw and is now showing online in portal. This wasn't the case when initially asked (and yes after a browser refresh)
This wasn't the case when initially asked (and yes after a browser refresh)
I found something that may have been causing it and think I fixed it. If you could test and report back I'd appreciate it.
BTW, how long before MOH is restored? Is there any way to not have the default?
BTW, how long before MOH is restored? Is there any way to not have the default?
Working on it now .. hopefully not too much longer.
Are the .nsw servers actually in Sydney as I'm still seeing 25ms pings which is the same as for the main servers.
Prior to all the recent issues I was getting 11-12ms on the .com.au which I believe was in Sydney
Not really ticket-worthy so I'll ask here
I have some extensions registered on various mobile phone apps around our business. It's a pain to get hold of everyone when I need/want to change the registration server.
Would there be any negative side to creating a CNAME record for our company domain name � eg. voip.mycompany.com with a low TTL and point it to the Telecube server of choice? Works in my head, unless there's some kind of host headers thing in the SIP protocol
Edit: Would also save me a lot of time updating 20 odd phones too
Are the .nsw servers actually in Sydney as I'm still seeing 25ms pings which is the same as for the main servers.
They are .. could be that routing from your connection is through Melbourne though, if you send a trace route to support and ask we can have a look
Would there be any negative side to creating a CNAME record for our company domain name
It should be fine if you point it to one of the static hostnames, sipN.(nsw|vic). etc but wouldn't work if you point it to one of the hostnames currently running through the proxies.
Can't call Germany....returning busy signal since yesterday.
Can't call Germany....returning busy signal since yesterday.
We've an outage at the moment for international calls .. please subscribe to the alerts for updates.
Were there intermittent DID incoming issues this afternoon? Looking at the CDR some of our incoming calls dropped at 2 sec and some calls came in through the PSTN failover line and then cut out. I can also see one call with an 8 digit caller id number; used to be 10 digits.
@keychange Did you find a solution to your 15 minute call cut out issue? I have exactly the same problem using an Aastra handset......
Is anyone getting horrendous battery usage from Cloud SoftPhone on Android?
I have been using it for the past couple of months without issue. In the last few days it has been really bad though. It has used 18% of my battery in 2.5 hours.
No config changes or anything. Push notification in use, etc.
"We are sorry, international calls are currently disabled on your account"
Anyone anyideas?
Edit, just read above post.
I am planning to hook up a small business I know to VoIP in a month or two, including porting their current number over. They will have a large number of incoming calls and a small number of outbound calls since they use SMS alerts instead of calling clients. I am wondering if this is ok since John posted this in another thread: whrl.pl/ReEj1F
Also wondering if the 'troubles' are well and truly over? I still see minor issues every now and then.
Edit, just read above post.
If you mean the message from John 24 hours ago with a link that says no problems?
Yes I'm getting the same message as you.
"We are sorry, international calls are currently disabled on your account"
Anyone anyideas?
Edit, just read above post.
In case you missed the email...
International calls are restored now.
�
The fault turned out to be related to a distributed database configuration problem and unfortunately you will need to set the international calling preferences in the portal again.
�
Apologies for the inconvenience.
I think the status page should maintain a 7 day rolling record of issue/resolution.
They will have a large number of incoming calls and a small number of outbound calls since they use SMS alerts instead of calling clients. I am wondering if this is ok since John posted this in another thread: whrl.pl/ReEj1F
It's against the FUP, we may charge for inbound calls
I think the status page should maintain a 7 day rolling record of issue/resolution.
Yep I'll include resolved events in a section below on the main page
we may charge for inbound calls
Did you just land from the US ? :)
What would you consider a healthy ratio?
What would you consider a healthy ratio?
It will depend on the volumes, the main reason for the policy is to stop calling card type services using our platform.
Ultimately if it's a profitable account I'll probably let it slide.
main reason for the policy is to stop calling card type services using our platform.
Thanks John. This is a small medical practice so call volumes are not very high; may be around 30 a day inbound and 0-5 a day outbound.
small medical practice
Do they have internet. John does internet.
Have you seen this... http://www.medicalchannel.com.au/
Value add, upsell, blah blah blah :-)
Also wondering if the 'troubles' are well and truly over?
There's still a couple of areas related to database distribution that make us vulnerable to a specific series of events but I should have them tidied up over the weekend.
If people start registering to the static hostnames; sipN(nsw|vic).telecube.com.au that is essentially the original way we have been operating up until the recent problems and we'll balance load by having people register to different hostnames.
We're also providing a reference in the portal to the current percentage used of max capacity that a server can handle so people can select the lightest loaded hostname to use.
We're also providing a reference in the portal to the current percentage used of max capacity that a server can handle
Yes, just discovered this a few days ago.... nice idea! Good to have the proper sipN(nsw|vic).telecube.com.au names provided in the portal as well. That should make things a lot clearer for people.
There's still a couple of areas
Are international calls from IP-auth connections allowed, yet? I've turned it on in the portal but still no joy. Not a show-stopper, just wanting to know where it's at.
Has IP-auth been integrated into the new platform?
Hi anyone else have an outage on NBN, I'm in Brunswick??
What email?
And how do you set the international calling preferences in the portal again?
And how do you set the international calling preferences in the portal again?
If you go to the preferences section there's an option there to enable international calls.
Hi anyone else have an outage on NBN, I'm in Brunswick??
We've had reports of ADSL/NBN services going down, I've no confirmation from upstream at this stage but we are investigating.
Please subscribe to the status alerts for updates www.telecube.io
We're also providing a reference in the portal to the current percentage used of max capacity that a server can handle so people can select the lightest loaded hostname to use.
Anyway you can display/highlight the current hostname that a device is using?
Anyway you can display/highlight the current hostname that a device is using?
Yep, where would you like it displayed?
works now thanks John.
We've had reports of ADSL/NBN services going down, I've no confirmation from upstream at this stage but we are investigating.
Please subscribe to the status alerts for updates www.telecube.io
Hi John. I am subscribed but haven't received anything about an outage.
Hi John. I am subscribed but haven't received anything about an outage.
I sent an alert out at 21:59 tonight .. did you not get it?
Do they have internet. John does internet.
They are under contract......
Have you seen this...
No but interesting.........targeted advertising, literally!
If people start registering to the static hostnames; sipN(nsw|vic).telecube.com.au that is essentially the original way we have been operating up until the recent problems
John I am a bit confused here........don't you want all SIP signalling handled by OpenSIPS?
so people can select the lightest loaded hostname to use.
I am not so sure about this; normally businesses would like to set it once and forget it. Isn't there a way for you to handle server load in the background?
John I am a bit confused here........don't you want all SIP signalling handled by OpenSIPS?
Call invites yes but I want registrations direct to the servers, different extensions can register to different servers unless they are part of a BLF or Call Park/Pickup group.
IP based auth will send calls to sip.telecube.com.au which will be load balanced Opensips servers
I am not so sure about this; normally businesses would like to set it once and forget it. Isn't there a way for you to handle server load in the background?
They will only need to set it once, I'll keep adding hostnames and hiding them from the portal once they reach a certain load.
I'll keep adding hostnames and hiding them from the portal once they reach a certain load.
Aha.......good man :)
I sent an alert out at 21:59 tonight .. did you not get it?
It came through OK for me John. And I've had no problems with my ADSL � just checked the router and the connection has been up since my last manual reboot 5 days ago.
Is there still an issue with time based routing for 1300 numbers?
Getting a message that international calls on our account is currently disabled
Can you please advise if this is a known fault? Or fix the issue, all calls are enabled on all the extensions
I tried barring international calls on an extension, however it doesn't stick, if i go back into the extension settings the international calls are re-enabled by default
Can you please advise if this is a known fault? Or fix the issue
You have to log into your account & enable it again.
/forum-replies.cfm?t=2531065&p=78#r1541
And I've had no problems with my ADSL
Neither have we. I saw the outage notice but our connection was ticking along just fine.
For some reason, when I was with iiNet, my Asus DSL-AC68U used to report lots of issues, loss of connection, etc. whereas with Telecube, the only thing appearing in the log is the NTP updates every 12 hours. Very stable connection!
You have to log into your account & enable it again.
I've tried, however I can't load the preferences page
I've tried 3 different browsers and 2 different connections, however it sits on the "loading" animation
edit: although support are telling me there's no issues and to clear my cache
I've tried, however I can't load the preferences page
I've tried 3 different browsers and 2 different connections, however it sits on the "loading" animation
edit: although support are telling me there's no issues and to clear my cache
I had exactly same issue, with Safari on Mac
Tried a different browser probably Firefox or Chrome ( i dont remember) and its fine. I had never used that browser on Telecube before.
I've tried, however I can't load the preferences page
What browser are you seeing this problem in please?
Can you buy a NZ DID through Telecube?
Can you buy a NZ DID through Telecube?
Yes, please email support and ask and someone will help you.
What browser are you seeing this problem in please?
All safari (mac and iPhone)
Worked last i checked a few weeks ago
All safari (mac and iPhone)
Ok, I've found the problem .. should have it sorted later tonight.
Thank you
OK, I've fixed it now .. sorry about that. Javascript coding bug that affected Safari.
It's working again, thanks John
Another minor issue you may want to investigate, not sure if this is by design
This is the way call history is formatted and displayed for international calls in the extensions call history under the "details" column
Jun 03 17:17 0:51 98823333 {"destination":null,"useragent":"CM5K (706070)","peerip":"103.193.16
Just wondering if selecting call barring options will be working again any time soon.
Thanks for all the hard work and also glad to report that my customer is very happy once again. :)
Can anyone explain in detail how to "set the international calling preferences in the portal"
Log in, Account Details, preferences tab, International calls enabled: change from Off to On � All Extensions.
Thanks, also found it in the wiki
http://wiki.telecube.com
I have a question about the different SIP servers that are available. I just notice this in the portal:
SIP Hostname: sip1.nsw.telecube.com.au (55.84% capacity)
SIP Hostname: sip2.nsw.telecube.com.au (56.96% capacity)
SIP Hostname: sip1.vic.telecube.com.au (58.28% capacity)
Were does
sip.telecube.com.au [103.193.167.161]
sip.telecube.net.au [103.193.167.162]
come into this as the sip servers above are different.
sip1.nsw.telecube.com.au [103.193.166.33]
sip2.nsw.telecube.com.au [103.193.166.36]
sip1.vic.telecube.com.au [103.193.167.53]
I'm located in QLD so I don't have a local SIP server. I'm guessing it doesn't matter too much which one is used as long as it is not over utilised.
I'm located in QLD so I don't have a local SIP server
In QLD you should register to one of the new servers
sip.telecube.com.au will eventually just be used by IP auth trunks and not registered extensions
sip.telecube.com.au will eventually just be used by IP auth trunks and not registered extensions
And what about sip.telecube.net.au ?
All my extns are currently registered to this.
I'm in WA.
And what about sip.telecube.net.au ?
It will eventually be only used for ip auth trunks .. it's not something that is going to happen any time soon though, there'll be plenty of time to make changes.
We'll just gradually ween everyone off the registration proxy processes and onto the direct hostnames
I really think you want something more dynamic and automatic than that. Varying the probabilities and hosts in a per state SRV record seems like a reasonable place to start.
Sometime in the last couple of days my Telecube extension stopped registering. I only noticed it this morning after setting up a Siptalk account and adding it to my Minitar ATA.
Neither extension is registering. I have not changed any setting on the Telecube extension.
The servers I am using are sip1.vic.telecube.com.au and sim.siptalk.com.au. I'm located in Adelaide, ISP is iiNet.
I have rebooted the Minitar, rebooted the router and rebooted everything, but still neither are registering.
[edit] finally gave up on sorting it out and decided to use spare ATA to eliminate the Minitar or not. Worked fine. Went back to the Minitar, and it worked fine too. Go figure. It's getting old, maybe it just need a rest.
The indicator lights alongside the extension names no longer turn red on my T46G when any of the W52P extensions are in use. This was always the case before the recent tech problems. I have not changed anything at my end except the URL under instructions from the Telecube Team. My current SIP server is sip1.vic.telecube.com.au
Any assistance would be greatly appreciated.
My current SIP server is sip1.vic.telecube.com.au
Are all the extensions that are in the BLF group registered to the same hostname? It's important that they are.
Yes, John they are all registered to sip1.vic.telecube.com.au Port 5060
The indicator lights alongside the extension names no longer turn red on my T46G
If you restart the T46G handset do they start working again?
Is there any reason why incoming calls show up as my DID instead of the caller?
Is there any reason why incoming calls show up as my DID instead of the caller?
Check in the manage screen for the DID and you will see an option "DID as Caller ID" has been set to Yes .. change it to No
see an option "DID as Caller ID"
Ahh easy fix � I don't ever remember changing that. Maybe I thought it was the outgoing caller ID or something.
If you restart the T46G handset do they start working again?
No, John. I have rebooted everything every way which way. The only thing I have not done is repacked them and unpacked them again from their cartons.
I have rebooted everything every way which way
Email support please so we can have a look
John...
Should international calling be available now on IP-authenticated trunks? I've (re)activated it for 'all extensions' in the portal.
I cannot login to the management portal.... anyone else? I'm on Telstra Bus ADSL
I cannot login to the management portal.... anyone else? I'm on Telstra Bus ADSL
Works fine here (Telstra Business ADSL in NSW).
Thanks for that...
I'm finding BLF stops working every couple of days. Sometimes a reboot of the handsets fixes it, but other times it does not.
Would be great to have it work as we have people interstate and it's good to see when they're on the phone.
All extensions are registered to the same SIP server.
Sorry I might have asked this already but Call Barring settings don't seem to be sticking.
Is anyone else having the same issue?
See this thread
/forum-replies.cfm?t=2544833
Account details in the TC & ST portals show as the letter 'e'.
Details gone, replaced with 'e' ??
Yes, I had same problem and thought that setting was for outgoing Caller ID.
Not to Telecube: Suggest you add 'Outgoing' in front of label "DID as Caller ID:"
Can't seem to register from outside of Australia (in Japan) with either Zoiper or MicroSip.
Cloud Softphone seems to be working okay.
Suggest you add 'Outgoing' in front of label "DID as Caller ID:"
No, that's not correct. The option to set the DID as Caller ID means what you see as the Caller ID when the extension rings. The idea is that for set-ups with multiple DIDs pointing to the same queue you can see which DID the call is coming in on. If you set this to off, you will see the calling person's Caller ID.
I think maybe you mean "Incoming DID as Caller ID".
I think maybe you mean "Incoming DID as Caller ID"
Can't seem to register from outside of Australia (in Japan) with either Zoiper or MicroSip.
Cloud Softphone seems to be working okay.
Back to normal...
Portal details all have an "e" for one of my accounts.
Do I need to re-edit these back to proper name / address???
Regional routing for 1300 numbers seems to be back but it's not ringing the extensions configured. It's skipping the first answer point (extension) and going straight to the voicemail.
Have lodged a ticket #876-720-577
I am currently getting a message "Your call cannot be completed due to a temporary error" when trying to make an international call (tried both India and US). This occurs both from Asterisk and the Cloud softphone. International calls have been enabled in the portal.
Is anyone else experiencing this?
Yes, same here. Wife just tried to call UK and got a message to say international calls have been disabled.
Yes, same here. Wife just tried to call UK and got a message to say international calls have been disabled.
I initially got this message, but updated my settings in the portal to allow international calls (which had been barred). Following this I started to get the other message "Your call cannot be completed due to a temporary error"
I'm not getting that message but some tones and unable to call internationally. Calls between extensions ( one in Oz, the other abroad ) appear as normal.
Same here. No international calls possible. Tried Telecube and Siptalk, both the same.
Anyone else still having ex-Optus DID problems? Ours didn't come back online on the 20th June, so raised a ticket [#226-879-331] and was advised it would be sorted on the 30th June, but it is still not working. It was originally a Telstra number, ported to Optus and then onto Telecube. I still get DID Outage emails when someone tries to call it (which was handy during the election campaign), but would be good to get it working again soon :)
My June invoice does not work. I get this error: "Error: The PDF was not found. Please contact customer service on 1300 481 808"
My June invoice does not work.
Yes, I had that problem as well. I wonder when all these little bugs will be ironed out.
Yes, I had that problem as well.
John has addressed this issue, yesterday I think. He said he is working on the problem. If you search you will find his post.
Has anyone here figured out how to use the Intercom features of Yealink with Telecube?
I can call the extensions just fine, have made DSSKeys with Type: Intercom, and "Accept Intercom" is enabled in the target phone's Yealink receiving settings.
However, the target phone just rings and rings till someone picks up to answer � no intercom. I tried to add *50 to the number (won't make the call if I do this), and tried it as an extension with BLF (works like a regular call, no intercom).
It looks like a workaround is to set the target phone to auto-answer calls � and make a BLF DSSKey on other phones to call it. However, this works only if you don't make it part of a ring group or hunt group, otherwise it will answer all the calls.
I'm not getting that message but some tones and unable to call internationally.
Thanks John. It all works now.
No international calls for us either :(
No international calls for us either :(
Have you checked the over-ride setting in the portal? In my case this had clicked over to "disabled" � I think when John does work on the admin area it can sometimes wipe some of the settings.
Have you checked the over-ride setting in the portal? In my case this had clicked over to "disabled" � I think when John does work on the admin area it can sometimes wipe some of the settings.
I don't know how to do proper quoting on my phone. Sorry.
Yep me too. Support told me to re-enable international calls and it works like a bought one. I'll know next time.
How long do voicemail greetings uploaded via the web site take to kick in? I set them at 4.30 this afternoon and they haven't kicked in yet (no voicemail message at all played, previously was the default "the person at extension XXXXX... ").
The last time I did this i think they took an hour or so to work but it's been 6 hours now.
What format is your message file?
I found .m4a did not work but .wav worked straight away.
Any way to get rid of the OTP? (One time pin)
Any way to get rid of the OTP? (One time pin)
There's an option in the manage otp screen to authorise a browser for a maximum of 7 days
I saw that, but we have 4 people on call, and they change the divert every fortnight, so the 7 day thing is useless.
In the web portal at the top of the list of extensions there is an option to set "Save State" on or off.
What does this do?
I saw that, but we have 4 people on call, and they change the divert every fortnight, so the 7 day thing is useless.
Add their email addresses as endpoints then they can get the otp themselves and make the changes normally.
In the web portal at the top of the list of extensions there is an option to set "Save State" on or off.
What does this do?
It should remember the search, ordering or number of records selections for when you return to the screen.
Does anyone here know how to make Call Barring work?
I've ticked the boxes on the extensions I want to deny outbound calls, the pages say "update success", but calls can still be made from them.
Does it take time to happen? Do I need to re-register the phones? Is there another hidden setting (Yealink T23G)?
Does anyone here know how to make Call Barring work?
There was a bug .. it should be fixed now.
Hi John, are tc to tc fixed line calls free? both did's are in the same account yet we're being charged for the calls
are tc to tc fixed line calls free? both did's are in the same account yet we're being charged for the calls
From experience.
If you dial the extension, they are free. If you dial the DID, it will be charged.
If you dial the DID, it will be charged.
With TC the calls seem to go into the pstn system ( hence the local call charge ) then back in via the DID. With MNF for example the call stays within their system ( on-net) and it remains an ext-ext call and no charge. It just depends on how the vsp has set it up.
are tc to tc fixed line calls free?
No, DID to DID calls are charged.
For free calls between extensions either dial the extension number or set up three-digit aliases and dial these.
Thank you for clearing that up guys...
I'd rather pay not to remember the extensions
Extensions on the same account can be assigned a three-digit alias.
For example, say you have two extensions 1234567 and 6424743. You can assign them as 201 and 202 respectively. Then just dial those numbers to call the extensions, for free.
Thanks for the explanation habakkuk
It's mainly for my mother so I try and keep things simple
No, DID to DID calls are charged.
This is actually on the TC Feature Request list so there may well come a day when calls to TC and Siptalk DIDs are free when made from a TC or ST extension.
It's mainly for my mother so I try and keep things simple
This feature would be perfect for your mother.
She wants to call Ali, just press 123.
Or even ALI = 254
BTW, how long before MOH is restored? Is there any way to not have the default?
Working on it now .. hopefully not too much longer.
Hi John, any update on this?
Hi John, any update on this?
I've got it open in front of me now and working on it .. probably end of the week.
I've got it open in front of me now and working on it .. probably end of the week.
Wonderful, thank you. =)
Hello all. I'm using a VG3631 which has two sip accounts configured. One passes through the callerID of the caller, but calls to my telecube hosted DID appear to the handset, and to the modem call log, to have my DID number as the caller ID. Looking in my call logs in telecube 'My Account' page I can see the real caller IDs however.
Any ideas how i can get the real caller ID?
but calls to my telecube hosted DID appear to the handset, and to the modem call log, to have my DID number as the caller ID
On the DID configuration page in your account, turn off "DID as Caller ID".
Thanks :)
Is anyone else getting registration failures "Failed (408)"?
Edit: It's working again. Must be a temporary "blip".
Is anyone else seeing strange media offer behaviour that does not match the 'codec order' specified in the DID configuration on telecube site? It appears Telecube always prioritises ULAW above everything else if it is listed as an allowed codec. If ULAW is not listed, things work as configured.
When codec order first field is ULAW, first offer codec in the SIP Invite packet is PCMU
When codec order first field is G722 and second codec is ULAW, first offer codec in the SIP Invite packet is PCMU. This represents a defect.
When codec order first field is G722 and third codec is ULAW, first offer codec in the SIP Invite packet is PCMU. This represents a defect.
When codec order first field is G722 and fourth codec is ULAW, first offer codec in the SIP Invite packet is PCMU. This represents a defect.
When codec order first field is G722 and fifth codec is ULAW, first offer codec in the SIP Invite packet is PCMU. This represents a defect.
When codec order first field is G722 and ULAW is not listed as the second, third, fourth of fifth codec, then the first offered codec in the SIP Invite packet is G722. Behaviour is correct.
Anyone else seeing this? Any workarounds other than removing ULAW from the allowed list of codecs?
See screenshots of telecube config and associated packet capture.
Test 1 (expected bahviour) � http://i68.tinypic.com/wl3uio.png
Test 2 (defect) � http://i65.tinypic.com/257ojrl.png
Test 3 (defect) � http://i65.tinypic.com/2vkylba.png
Anyone else seeing this?
I think Robnll touched on this sometime back.
Telecube may have a preferred codec option set in their system.
Any known workaround (other than exclusing ULAW from allowed codecs list)?
Any known workaround (other than exclusing ULAW from allowed codecs list)?
Yes the actual codec order for an incoming call doesn't necessarily match the setting in the portal and never has. I actually like G711u media so I have put my portal order as Ulaw/Alaw/G722 and an Invite coming from a TC DID always offers codecs in this order so I accept PCMU and the call proceeds with G711u media.
It's very odd. Everything seems to works as it should until you add the G711U codec and all of a sudden it overtakes everything. Very annoying that it's not documented. I wasted quite some time checking config, testing and re-checking but in the end it looks like a known unpublished telecube defect.
Maybe there is a reason for it and John will chime in.
Hi,
tried to make an International Call. It wouldn't allow me to do this. This is the second time this has happened.
So I checked my account on the telecube site, and the status is both ACTIVE and ONLINE.
Then went to my gigaset config page, and it's status is REGISTRATION FAILED.
Why is telecube's platform so unstable? Why are the settings dropping out so often?
Really, all we want is to be able to make the odd phone call whenever we need to. I can understand the odd problem, but this seems to be regular and ongoing. Why?
What settings do I need to add now, in order for the registration to be active within my gigaset config page?
Getting too hard if you ask me and not worth the trouble.
registration to be active
Maybe a symptom of the problem but lack of registration is probably nothing to do with inability to make calls.
Why is telecube's platform so unstable? Why are the settings dropping out so often?
I'm not having any issues with registration instability on either a Gigaset C470 or a Fritz!Box.
In the C470 I set the following settings:
- Authentication Name: Extension number
- Authentication password: Extension password
- Username: Extension number
- Display name: Extension number
� - Domain: sip1.nsw.telecube.com.au
- Proxy Server address: sip1.nsw.telecube.com.au
Registrar server: sip1.nsw.telecube.com.au
Everything else I left as default.
Maybe a symptom of the problem but lack of registration is probably nothing to do with inability to make calls.
The system is a gigaset 530.
The system can't make calls as it's registration keeps falling off the perch. So how can I make a call if it is not registered?
I'm not having any issues with registration instability on either a Gigaset C470 or a Fritz!Box.
In the C470 I set the following settings:
Authentication Name: Extension number
Authentication password: Extension password
Username: Extension number
Display name: Extension number
�
Domain: sip1.nsw.telecube.com.au
Proxy Server address: sip1.nsw.telecube.com.au
Registrar server: sip1.nsw.telecube.com.au
Everything else I left as default.
Thank you for your settings.
I changed them to reflect yours, but unfortunately it still didn't work.
There were some notable differences though. My sip goes through Vic so I left that as is. I did have some unusual number in the authentication name which is probably my account number but that is what support told me to put there and it did work for a while.
So I changed that to my extension number with my area code as well.
Is the area code also required, or not?
Is the area code also required, or not?
You definitely don't want the area code.
Your extension number is not a DID.
You definitely don't want the area code.
Your extension number is not a DID.
Thank you again.
I tried without area code. It didn't work.
I will have to call customer support tomorrow. :(
I will have to call customer support tomorrow. :(
Here is mine using C610 with siptalk.
https://i.imgur.com/MLThaIC.png
Change sip.siptalk.com.au to what shows in you extension management screen.
Here is mine using C610 with siptalk.
Mine is similar for Siptalk:
https://farm9.staticflickr.c
Is their some sort of delay or something with recorded calls? We had a phonecall this morning at 9:50am perth time and I still haven't received a copy of the call into our email? I've noticed this over the last few days
I still haven't received a copy of the call into our email? I've noticed this over the last few days
It's a common mis-conception that email delivery is instantaneous. It ain't necessarily so. The receiving mail server (ie yours) might be causing the delay here.
Call records are still showing extra info such as codec, IP etc.
Seems to currently be happening with international calls.
Can this be removed as it serves no real purpose and hogs space on the call records page.
Thanks.
Is their some sort of delay or something with recorded calls? We had a phonecall this morning at 9:50am perth time and I still haven't received a copy of the call into our email
Just had staff report this happening recently as well. This makes me doubly keen to see the feature to check voicemail box directly via handset and PIN.
Hi forum members � I have an strange issue and thought I would start here.
Setup is Telecube and MNF on a Fritz7390 and 2 voip handsets.- 1 each to the voip accounts.
I can call out on each on each voip account ok. The fritz has a facility where I can dial a Preselection code from a handset to call out on a different provider. This has worked in the past � but not now. This is great if you want to use a different provider for different rates.
EG: if I take the handset registered to the MNF a/c and dial the Preselection code for Telecube, I get a dial tone (and it shows the telecube extension), but when I call a number, I get a busy signal and call rejected.
If I change the handset to register with Telecube � the call goes thru. Has anyone experienced this receently with telecube?
It's a common mis-conception that email delivery is instantaneous. It ain't necessarily so. The receiving mail server (ie yours) might be causing the delay here.
Maybe but we use office365 and usually its instantly. Strangely enough the original still hasn't come through but others AFTER this time have come through. Looks like some are not making it at all? Im going to go back tomorrow and check since when this has been happening. I'd guess at least a week judging by some of the patterns Im seeing.
I just want to say thank you to John and Toran for their help today.
Toran was very helpful in isolating an issue. In the end it was a manufacturer hardware issue at my end but I really appreciate the prompt response from telecube staff.
So thanks for the help and enjoy a positive post in this thread for a change.
I've just setup a 1300 number and set a basic route to an existing extension (which is registered to our Asterisk 1.8 PBX). All working fine but Callerid shows number but prepends "Unknown" in front.
This looks like an Asterisk setup but any suggestions on where to look or a Telecube config change to fix?
Cheers
Is it just me? No incoming on DID. Caller gets offered VM although AFAIK I have none setup.
Now getting 500 � server failure.
Ticket raised � #934-934-766.
And fixed thanks Toran. Outstanding support.
Well actually not... ongoing.
Is there still an issue with time based routing for 1300 numbers
I ended up finding that using a � in the file name for a IVR voice clip makes it only play a few seconds before it loops again.
Has anyone else noticed call pickup has changed recently?
I used to be able to pickup any extensions ringing in my group by dialling *8# � this stopped working and I get "Forbidden" when i try it.
Yes, I have setup call pickup in the past, and the extensions are all listed correctly- I didn't change anything recently in my setup either.
I have recently noticed that Telecube is no longer connected to Micron21....
Does this mean that Telecube is no longer DDOS protected and can be the target for an attack again?
This is very worrying to say the least.
I have recently noticed that Telecube is no longer connected to Micron21....
Still connected, Still Protected... Nothing to worry about :)
Kindest Regards
James
I have recently noticed that Telecube is no longer connected to Micron21....
We are still connected .. no need to worry.
Anyone else having problems calling out? Extensions say they online and active and I get a dial tone from both, but every call gives me a non-connection � not sure if it's an engaged signal or an out-of-order signal. Tried phoning my own mobile and it's the same.
Just tried my mobile and a 1800 number. Both rang and were answered ok.
Thanks for checking Thunderbird1. I found out the problem � my account had gone down to -4c. I'd been overcharged in the previous bill and they'd reversed some of it but it doesn't actually get credited back into your account until the next billing cycle (4 weeks away).
Interesting to note that if your account goes into the negative you can still register your lines, they all still show up as active, you can still get a dial tone... it just won't complete the call.
No probs!
Interesting to note that if your account goes into the negative you can still register your lines, they all still show up as active, you can still get a dial tone... it just won't complete the call.
The amount of credit on your account should have no bearing on the actual registration status of your ext/s and rightly so.
I don't think you would be overly impressed if you had to maintain a positive balance in order to be able to make internal ext � ext calls ;-)
I don't think you would be overly impressed if you had to maintain a positive balance in order to be able to make internal ext � ext calls ;-)
Yep. hadn't thought about that. Well, live and learn. I'll know next time. :-)
I found out the problem � my account had gone down to -4c.
Do you have email alerts set for account balance? I have ours set for "Daily". I find it very useful.
I found out the problem � my account had gone down to -4c.
It would also help if TC sent an email when they deduct the monthly internet fee from our balance. A few days ago, I noticed that the fee had been deducted but I never got an email about it. If you had received an email, you would have seen immediately that you had been overcharged, so that you could have taken steps to avoid going negative.
It would also help if TC sent an email when they deduct the monthly internet fee from our balance
An email is supposed to go out but we've had an issue with it, it will be fixed before next bill run. Apologies
John,
Any news when wav uploading for IVR will work again?
Ticket #266518678
Thanks.
Any news on when IAX will be restored?
Thanks,
S.
Hi John
Is there any CTI/click to dial functionality for the hosted pabx?
Dave
Unable to make International Calls.
It was an important call too.
Why are International calls disabled?
International calls disabled?
This is the default position. TC want to talk to you on a Aus mobile number just to check you are a legit customer (prevent fraudsters using your account).
Shoot support an email along with a contact mobile numbe. You only need to do it once.
TC want to talk to you on a Aus mobile number
Do you do this before activating international calling under Preferences?
Any news when wav uploading for IVR will work again?
+1 for MOH.
These should be restored by the end of today or over the weekend. Apologies for the delay.
Any news on when IAX will be restored?
IAX should work if you register to one of the static hostnames.
Is there any CTI/click to dial functionality for the hosted pabx?
There's not sorry, it is planned but no ETA at this stage.
Unable to make International Calls.
You need to enable international calling in the preferences section in the portal
Thank you John :)
Also, another unrelated question in an area new to me....I have a NAS that can host Asterix � is there any benefit in running Asterix on it and can i still go through Telecube to take calls in ? What is the purpose of Asterix over just using Telecube directly via the modem/internet ?
I've just tried to get IAX working again, but it still doesn't register (times out).
is there any benefit in running Asterix
It's really only a benefit to run your own asterisk server if you plan to incorporate a PSTN backup or have some special functionality that we don't provide.
Hi all, Is there no way to have a 1300 number as a Caller-Id on a VoIP extension ?
I get a pop-up saying it must be an Aussie number or a mobile number.
Thanks, Andrew.
Hi all, Is there no way to have a 1300 number as a Caller-Id on a VoIP extension ?
I asked that ages ago, and never ended up doing anything in regards to it (we never ended up getting a 1300 number). IIRC John said that they can set it up manually, but that the number shown will have an extra zero at the front of it. Things might have changed since though.
It's really only a benefit to run your own asterisk server if you plan to incorporate a PSTN backup or have some special functionality that we don't provide.
Ah fair enough � well the PSTN line is where we get our net from (for now, looking at Cable soon which would make it redundant unless it's kept for emergencies) and you have all the functionality � just thought running part of it locally would be different (i.e. calls between extensions in the same network/subnet)
Hi all, Is there no way to have a 1300 number as a Caller-Id on a VoIP extension ?
Australian carriers don't allow 1300 numbers as caller id
i.e. calls between extensions in the same network/subnet
Yep this can be an advantage if you want to talk between extensions on the local network instead of across the internet
Australian carriers don't allow 1300 numbers as caller id
Is it possible to have a name as a caller ID? I know it can be setup for use with SMS messaging, but I've got no idea if it can be setup for phone calls.
It's actually something we might be interested in down the track if it can be done.
Is it possible to have a name as a caller ID?
Nope, also not allowed by Australian carriers
Yep this can be an advantage if you want to talk between extensions on the local network instead of across the internet
Excellent � I'll have a look at it tonight (if I don't get bamboozled by the jargon !).
Also, does Telecube support Grandstream phones ? I'm looking at the GXP2160 to replace the Yealinks.
Nope, also not allowed by Australian carriers
I thought that might be the case. Pity.
IAX should work if you register to one of the static hostnames
Could you please advise which of the host names are static.
I am offered
SIP Hostname: sip1.nsw.telecube.com.au (65.12% capacity)
SIP Hostname: sip2.nsw.telecube.com.au (61.44% capacity)
SIP Hostname: sip1.vic.telecube.com.au (55.76% capacity
I tried the sip1.vic host but zoiper failed to register.
I recall that last year when it worked there was an IAX option when I set up the extension but I no longer seem to have that option.
Do I still need to select IAX in the extension setup?
Thanks,
S.
Also, does Telecube support Grandstream phones ?
You can use any voip phones with our service.
I use Grandstream GXP-2000 phones with no problems at all.
You can use any voip phones with our service.
I use Grandstream GXP-2000 phones with no problems at all.
Thanks guys � just bought a GXP-2160 :)
I recall that last year when it worked there was an IAX option when I set up the extension but I no longer seem to have that option.
I seem to remember having to set that option previously as well
It doesn't appear to exist at the moment
I haven't been able to get iax working either,
I seem to remember having to set that option previously as well
It doesn't appear to exist at the moment
I haven't been able to get iax working either,
Thanks for the confirmation.
It has been broken for so long my memory is getting a bit dim with how it worked.
I use IAX when I am overseas and a hotel blocks SIP on their Wi-Fi in which case IAX often gets through.
I am off again next week and would like to find another solution if anyone has any recommendations.
Thanks,
S.
Just another +1 for IAX support.
As mentioned above, it's invaluable when traveling abroad and finding oneself behind restrictive firewalls. Much better than SIP/RTP at getting through NAT as well.
IAX plus iLBC plus Zoiper is the magic solution, providing iLBC is set up correctly.
IAX plus iLBC plus Zoiper is the magic solution, providing iLBC is set up correctly.
Yes, i think it would be a good thing if Telecube got that working properly. That combination on Telecube worked poorly previously and I gave up on it.
This is the default position. TC want to talk to you on a Aus mobile number just to check you are a legit customer (prevent fraudsters using your account).
Shoot support an email along with a contact mobile numbe. You only need to do it once.
Thank you Pedrov.
I didn't know this. I will shoot them an email.
You need to enable international calling in the preferences section in the portal
Thanks mate.
Just updated my preferences.
TC want to talk to you on a Aus mobile number
I didn't know this. I will shoot them an email.
You don't have to do this any more .. just log into the portal and go to the Preferences section and enable international calling there.
That combination on Telecube worked poorly previously and I gave up on it.
For me, when it was working, it worked like a dream and since others are looking for its return, it must have worked for them as well.
Yes, i think it would be a good thing if Telecube got that working properly. That combination on Telecube worked poorly previously and I gave up on it.
I used IAX and Zoiper but without iLBC without much problem.
Was it the IAX or iLBC that was causing the problems?
S.
I am off again next week and would like to find another solution if anyone has any recommendations.
VPN software would be useful to have. Perhaps one day, there will be Telecube "calling card" (call thru) access via their own access numbers and importantly via the numerous SIP Broker gateways around the place. ClickNCallNow does.
There's Skype too (shudder). ippi.fr offers a Skype to SIP gateway which is useful for any provider that allows calling card access via (anonymous) SIP.
Was it the IAX or iLBC that was causing the problems?
iLBC over SIP has been problematic. Do you still need that ticket, John? I haven't got around to retesting and submitting one.
Was it the IAX or iLBC that was causing the problems?
I suggest you read the discussion either side of this post /forum-replies.cfm?t=2450974&p=17#r326.
Murray from Telecube and I did some testing back in November 2015 using Android Zoiper iLBC with IAX and SIP and we found the quality was unsatisfactory. I forget the details now, but I do remember Murray said he would hand the problem over to John for his attention. I thought there was an incompatibility between Telecube's and Zoiper's implementation of iLBC. Murray and I both observed there were problems with a registered IAX extension on Zoiper dialing to a SIP handset. However, as I said, I can't remember the details now and I've not tried to use Zoiper again.
Murray from Telecube and I did some testing back in November 2015 using Android Zoiper iLBC with IAX and SIP and we found the quality was unsatisfactory. I forget the details now, but I do remember Murray said he would hand the problem over to John for his attention. I thought there was an incompatibility between Telecube's and Zoiper's implementation of iLBC. Murray and I both observed there were problems with a registered IAX extension on Zoiper dialing to a SIP handset. However, as I said, I can't remember the details now and I've not tried to use Zoiper again.
Thanks for that but it highlights that your problem is iLBC whereas I want IAX with any working codec.
I just want IAX to work.
Finite State Machine writes...
VPN software would be useful to have
Often if SIP is blocked I find VPN (PIA) is blocked also. I used Pennytel over PIA but I have found Telecube does not work even in Australia.
Perhaps one day, there will be Telecube "calling card" (call thru) access via their own access numbers and importantly via the numerous SIP Broker gateways around the place. ClickNCallNow does.
These days I pay for a service to make phone calls and I don't want to spend the time experimenting with stuff like SIPbroker, calling cards and especially Skype.
IAX worked until mid November last year when it was broken when everything was changed for the great password hack. I opened a ticket but that was closed without resolution along with my other open tickets. I opened another ticket a month ago and nothing happened.
A few days ago, John posted on here that it should be working but I can't get it to work and I got no answer to my questions.
I leave in a week and now don't have any time to experiment so I will just put some more credit on my TravelSIM and get on with my life.
We will continue to maintain the Telecube account as a backup including paying the $5 per month minimum but I think it would be better if the ticket system was re-established as the priority for support calls rather than the squeaky wheels on Whirlpool.
S.
I use IAX when I am overseas and a hotel blocks SIP on their Wi-Fi in which case IAX often gets through.
I am off again next week and would like to find another solution if anyone has any recommendations.
As well as hotel WiFi, I've also been in situations overseas where SIP won't play nicely with the mobile carrier I've been using at the time. Most recently, it was Spark in NZ. Even in Australia, I only get one-way audio if I use the Cloud Softphone when using Optus (Zoiper is OK, though.)
Since IAX stopped working with Telecube, my fallback when SIP won't work at all is to use IAX and FaktorTel. The rates aren't as good as Telecube, but at least I can count on it.
I'm also waiting for IAX support to resume, for all the reasons mentioned above.
John,
Is there an intercom function/extension on the Telecube system ? I'm trying to find something to implement between our two phones.
Is there an intercom function/extension on the Telecube system ? I'm trying to find something to implement between our two phones.
Calls between extensions are free, as long as you call the extension number and not a DID.
Calls between extensions are free, as long as you call the extension number and not a DID.
And you can create alias shortcuts to making dialling different extensions internally on the account easy.
/forum-replies.cfm?t=2323366&p=69r1376
I've already got that setup, i'm looking more at the intercom function that most IP phones can do (i.e. it emits a short beep then opens the other extension's microphone on speaker so you can chat to them without them picking up)
it emits a short beep then opens the other extension's microphone on speaker so you can chat to them without them picking up
I think Yealinks have 'Internal auto answer' feature, not guaranteed to work in every situation though. Other phones may have similar functionality.
Normally this is done through Alert-Info, but you'll need control over your PBX's dial plan to do that.
+1 for IAX. I use it when traveling OS and when out and about for international calls.
I think Yealinks have 'Internal auto answer' feature, not guaranteed to work in every situation though. Other phones may have similar functionality.
Normally this is done through Alert-Info, but you'll need control over your PBX's dial plan to do that.
I've got a Yealink T26 and also a Grandstream GXP2160...i'll have a look at the settings tomorrow.
Does updating voicemail recordings by dial-in work? Previously I was told you could only do it by uploading a file, but I'd love my users to be able to manage this themselves.
John,
I think I might've mentioned it to you recently, but wanted to check something:
I'm research the possibility of locally hosting our PBX (using incoming phone lines as opposed the net) but still having a fallback if need be.
What happens with the 1300 number in terms of becoming a PSTN based number as opposed to IP ? We have an (03) local number which has been (until we get cable on Friday) the backbone for our ADSL but seeing as it frees up, I don't want to give out a new number again, and our 1300 number is a good one :)
Telecube not registering..... all other providers registered.
What happens with the 1300 number in terms of becoming a PSTN based number as opposed to IP ?
Different rates apply for PSTN termination.
Local 4.9c
National 7.9c
Mobile 9.9c
Telecube not registering..... all other providers registered
Hmmm, seems to have come good. No idea!
Ah so I can retain the phone number, point it towards our landline .... then does that mean you'd either answer it on a traditional phone (if you chose) or use a local IP PBX ?
I'm curious about the 2 line FXO machines, meaning you'd have to have a second line for incoming (or two as outgoing, depending on who made the calls) � how would that work with the 1 x 1300 number?
If you're planning to drop the phone line, you should be able to port the number to Telecube after you've cancelled the ADSL service and terminate that to your IP PBX; then you'll pay the same VoIP rate as you are now.
I'm actually thinking of keeping the phone line as it's part of the package we're getting anyway and could be a good way of making personal calls (keep the business on the 1300 number) OR use it as the incoming for Telecube termination to the local IP PBX.
We're basically trying to get a better speed/less jitters/delayed/ordinary calls by taking that part of the infrastructure in house if we can.
Also, for a call terminated to PSTN instead of a VOIP/net connection, is it less prone to dropouts/jitters/delays because it's not going through another process in your local net connection ? I'm curious if the quality to PSTN would be better.
Is there such a function as roll-over that can be implemented so that if the first PSTN termination is busy that it rings the 2nd PSTN (if we had one in place) but still under the same 1300 number ?
Does anyone know how to configure call parking with a yealink T4x series..just want a dss key to dial #5....
Don't worry � found it � Set the DSS Key type to DTMF
I'm actually thinking of keeping the phone line as it's part of the package
Use it as a backup and for any free call inclusions. Just remember than cometh the NBN, POTS will be no more.
Also, for a call terminated to PSTN instead of a VOIP/net connection, is it less prone to dropouts/jitters/delays because it's not going through another process in your local net connection ?
Yes, YMMV. If Telecube is hosting the 1300 number there is still an IP connection between its and its upstream providers but it should have good QOS.
Is there such a function as roll-over that can be implemented so that if the first PSTN termination is busy that it rings the 2nd PSTN (if we had one in place) but still under the same 1300 number ?
A hunt group from your PSTN provider is the traditional way to do that and will also work with your local number too. Setting a forward on busy on the first line is way to do it without subscribing to a service with each call being charged if not included in a plan. I don't know whether Telecube forwarding can be set up to try another number when the first is busy.
Hi,
Is something wrong with the site � trying to get past the one time password page, and after submitting code it just says:
The manage.telecube.com.au page isn�t working manage.telecube.com.au is currently unable to handle this request.
HTTP ERROR 500
Also, this OTP thing is getting really frustrating, I don't know why you couldn't just use Google 2FA app, and then maybe remember my computer for 30 days or something....I've got enough emails smashing my inbox as it is
Also, this OTP thing is getting really frustrating, I don't know why you couldn't just use Google 2FA app
+1
You can authorise your browser for seven days.
Thanks, I didn't realise that � for anyone else's benefit it's under profile -> manage OTP. That will hopefully help reduce the frustration
I've received two invoices for July, same invoice numbers, but different $ amounts?
Which is the correct one?
I've received two invoices for July, same invoice numbers, but different $ amounts?
Same here, I assumed the first was a mistake and the second was the correct one
I noticed that the August invoice was a lot more convoluted than the July one. Whereas the July one was simple to follow, the August invoice had a number of extra entries such as adjustment debits and credits that didn't appear on the July invoice. I think it would be better to keep the invoices as simple as possible.
Any outages at the moment, I seem to have lost registration?
Yes, down for me in Sydney on TPG ADSL.
Yes, down for me in Sydney.
Brisbane here, thanks for the reply � appears I'm not alone.
Same, lost registration...
Showing as registered on my Gigaset using Telstra Business ADSL in NSW.
Down/unregistered here in Adelaide as well.
Showing as registered on my Gigaset
But can you make a successful outgoing test call?
On Siptalk (but the same platform) and dialing 99599 had it give me account balance.
I had been using sip.telecube.net.au and until today used it fine (pretty sure it's the only address I used on Telecube). I then logged in and saw three different addresses. Choosing one of these worked.
"Select a hostname from below that is closest to you and at lowest capacity.
If you use BLF or Call Parking/Pickup you must use the same hostname in all devices.
SIP Hostname: sip1.nsw.telecube.com.au (51.84% capacity)
SIP Hostname: sip2.nsw.telecube.com.au (50.80% capacity)
SIP Hostname: sip1.vic.telecube.com.au (43.48% capacity)"
On Siptalk (but the same platform) and dialing 99599 had it give me account balance.
Ah fair enough � thanks.
Looks like my ext has just come up and registered ok.
Down here as well in WA.. Gigaset showing not registered
Just adding to the group... SIP Registrations Down in VIC � TPG EFM.
Tracerts are fine, receiving 503 Service Unavailable/Register from local since 7:48PM, we were using sip.telecube.net.au (and receiving a different error 408s for this).
Incoming calls to extension which are sent directly to our IP address/trunk are working though
Clearly John must be messing around with the DNS config for telecube.net.au
Only one of my Telecube extensions has been configured with sip.telecube.net.au but since ~7:40pm last night registration has been failing with DNS resolution failures except for a few short periods of time.
All extension registrations to sip.telecube.com.au have not been affected.
So if you've had registration problems since last night change your registration server address to one of the telecube.com.au addresses.
All extensions down, time for me to change registration to ... ~sip.telecube.com.au
SIP Hostname: sip1.nsw.telecube.com.au (55.68% capacity)
SIP Hostname: sip2.nsw.telecube.com.au (53.84% capacity)
SIP Hostname: sip1.vic.telecube.com.au (47.76% capacity)
Just noticed that routes from NSW Telstra to the NSW servers no longer go through Melbourne. Thanks John.
Oddly. the Victorian server is 3 hops closer to me.
time for me to change registration to ... ~sip.telecube.com.au
I would have thought that amongst all the recent turmoil(s) that John could have found time to send his loyal customers ... an email advising the changes to sip registration?
I would have thought that amongst all the recent turmoil(s) that John could have found time to send his loyal customers ... an email advising the changes to sip registration?
This update was emailed to me at the time of the event � http://telecube.io/event-detail.php?id=8
an email advising the changes to sip registration
One of the SIP proxy services and the sip.telecube.net.au hostname is having intermittent issues and changing to one of the static hostnames will resolve your registration problems.
I would like to remind people to subscribe to the status alerts please if you haven't already.
I would like to remind people to subscribe to the status alerts please if you haven't already.
So ... I'm a slow learner.
Thanks John. :)
Brisbane � down for 15mins now, drop outs before that.
Seems like only Cloud Softphone is affected. All our other extensions on the vic domain name are okay.
I changed to sip2.nsw.telecube.com.au because of issues, but the regestration keeps dropping and then re establishing...it was in a pattern of every 10 or 20 mins exactly....too early to see a pattern at the moment
Can't dial in to PBX on the extension even though it says it's registered...tried from external line and from another telecube extension...
update: seems to be a strange issue of co-incidence...looks like the PBX itself may be playing up...just added the same sip account to one of the yealink phones instead of the pbx and calling in and out worked like a charm.
Here we go again... just lost registration.
[Edit]
....and its back up again!
..changing to one of the static hostnames will resolve your registration problems.
I wish I could � I just lobbed in France for three weeks and now I've got no incoming calls...
One of the SIP proxy services and the sip.telecube.net.au hostname is having intermittent issues and changing to one of the static hostnames will resolve your registration problems.
I'm on a boat in Banburg in Germany and I have been asked to fix this for the office.
I can do it from here but I have no idea what a proxy service with a static host name is.
Any suggestions greatly appreciated and I realize it is 0220 there.
S.
I can do it from here but I have no idea what a proxy service with a static host name is.
Try one of the three mentioned in this post: whrl.pl/ReH0Da
I would like to remind people to subscribe to the status alerts please if you haven't already.
Maybe a reminder on the phone bill would be a good idea?
I've been a telecube customer for a while now and didn't know that subscribing was a requirement if I wanted to know when my service was down.
Maybe a reminder on the phone bill would be a good idea?
That's a good idea I will add a link in the email.
I am really surprised that something like sip.telecube.net.au and sip.telecube.com.au isn't being used as a failover hostname so that it can direct the connection to any one of the 3 servers based on certain criteria like server load or whichever server is appearing up/online.
Something like this would be far more beneficial to everyone instead of telling them to use just one of the 3 servers.
At least I know this is how a lot of the VOIP networks are setup when I was working in America sometime ago.
If this isn't something that happens in Australia, it speaks for itself that Australia is quite a long way behind the times.
Anyone else having issues with call transfers? I've been having them for months now
If i reboot my Cyberoam router and power-cycle all the handsets, calls can be transferred fine within our groups for a day or so, but after a variable time, they will start to fail, and instead get "forbidden" messages when pressing the transfer button, or the transfer calls get bounced back to the initiating handset.
We're using Yealink handsets � they worked fine when TC was using sip.telecube.net.au, but this issue seems to have appeared after we migrated all to sip1.nsw.telecube.com.au and sip2.nsw.telecube.com.au. I've emailed support, no response though.
Can we still use the old sip.telecube.net.au server address?
Does it matter if we list sip1.nsw.telecube.com.au and sip2.nsw.telecube.com.au as SIP server 1 and Sip Server 2 in Yealink configs? Or should we only use one of them? I ask because I am wondering if the transfers fail because some of the phones have failed-over from sip1 to sip2, and transfers might only work when all are on the same sip server.
I'm on a boat in Banburg in Germany and I have been asked to fix this for the office.
I can do it from here but I have no idea what a proxy service with a static host name is.
Any suggestions greatly appreciated and I realize it is 0220 there.
24 hours later and no response.
I guess John is too busy to help me.
I guess John is too busy to help me.
Ozimarco linked you to the answer 13 minutes after you posted;
Try one of the three mentioned in this post: https://whrl.pl/ReH0Da
24 hours later and no response.
LOL, don't you read my posts, sweetpea? I am not on your blacklist, am I? :)
I have no idea what a proxy service with a static host name is.
If you're having intermittent registration issues DNS might be your problem. If your nominated DNS server is unresponsive or slow it may not be resolving the SIP server/proxy.
Try using one of the following IP addresses instead...
103.193.167.161 � sip.telecube.com.au
103.193.166.33 � sip1.nsw.telecube.com.au (AKA sip1.telecube.com.au)
103.193.166.36 � sip2.nsw.telecube.com.au (AKA sip2.telecube.com.au)
103.193.167.53 � sip1.vic.telecube.com.au (AKA sip3.telecube.com.au)
Ozimarco linked you to the answer 13 minutes after you posted;
I didn't ask Ozimarco and I don't consider his response to be helpful. I still don't know what a proxy service with a static hostname is.
Try one of the three mentioned in this post: https://whrl.pl/ReH0Da
Again, being referred to a post by someone non authoritive is not the answer I was seeking.
I've solved the problem by changing to another VSP until I get back to Australia and I'll work it out then.
I'm happy to pay for a stable and reliable service and I don't expect to get something for nothing and I will pay for professional support.
I'll go back to cruising the rivers of Europe and enjoying my holiday.
S.
I'm happy to pay for a stable and reliable service and I don't expect to get something for nothing and I will pay for professional support.
Well did you email support? Whirlpool is not an offical support method � John may reply but if you want an official response you should email support not post on a public forum.
I still don't know what a proxy service with a static hostname is.
In your Telecube registration setup, replace the sip server address with the alternate server address (pick one of 3), and hit save.
Well did you email support? Whirlpool is not an offical support method � John may reply but if you want an official response you should email support not post on a public forum.
Sadly my history of successful emailed support requests is not good. The tickets are either closed without resolution or are just not resolved even after nearly a year.
I am on holidays on the other side of the world and someone from my office sent me an email quoting a problem that was mentioned on whirlpool.
I quoted John and asked for clarification and Ozimarco linked to another post that did not answer my question to John. The next day John referred me to the same non authoritative post that still didn't answer my question and Ozimarco seemed miffed that I ignored him.
John has nearly 3800 post on whirlpool and I think I have probably read most of them. I can even point to the posts where he says you will get better response to your logged faults by notifying him here so it certainly does seem to be the main way to get support even if it is not official.
There is apparently no useful documentation any more and none of the official documentation seems to match reality any more.
As I said I have circumvented the problem by changing VSP for the time being. In the meantime I am on a boat on the Main-Danube canal and watching the country side drift by and traveling up and down 25 metres in the locks is far more pleasant
It is night here so I wish you goodnight.
S.
Any suggestions greatly appreciated and I realize it is 0220 there.
then ..
Try one of the three mentioned in this post: whrl.pl/ReH0Da
so ..
I didn't ask Ozimarco and I don't consider his response to be helpful.
doesn't really seem fair.
I can do it from here but I have no idea what a proxy service with a static host name is.
doesn't really seem fair
What doesn't seem fair is that you don't answer a simple question and tell me what a proxy service with a static host name is.
At this stage I would be happy if Ozi Marco could answer the question as well.
At the moment everyone is on their high horse picking on me and ignoring the fact that I asked a simple question about a statement that John made in a message and two days later have not received an answer.
I get that this is Whirlpool but this is where John made the statement.
S.
What doesn't seem fair is that you don't answer a simple question and tell me what a proxy service with a static host name is.
Where you currently have sip.telecube.net.au change it to one of the options listed on http://telecube.io/event-detail.php?id=8
What doesn't seem fair is that you don't answer a simple question
Your question was answered multiple times, there's nothing I can add.
If you haven't already please subscribe to network notifications at www.telecube.io
Ozimarco seemed miffed that I ignored him.
Not at all. It was meant as a bit of light-hearted banter, hence the LOL and the smiley. I'm sorry to hear my post was not helpful to you, even though, if you had passed on the advice to the person who had emailed you, their problem would have been solved.
In the meantime, enjoy your cruise on the Danube � it's something I want to do, too, probably next year � and I'll try again next time, hopefully with more luck.
At the moment everyone is on their high horse picking on me and ignoring the fact that I asked a simple question about a statement that John made in a message and two days later have not received an answer.
O.K. everyone..... let's all take a chill pill.
@Sweetpea � you posted:
I'm on a boat in Banburg in Germany and I have been asked to fix this for the office.
I can do it from here but I have no idea what a proxy service with a static host name is.
Any suggestions greatly appreciated and I realize it is 0220 there.
To be fair, you made a statement (not a question) indicating you don't know what a proxy service with a static host name is. Perhaps you inadvertently omitted a ? mark which the statement would then read as a question. Hence all the subsequent responses failed to meet/address your expectations.
To further complicate matters, you also stated "any suggestions greatly appreciated" which Ozimarco and others put forward to assist you in overcoming your issue.
Again, the context of your post didn't indicate that you wanted a response specifically from John.
Judging by all the various comments and responses, it appears that there has been a breakdown in communications between all involved.
Even I'm prepared to admit that I'm experiencing issues trying to get my head around the whole concept of the various static proxy services.
So, to put this little anomaly to rest:
@John � for the benefit of Sweetpea, myself & anyone else that was afraid to ask, can you put up a post that explains the proxy service concept as it pertains to Telecube, how it benefits the customer and maybe include a sample scenario that showcases the concept?
can you put up a post that explains the proxy service concept as it pertains to Telecube
I had planned a set of sip registration proxy servers behind a single hostname but it didn't scale as expected so I have reverted to single hostnames statically mapped to registration servers, currently 2 in Sydney and 1 in Melbourne.
sip1.nsw.telecube.com.au
sip2
sip1.vic.telecube.com.au
I am planning to further simplify it to sip1.telecube.com.au, sip2.telecube.com.au, sip3.telecube.com.au .. etc
At the moment we are having issues with the registration proxy that sip.telecube.net.au is mapped to and people should change to one of the 3 static hostnames
get my head around the whole concept of the various static proxy services.
Doesn't it just boil down to an attempt was made to automatically load balance between various servers.
It failed and the proxies/servers were exposed to the public for them to determine (based on load percentages) which to use.
Edit: Beaten. Must increase typing speed.
Must increase typing speed.
I'll give you a tip. When writing a long reply, write a few words or short sentence, then post. Now edit your post. This guarantees that your post will appear before posts from others who do not do it this way.
I am planning to further simplify it to sip1.telecube.com.au, sip2.telecube.com.au, sip3.telecube.com.au .. etc
I think the current is better as it allows users to pick the closest server.
Does Cloud Softphone on iOS use sip.telecube.net.au? I was still using it via Siptalk but it kept having registration issues that only started in the last few days. I know its unsupported more curious. I switched to Bria (already had a licence from using it on MNF years ago) and the problem disappeared.
Also to fix it for people using the old domain why don't you just put multiple A records and let round robin DNS select servers � will make things works for people on the old domain.
Also to fix it for people using the old domain why don't you just put multiple A records and let round robin DNS select servers � will make things works for people on the old domain.
Good thinking .. done. It will break BLF and call parking though if anyone is using it with the cloud soft phone.
Good thinking .. done. It will break BLF and call parking though if anyone is using it with the cloud soft phone.
Looks good from here:
;; ANSWER SECTION:
sip.telecube.net.au. 60 IN A 103.193.166.36
sip.telecube.net.au. 60 IN A 103.193.166.33
sip.telecube.net.au. 60 IN A 103.193.167.53
Does it matter if we list sip1.nsw.telecube.com.au and sip2.nsw.telecube.com.au as SIP server 1 and Sip Server 2 in Yealink configs?
I had issues in my last rollout using different sip servers, both with blf and with transfers. I guess they must not announce their presence properly between different registration servers?
Anyway, I'd recommend keeping them all the same
deleted � misread post
I think the current is better as it allows users to pick the closest server.
+1. Please retain the state specific domain names John for those of us who wish to differentiate. For me the difference atm is about 10ms.
Also to fix it for people using the old domain why don't you just put multiple A records and let round robin DNS select servers � will make things works for people on the old domain.
Good thinking .. done.
I don't think it should be entirely round robin. They should by ordered by closest/least loaded server. For those clients that support it, providing a SRV record would be better. Your current loadings could drive the SRV weights e.g.
Assume server loads of 60% and 30% the SRV record for sip.nsw.telecube.com.au could return:
_sip._udp.sip.nsw.telecube.com.au service = 1 75 5060 sip1.nsw.telecube.com.au.
_sip._udp.sip.nsw.telecube.com.au service = 1 150 5060 sip2.nsw.telecube.com.au.
_sip._udp.sip.nsw.telecube.com.au service = 2 100 5060 sip.vic.telecube.com.au.
Connections to the NSW servers are tried first. Connections will be allocated between sip2.nsw and sip1.nsw on a 2:1 basis.
The weight calculation I have used assigns a weight that is inversely proportional to the relative current load of the server:
To calculate W1, W2 (75, 150) of the 2 Sydney servers
Wn = round ((A / Ln) * S)
A = ( L1 + L2 ... + Ln)/n
where:
n is the number of servers: 2
Ln is the load on server n (must be non-zero!)
A is the average load across servers
S is scaling factor for the required digits of precision. Here S = 100 for a minimum precision of 1 digit at 1% load difference.
gives:
Wn the relative weight for that server (amongst its peers of equal preference).
If the load updates are low or the cache long lived consideration should be given to dampening larger weights e.g.
Wn = round(log(A/Ln*10)*10)
Finite State Machine writes...
I don't think it should be entirely round robin
Actually this was only for the sip.telecube.net.au hostname
Good thing I didn't spend long writing my post :-)
Finite State Machine writes...
Good thing I didn't spend long writing my post :-)
Yeh .. lucky ;-)
I've got a question that I don't think is worth submitting a ticket for...
We've got Yealink T23G handsets, and occasionally when transferring a call to another extension it rings on the destination, but the source says "Transfer failed" and the dialer is returned to source as if they are just on hold. The destination phone continues ringing until you pick it up, but there's no one there.
What is the most likely cause of this? It works 99% of the time, but in that 1% of the time transfers don't work from the source to any other extension.
EDIT: The annoying part about it not working, is that it doesn't work at all no matter how many times you try until "some time later". It doesn't seem to help powering the handset off and back on again. Sometimes it's only a matter of a minute or two later and it works fine again.
What is the most likely cause of this? It works 99% of the time, but in that 1% of the time transfers don't work from the source to any other extension.
We have the same issues with the Yealink W52 handsets. Seems like a random enough occurrence to safely ignore.
Finite State Machine writes...
Good thing I didn't spend long writing my post :-)
Well, at least you managed to seriously impress me! :-)
Không có nhận xét nào:
Đăng nhận xét