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