Continues from: /forum-replies.cfm?t=2323366&p=-1#bottom
This is a special offer for WP members and guests
Start your free trial first: https://www.telecube.com.au/signup/?fr=wp
Important
You must have an active Telecube account already and be logged in before clicking on the links below.
Then after you have an account created you can choose between ..
Timed National Calls
Calls to fixed lines 1.87c per minute
Calls to mobiles: 7.9c per minute
Offer Code: WP_OFFER_1_TIMED
Redeem at: https://manage.telecube.com.a
or ..
Untimed National Calls
Calls to fixed lines 10c per call
Calls to mobiles: 7.9c per minute
Offer Code: WP_OFFER_2_UNTIMED
Redeem at: https://manage.telecube.com.a
Billed in 1sec increments and no flagfall
Prices incGST
Once the rate is applied to your account it can't be changed for 14 days. If you want to change from timed to untimed (or vice versa) you will need to wait for 14 days to pass before being able to update your account to a different rate.
You must have made at least one topup payment into your account to be eligible for the special rates.
There is a free trial offer available to all new customers which will allow you to login and look around the portal and add voip extensions to your account. You will be able to make free calls between extensions to test the system and call quality.
No credit card needed for signup.
You can make a topup at any time and use the special offer codes above to set your account to the special rate of your choice.
Limitations
Customers on special rates aren't eligible for bonus rewards.
Monthly Line Charges or Setup Costs
None, nil, zip, nada, nothing, zero.. you get the idea.. :-)
Allowed Concurrent Calls
No limit on concurrent calls.
Go Mobile
We have a free mobile app for IOS or Android with G729 included and '1 Click' setup using QR Code scanner built into the app.
Once you have added voip extensions and activated one, the manage screen for an active extension will give you all the details, including the QR Code and where to get the app.
Local Number Porting
First number FREE for WP members � normally $33
Free port limited to 1 number per account .. extra numbers $11 each
Other porting costs
Cat A � Simple Ports free for first number then $11
Cat C � Complex Ports (1 � 5 Numbers) $55
Cat C � Complex Ports (6 � 100 Numbers) $110
Cat C � Complex Ports (100+ Numbers) $165
DID Costs
55c/month for WP members � normally $2.95/month
Applies to DIDs ported in or allocated by us
Porting Out
You can port DIDs out without penalty. Applies to DIDs we allocate to you or that you port in.
Fax to Email & Email to Fax
Sending Faxes = 4.9c per minute � no monthly charges or minimums
Receiving Faxes = 4.9c per minute � requires a did at 55c per month
Offer Expiry
Until further notice
Where is it?
Glad you asked.. :-)
www.telecube.com.au/voip/
Have Questions or Need Help?
Phone: 132823
Email: support@telecube.com.au
Skype: telecube.com.au
Voip: 1000 or 1001000
Already a Telecube customer?
Existing customers are also eligible for the special rates.
Monitoring Quality and Stability
This is all Australian based routes on tier 1 carrier networks. If you have any quality issues please let me know.
It's my expectation that this service should be as good as any other voip provider in the market and better than most.
If it's not please do let me know so we can work on the problem.
Telecube Special Numbers
- 99299 � Echo test
- Test the latency between your device and our system
- 99599 � Balance check
- Get the balance of your account from a voip extension
["For example, a Telstra IP address pool it allocates from is 101.160.0.0/11 which is some 2 million addresses in the range 101.160.0.0 � 101.191.255.255. Allowing registrations only from that pool for an extension will be fine if you are dynamically allocated one of these addresses.
I've looked at some of the extensions that are dynamically allocated IPs and I think it will be reasonably effective to limit access to ranges or lists of ranges.
"]
Unfortunately, that is not the only allocation Telstra use.
My Boost mobile which is on the Telstra network currently gets an IP of 1.136.n.n or 1.152.n.n and
My Bigpond cable currently has an IP of 121.210.n.n
How would one go about finding the definitive range just for Telstra?
It still doesn't solve the problem when travelling.
S.
DID Costs
55c/month for WP members � normally $2.95/month
Is this my correct understanding, for a hosted PBX with 10 DIDs, the ONLY recurring cost will be $5.50pm plus calls?
I was just looking at your service as an backup contingency for an ageing asterisk PBX box, and the origanal supplier has closed, just created a login and having a look at the features to see how closely it meets our needs. I may just toss the old box and jump on board with you.
I assume I shall have no issues with connection our SNOM 300 handsets. But your Wiki does say Music on Hold is not available, is that still correct? Thats a little important for us, because we did pay a lot of money for our 'jingle'
My Boost mobile which is on the Telstra network currently gets an IP of 1.136.n.n or 1.152.n.n and
These are all covered in a single allocation � from a whois:
inetnum: 1.128.0.0 � 1.159.255.255
netname: TELSTRAINTERNET49-AU
descr: Telstra
Which can be added to an ACL as 1.128.0.0/11
Is this my correct understanding, for a hosted PBX with 10 DIDs, the ONLY recurring cost will be $5.50pm plus calls?
That's correct.
I assume I shall have no issues with connection our SNOM 300 handsets.
The SNOM handsets should be fine is they are SIP capable.
But your Wiki does say Music on Hold is not available
Custom MOH isn't manageable in the portal just yet (coming soon) however I can manually include it for your extensions if you send a request through to support.
Brilliant John, that is handy to know.
One more complicated question, and its harder to understand the control panel without real working accounts, is that I have SIP providers all over the place. I dont really need them all but I do have some New Zealand DIDS which you dont do, because I have an office in NZ, and need to retain those, and handle them somewhat seperately in the routing.
Can a SIP account for another provider be hosted in your PBX?
If not i guess I can forward them to a dedicated Telecube DID, and set up the routing accordingly?
I thought I better support some other Australian VoIP provider than my regular, so I signed up with you guys.
You might want to poke your webdev though, the signup page is using col-md- instead of col-sm to switch to mobile view. This means that I see stuff like this:
http://i.imgur.com/wXrrgs8.png
We've been doing a LOT of work with mobile-friendly stuff in FreePBX 13, so this just jumped out at me 8-)
Do you guys have a pay-as-you-go account?
Edit: Duh. They're all pay-as-you-go. Ignore me.
Do you guys have a pay-as-you-go account?
That is exactly what the WP special is � see rates at the start of the thread.
I dont really need them all but I do have some New Zealand DIDS which you dont do
Actually we do, just don't advertise it. If you want to email support we can let you know pricing.
Can a SIP account for another provider be hosted in your PBX?
Hmm .. if you mean register to an extension on another provider .. no. If you mean DIDs on another provider ringing in to your Telecube extensions .. possibly.
I'm not against adding the functionality but would need to understand how it would need to be setup to be useful.
Hmm .. if you mean register to an extension on another provider .. no. If you mean DIDs on another provider ringing in to your Telecube extensions .. possibly.
I'm not against adding the functionality but would need to understand how it would need to be setup to be useful.
Forgive my naiveness. What I know about asterisk is limited to the proprietry interface my supplier built for his small business, and I am pretty adept at configuring that, but I assume its all standard features � seems similar to yours for the most part. Except for the section called "Lines"
What I have now 3 Pots lines (digium card obviously cant be used with your solution), and 4 SIP accounts across a couple of providers, and these are configured each as an indivudual Line. Each line has its incoming routing individually configured, though some are common. I have three IVRS, and a couple of Ring Groups that the lines get routed too, depending on Day/night/Line
One example of what I need to customise, is we want our native NZ speakers to answer the NZ calls when our NZ outpost is unnattended which is often (apparently they take 4 hours off for lunch/nap). So these calls overflow into our Pabx, get routed differently to our normal calls, and we flash ALERT KIWI on the handset.
Does all this make sense?
Does all this make sense?
Ok, incoming calls can be customised to your needs without having to connect with your existing voip provider. You can either port the numbers in to us or forward them to our network through your current provider.
Assuming you would have a specific IVR setup on our network for the NZ calls you could have the calls routed into separate extensions on the handsets and label the extensions so the operator would know where the call was coming from.
I presume the IP alert is based on Contact information rather than from received = .
I have a dynamic IP but use local IP in the Contact and apart from the initial batch of alerts and the fact my public IP has changed several times no further alerts. My alert notification is on.
I don't want to change this but was surprised others are getting heaps of alerts.
Info
I've added IP based ACL to the platform. You can control it from the manage screen of a voip extension.
It will take multiple IP addresses or ranges in the format:
192.168.0.123
192.168.0.0/24
192.168.0.0/255.255.255.0
You can add multiple addresses/ranges in the text area one per line.
To turn it on add an ip and update .. to turn it off delete the ip and update an empty text box.
It can take a minute or 2 to propagate changes to the registration servers.
Hi John, is the IAX server working? I had 2 asterisk trunks registered yesterday but neither is registered tonight.
Hi John, is the IAX server working?
It appears to be, I can see extensions registered on it and I just registered Zoiper to an IAX ext and called out ok.
Just want to give a quick testimonial for Telecube, having used them for quite a while at a previous company I worked for, I can't recommend them enough. Good phone voice quality, great support, very happy customer.
I've added IP based ACL to the platform
Thanks John, have you also listed the ISP's ranges? Some people might not know what /10 and /24 is. Love the converter, good idea :)
iiNet/Internode is: 121.44.0.0/15
iiNet/Internode is: 121.44.0.0/15
iiNet and Internode have quite a few IP ranges. One of our iiNet IP adresses is 121.127.*.*, and we've got numerous Internode IP addresses in the 203.206.*.* range.
iiNet and Internode have quite a few IP ranges.
There's a big list here:
https://kb.cadzow.com.a
There's a big list here:
Plus how accurate is it. I have TPG in the 202.161.x.x range and no 202.x.x.x is included in the list.
Plus how accurate is it. I have TPG in the 202.161.x.x range and no 202.x.x.x is included in the list.
http://bgp.he.net/AS7545#_prefixes
That would be a much more accurate list for TPG IMO.
Kind Regards,
Dean
That would be a much more accurate list for TPG IMO.
Yes a much more comprehensive list but still a bit out of date as mine is 202.161.127.x
I think it demonstrates that blocking on IP will end up blocking genuine dynamic ips or be so coarse as to be useless. Only accepting traffic from your vsp on that trunk and preventing sip uri calls seems best to me plus not allowing auto top-up.
You don't need to allow every IP an ISP has, you'll find most major ISP with lots of IPs will be allocating ranges by location.
Looking at extensions regularly used on the road with Telstra we are seeing they are allocated IPs from 2 or 3 ranges.
I am building an alert now that will email you when an IP is blocked due to an ACL and it will also lookup the ISP IP range, convert it to a CIDR address for you and include it in the email alert so you can have a quick look at the carrier, confirm it's yours and add it to your ACL rule.
We have a couple of weeks of logging IP addresses and even extensions registering from overseas with dynamic IPs are boiling down to 1 or 2 CIDR ranges.
I'll be adding a tool to the ACL manage screen shortly that will scan the IP addresses that an extension has connected from, run a whois and get the ranges, convert them to CIDR format that you can just copy/paste into the ACL list text field.
Thanks John, have you also listed the ISP's ranges?
I'm going to add a tool there that you can just add a single IP address into and it will run a whois, get the range and display it for you as well as present the CIDR format that can be added to the list.
Love the converter, good idea :)
:-)
Yes a much more comprehensive list but still a bit out of date as mine is 202.161.127.x
It's there. Your IP address falls under their /18 � 202.161.64.0/18
BGP.he.net generally is not 'out of date' because it's listing every IP TPG are announcing under their AS. If they add more addresses, HE usually picks them up within a few days.
Kind Regards,
Dean
If they add more addresses, HE usually picks them up within a few days.
Thanks for that info.
Your IP address falls under their /18 � 202.161.64.0/18
Must get brain into gear before typing.
IP to CIDR via whois tool is in the portal now too.
Hi John,
I signed up a few weeks ago and put through a request to port some DIDs over from another provider about 10 days ago. Just curious as to how long it usually takes?
Not sure if I did it properly though.. do i need a working voip service with you guys before porting or does porting automatically create the voip services.
Cheers!
Just curious as to how long it usually takes?
Have you received email communications about the ports?
do i need a working voip service with you guys before porting or does porting automatically create the voip services
You will need to manually create extensions and setup your device(s) to be able to receive calls.
Thanks John,
Have you received email communications about the ports?
Yeah I received (what seems like an automated) an email with a "confirmation that the following numbers have been loaded into our porting system" 10 days ago.
I've just created two extensions (for the two numbers I'm porting) and chucked the details into the router at home.
Cheers
Yeah I received (what seems like an automated) an email with a "confirmation that the following numbers have been loaded into our porting system" 10 days ago.
Can you reply to that please and ask for an update?
Can I forward an incoming call directly to voice mail? I tried setting forwarding to the V2E ID but no go...
Can I forward an incoming call directly to voice mail? I tried setting forwarding to the V2E ID but no go...
Yep, add a ring group and vm but no extensions and set the DID to use the ring group
that works just as I wanted.. thanks.. great product/service
I've recently ported my DID from MyNetFone to Telecube so I can finally close down my MNF account but have noticed that all incoming calls are all showing their caller ID in international format, i.e. +614xxyyyzzz. This wasn't the case on MNF and would prefer it to show as 04xxyyyzzz.
Is this something that can be configured with Telecube? If needed, I'm running a Fritz!Box 7490.
Thanks.
I submitted my engin DID for porting to Telecube on the weekend and I received an email earlier today to say I'd contacted engin customer service and would i like to provide any feedback. I assume this was Telecube?
all incoming calls are all showing their caller ID in international format, i.e. +614xxyyyzzz. This wasn't the case on MNF and would prefer it to show as 04xxyyyzzz.
I've noticed this myself, kind of makes it redundant blocking "VIC" when you place in 03 as a block in the telecube system but it comes up as 613xxxxxxx
sorry mate my bad it was misconfigured
must have hit a threshold and started getting emails about low credit. do we need to manually add credit to the TC account or I thought it was set up in such a way that it automatically took credit from a credit card once a low balance was hit.
I thought it was set up in such a way that it automatically took credit from a credit card once a low balance was hit
You can set it up that way if you want to .. have a look in Accounting > Auto Topup
I had a low credit the other day (was above $5) and manually topped up, but I've had 2 more low credit emails, even though the credit is above $15.
even though the credit is above $15.
If you think email alerts are coming out to you when they shouldn't please email support so I can have a look
Done
Responded to your email now, you need to choose 'Critical Balance Only' if you only want notifications when the balance is low.
My balance doesn't change often but don't have a need to recharge yet � is it possible to notify once a week instead of every day for users where the balance hasn't changed from the previous day?
is it possible to notify once a week instead of every day
You can set balance notifications to send once per week on a specific day and set a critical balance limit which, when reached will cause notifications to be sent daily.
Info
There's an option in the voicemail to email services now to select MP3 as the message format.
It may not yet be live for some DIDs but will be updating through the week, if you select the option now the messages will start coming through as MP3 over the next few days.
Hi John
I'm a little confused about pricing for untimed whirlpool plan
Are local/mobile numbers charged in local currency
But international charged in USD?
If so can you have int calls charged in local currency to avoid the crappy exchange rate?
Thanks
I suspect John doesn't want to take the FX risk � he will be paying his counterparty in USD... All the inter telco payments will be in USD
Confirm local calls are in AUD
If so can you have int calls charged in local currency to avoid the crappy exchange rate?
Unfortunately, the crappy exchange rate affects us all. Once calls complete you will see the actual AUD charge in the call records in the portal as well as the USD rate and the exchange rate used to calculate the call into AUD.
It's the neatest way for us to process the calls, if we need to play the FX game then we'll need to add margins on top which will bump the prices up.
ok cool
Thanks
Feature request: The ability to use an IVR as a "voicemail" in ring groups instead of being limited to actual VMs.
Not sure if there's a technical reason why this can't happen but it would be handy as we use an IVR after hours (to select two different options) and a standard VM during business hours to catch missed calls.
Not sure if there's a technical reason why this can't happen but it would be handy as we use an IVR after hours (to select two different options) and a standard VM during business hours to catch missed calls.
+2. I have this arrangement in my stand alone asterisk now and would hope to have it once/if I convert to Telecube.
After hours routing:
1 � ring all extensions (in case of night owls or weekend warriors wandering about)
2 � then After Hours IVR offering choices depending on purpose of call.
All the after hours IVR choices go to a voicemailbox directly, the relevant dept will handle accordingly.
The switch has been flicked.
Sorry, you need to flick it again. There was a bug that stopped the change to MP3 from being saved.
Apologies
Feature request: The ability to use an IVR as a "voicemail" in ring groups
I'm considering ways to implement this, not sure whether to allow the use of a full IVR or add a special keypress/vm option to the ring group itself.
How does one delete a DID?
I've used two of the included numbers to serve two phones connected to the Gigaset C610.
My question: can I use two other numbers from the series but route the calls through a Gigaset A400? (That system is connected directly to the Phone 1 port on my Fritz!Box 7390, whereas the 610 is not directly connected to a phone line...)
not sure whether to allow the use of a full IVR or add a special keypress/vm option to the ring group its
Can I vote for the full IVR? Mainly for the selfish reason that I've already set everything up how I want, and the only missing link is the ability to automatically redirect to the IVR after hours.
automatically redirect to the IVR after hours
Would the IVR have options to call out to fixed/mobiles or just have keypress options for specific voicemails?
Would the IVR have options to call out to fixed/mobiles or just have keypress options for specific voicemails?
I have it set up to do both at the moment... Here's the flow:
When someone rings during normal hours, it rings our office phones. If no-one answers after 15 seconds, it goes to a 'we're on other calls, leave a message' "work hours" vm.
When someone rings after hours, the IVR picks up and gives the spiel about being closed, but offers two options: 1 to contact the on-call tech, 9 to leave a voicemail.
When you press '1' it rings the "pager" mobile (Android phone connected to Telecube using CSipSimple) for 5 seconds to ring and trigger a recurring alert on the phone, and then overflows to an "on call" VM. That voicemail is then emailed to the "pager" mobile phone.
When you press '9', it forwards to a 'thanks for calling, we'll ring back between 9 and 5' "after hours" VM.
Have I made it complicated enough yet? :D Everything works perfectly so long as I manually change from queue to IVR after hours. If there was a way to make that automatic everything would be completely as I wanted.
Any takers?
Any takers?
You can add your extensions to any device you want.
Is that what you were asking?
(That system is connected directly to the Phone 1 port on my Fritz!Box 7390, whereas the 610 is not directly connected to a phone line...)
You can add your extensions to any device you want.
You configure the handset/phone/base station connected on FON 1 (or 2) to respond to whatever number you wish. You can even set it to dial out on any other number/line configured on the F!B.
Have I made it complicated enough yet?
Maybe FreePBX is the answer here, or even RasPBX if you don't have hardware you can spin up a VM on.
Thanks for that, Onhold, shall try this.
Rgds,
LMH
Thank you, bigglesworth � that's what I thought. But being a newcomer to Telecube I thought I'd better check...
Cheers,
LMH
Maybe FreePBX is the answer here, or even RasPBX if you don't have hardware you can spin up a VM on.
We're already with Telecube and everything works just fine. The pricing is too good to bother rolling our own. I'm sure the functionality we need will be added eventually, or a method to do what we want with existing functionality is found. One of the support guys said he'd create a dummy 1300 number to do it but I think he's forgotten heh
Have I made it complicated enough yet? :D
Sounds fine to me. I have several arrays of complexities like that in mine, and I have totally forgotton how they all work. Every year on the local public holidays the boss asks me what we are doing about the phones. My answer is that I dont remember but the Manual says if we press that 'button' there, it all works marvelously.
I have a SNOM 300 which has had an issue ever since the update of the SIP server back in August. I opened a ticket and have had many chats with support but no resolution.
I have to use the old server IP to dial out and the new server IP on another extension to receive calls has anyone else had an issue or is it just me. I have done factory resets on both phone and router.
Router is Billion 7800N
I have a SNOM 300 which has had an issue ever since the update of the SIP server back in August.
What is the issue?
I have to use the old server IP to dial out and the new server IP on another extension to receive calls
Could you explain this a bit more? Are you talking about TC extensions?
Router is Billion 7800N
Is SIP ALG disabled?
Hi John is there a way to setup landline pstn calls to a DID showing CID rather than anonymous. Mobile incoming calls are ok. This would enable effective call screening.
Edit. All fixed promptly thanks Toran
The pricing is too good to bother rolling our own.
I'm not saying switch from Telecube, the pricing will still be the same using FreePBX. That will just be your interface to Telecube i.e. you'll pop your trunks into FreePBX.
I'm not saying switch from Telecube, the pricing will still be the same using FreePBX. That will just be your interface to Telecube i.e. you'll pop your trunks into FreePBX.
Oh, right! I have no idea what I'm doing with VOIP so I'll just roll with Telecubes setup for the minute. I now have a dummy 1300 number doing what I want, so looks like it's all working a treat. Although it does lead me to the next request...
Feature request: Choosing whether to observe DST or not. Looks like it follows Melbourne time at the moment, which means us Queenslanders who don't fiddle with our clocks twice a year have to remember to change the time based routing manually.
Query: Am I smoking drugs, or was there an option for sending SMS in the control panel for a while? I swear it was there when I was poking around but beggared if I can find it now.
What is the issue? Can call in on new sip server but can't call out and visa versa on old sip.server
Are you talking about TC extensions? � Yes
Is SIP ALG disabled? � Yes but I changed it and made no difference.
today I did factory reset on phone, router and replaced ethernet cable � now I can't call out on any extension.
So I installed a clean never used D-Link 2740-B and same result.
Reinstalled 7800N and placed Snom in DMZ ...no change. As soon as I hit the dial button I get error message:
"Disconnected Unacceptable"
Sip log:
Received from udp:103.193.167.41:5060 at 6/10/2015 19:12:14:658 (487 bytes):
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4
From: "Andrew " <sip:1016XXX@sip.telecube.net.au>;tag=h2md2g9aqo
To: <sip:66XXXXX@sip.telecube.net.
Call-ID: 5613825e5ba0-0niqeujqxwp9
CSeq: 1 INVITE
Server: Asterisk (Telecube) PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
Well after 6 weeks I was reading through http://stackoverflow.com/ and found "Turning RTP encryption off" and that appears to have fixed it. For some reason until today the old sip server was not an issue with dialling out but today it also failed ... the 488 error was the clue. Thanks to stackoverflow .
Is there an issue with caller ID verification at the moment? Have tried a few times to verify ID but every time it keep cutting out after the second or third digit. And when I keep retrying I keep getting locked out for 60 minutes.
Matt
Caller ID verification now working well :) Thank you to whoever fixed it!
Is there anyone using a Telecube DID (well in my example it was ported from MNF) with a FritzBox 7490 inbuilt answering machine? Since the number has ported over, the call seems to divert to the answering machine but doesn't play the greeting, there is just a beep.
If I call my home phone number which is connected to the same answering machine it plays the greeting and works fine. The answering machine used to work just fine with the same DID when it was with MNF so I am a little confused...
Cheers.
How is the F!B answering Machine configured? Is it set to
React to all numbers
What happens if you set it to
React only to the following numbers
and select to respond only to the TC number?
Otherwise delete the AM and configure it again.
It's set to only react to my home phone number and TC DID.
If I set it to react to just the TC DID the same thing happens when I call the TC number, just a beep and no greeting.
If I set it to just the home phone number it works fine when calling the home phone number.
Dunno then, weird � Do you have a greeting delay rather than immediately or whatever; try changing that setting?
As I said delete the AM and try again........
Oh, and when all else fails, do a restart.
(Good luck)
Thanks! I've tried rebooting, no luck.
I just tried creating a new answering machine and assigning the TC account to that with no success.
I didn't want to delete the existing one as it would take out existing greeting and saved messages etc.
Seems to be something specific to TC that isn't playing nice.
Appreciate the pointers though :)
It's set to only react to my home phone number and TC DID.
When a call comes in from a DID the actual DID number does not appear in the invite which comes in from the extn the DID is linked to. Only the CID and extn username is referenced.
Well the call that comes in seems to hit the answering machine but then beeps and drops, according to the call logs:
07.10.15 11:09 61405xxxyyy Answering Machine 2 028090xxyy 0:01
Edit, I've just assigned my Maxo DID at the same answering machine and it works perfectly, the accounts are also configured the same on the FritzBox, so something definitely happening with the TC did.
07.10.15 11:56 0405xxxyyy Answering Machine 2 024304xxyy 0:01
When a call comes in from a DID the actual DID number does not appear in the invite which comes in from the extn the DID is linked to. Only the CID and extn username is referenced.
The fritz box refers to a VoIP account by it's 'number', which is not taken from any SIP data � but is the number that you enter into that field when you setup the VoIP account in the fritz interface. Basically when you assign an answering machine to a 'number', you're just attaching it to a specific account ID.
@Nitro � this seems more like a Fritz!Box issue than a Telecube issue; so you may get more luck finding a solution in one of the various existing Fritz threads, or starting a new one in the VoIP forum.
@Nitro � this seems more like a Fritz!Box issue than a Telecube issue; so you may get more luck finding a solution in one of the various existing Fritz threads, or starting a new one in the VoIP forum.
Thanks but I just updated my last post, my Maxo account which is configured identically to my TC account pointing to the same answering machine is working fine, the TC one is not.
you're just attaching it to a specific account ID
Wouldn't the account ID be an extension number not a DID?
Take the DID 0212345678
I'm sure with Maxo if you are looking at the Asterisk CLI you will see FROM_DID 0212345678 when a call comes in.
Using that same DID with TC, the CLI would show FROM_DID 10XXXXXXX where 10XXXXXXX is the extension the call is coming in on.
Wouldn't the account ID be an extension number not a DID?
No � you can type whatever you want into that field when you're setting up the account � it's just an internal account identifier.
The field where you enter the extension number is the "Username" field.
The field where you enter the extension number is the "Username" field.
A standard incoming sip invite contains a TO field which identifies your sip account. Whether is called an extension number or a authUserID or whatever it doesnt matter .
It also contains a FROM field which identifies the Caller and is information extracted when the caller initiates a call.
You may be able to use your DID number to identify your account but this is not usual as the DID may be linked to one or more accounts.
The actual DID number does not appear in a TC invite anywhere.
In an mnf account it is often added as a Diversion= DID but cant be used for call outing.
A standard incoming sip invite contains a TO field which identifies your sip account. Whether is called an extension number or a authUserID or whatever it doesnt matter .
That's all lovely � but Nitro is using a Fritz!Box, so I clarified what that particular field actually means in a Fritz!Box � it has absolutely nothing to do with the SIP standard. It has nothing to do with authenticating with the server.
The "Telephone Number" is a label. A label that you can type whatever number you like in there. When assigning a VoIP account to a handset, or to an answering machine, or anything else, you select the "telephone Number" for that service. When initiating a call you can have the "telephone Number" set as the FROM field, or you can choose numerous other options to populate the FROM field.
Seems to be something specific to TC that isn't playing nice.
I just set up one of my TC extensions in my Fritz 7490, and then setup the answering machine, and it works perfectly.
Can you call in via TC, and answer the call?
It's set to only react to my home phone number and TC DID.
Did you try the all numbers option?
Can you call in via TC, and answer the call?
Yep, it's been used for a number of weeks to receive calls, it wasn't until today when we weren't home and got a message from a family member saying that our answering machine no longer works.
Did you try the all numbers option?
I just tried this now and it didn't make a difference, a call to the TC DID ended with a beep and the call ending, the Maxo DID went to the answering machine and played the greeting successfully.
so I clarified what that particular field actually means in a Fritz!Box
That is important because there is no std for field identificaion and each device makes up their own name for each box.
it has absolutely nothing to do with the SIP standard
That is where I must disagree. Each sip UA has been scripted on the basis of the sip standard and not the GUI so if it doesnt find the info it needs in the correct place in an Invite the call will fail.
I only posted when Nitro said
It's set to only react to my home phone number and TC DID
That no device can react to a TC DID when it is not present in an Invite.
Edit. I just rang in via my TC DID and it is working perfectly.
That is where I must disagree. Each sip UA has been scripted on the basis of the sip standard and not the GUI so if it doesnt find the info it needs in the correct place in an Invite the call will fail.
I meant the label of the field in the Fritz!Box GUI has nothing to do with the standard. Nitro is looking at the GUI, therefore the label that is present in the GUI is what matters.
a call to the TC DID ended with a beep and the call ending,
Just activate packet monitoring and see what is happening.
Yep, it's been used for a number of weeks to receive calls, it wasn't until today when we weren't home and got a message from a family member saying that our answering machine no longer works.
I just tried this now and it didn't make a difference, a call to the TC DID ended with a beep and the call ending, the Maxo DID went to the answering machine and played the greeting successfully.
How odd.
Have you tried setting up a new TC extension in the FB, and seeing if that works?
You don't have any call diversions set up in the FB do you?
I'm assuming that with a call duration =< 1min that they're showing in the log as incoming calls, and not missed calls?
therefore the label that is present in the GUI is what matters
Totally agree so we are talking about the same thing . Its just a pity there is no std for GUI labelling.
How odd.
I'd definitely agree with that! :)
Have you tried setting up a new TC extension in the FB, and seeing if that works?
Not yet, I've always used the single extension but I will give that a go.
You don't have any call diversions set up in the FB do you?
Nope, no diversions in use, there are some pesky numbers that I've added to a rule that sends them straight to the answering machine but that's on the PSTN line only.
I'm assuming that with a call duration =< 1min that they're showing in the log as incoming calls, and not missed calls?
They definitely show as incoming not missed, it's like the call goes to the answering machine, something gets broken in the negotiation then drops but will try setting up a new extension and add that into the TC ring group and see how that goes.
Worst case, I might pester you via PM if that's ok to see what your settings look like to make sure there is nothing obvious missing.
Thanks for all the help, I appreciate it :)
Worst case, I might pester you via PM if that's ok to see what your settings look like to make sure there is nothing obvious missing.
Go for it.
Go for it.
Thanks for your suggestions, creating a new TC extension, adding it to ring queue then registering to that via F!B seems to have fixed it. Seems like it was either the extension on the TC side or the account on the F!B side. All good now, thanks so much again :)
There's an option in the voicemail to email services now to select MP3 as the message format.
That's great, and thanks for getting the caller ID fixed on the voicemail to email messages.
Everything works fine now!
What does it mean if one of our extensions keeps ringing from a CID of "1001"?
The DID only goes to our ring group or IVR, there is no routing for any separate extension.
What does it mean if one of our extensions keeps ringing from a CID of "1001"?
Do you have ports forwarded to your PBX?
Also had a missed call yesterday from a "friends123"
Nothing recorded in my Telecube records
Am using default 60/61 portts for my ATA/IP phone so suspecting someone just port scanning and dialing??
Do you have ports forwarded to your PBX?
Not really sure what that means.
We just have a Telecube account, and a few Yealink handsets. They're on their own ADSL connection (modem/router -> switch -> handsets) and the sum total of configuration was to login to the Yealink UI and change the login details and a few configuration options for BLF and the like.
None of the calls show up in our Telecube account thingo and it's only 1 of like 8 extensions we have that dials.
I'll just add it to the blacklist on the phones.
We just have a Telecube account, and a few Yealink handsets.
http://support.yealink.com/faq/faqInfo?id=268
Thanks! I'll give it a whirl when I'm back in the office. That looks like exactly what's going down.
Can you dial a queue or linehunt group directly on-net from an extension somehow?
(Useful for testing)
Just wanted to post about a positive experience...
I had a very strange fault that started last week, with a DID number being mysteriously re-routed to the wrong destination. Murray from Telecube thought he had the problem nailed on Friday night but it returned at odd times over the weekend.
Murray was back on it first thing this morning and called me to report that it is now fixed permanently.
I know some people have their doubts about working with small providers like Telecube. However, if this had been a major player, imagine how long one would wait and how many times one would speak to a call centre operator before anyone would even look at such an obscure problem.
This is not the first time that Murray has done more than expected in resolving an issue. Owner John and the Telecube customer base are surely the winners.
Also wanted to add praises for telecube.
I was with another prominent VoIP provider and my account was being used by a third party to make international calls. The provider took no responsibility so I looked through the VoIP forum and stumbled on this thread.
I signed up immediately and ported my numbers over. After a week or so � tried it for the first time and was impressed as to how good it sounded. Even family have pointed out how much better it sounds.
So thanks for a great service and the great customer support!
Không có nhận xét nào:
Đăng nhận xét