HELP! - Calls not allowed on SIP Trunk

[yaca]

Hi guys,

I'm working on getting my SIP Trunk to work for outgoing calls. The SIP Trunk form my providr seems to be OK because when I register SIP IP Phone (Linksys SPA942) directly to it I can make calls. The Trunk also registers with my VoIP providers but only when I use Generic VoIP as a template. What happens with 3CX system when I set 8 for number prefix I straight get a prompt that I am not allowed to place this call. I am using SPA3102 for PSTN termination and it works fine for calls from and to PSTN and set number prefix as 9 for PSTN calls.

Let me know which part of the log files you could take a look at to help me.

Thanks a lot.

 

[yaca]

Hi,

I just found this on the systemlog:

Provider:[TPNA] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport:

How can I interpret it?

Thanks.

 

[Nebula]

Is the outbound rule for your SIP trunk set to strip the 1st digit?

Paste the top bit of the server log (server status page) after making a failed call.

 

[yaca]

Thanks for quick reply. There you go. I replaced 2 things with xxx and fake domain. Hope it won't make any difference. Yes, I have set to strip 1 digit, that would be 8 in this case.

01:00:19.708 Call::Terminate [CM503008]: Call(36): Call is terminated
01:00:13.506 Line:rintEndpointInfo [CM505003]: Provider:[TPNA] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:10.163.17.11:5060]
01:00:13.506 CallCtrl:nAnsweredCall [CM503002]: Call(36): Alerting sip:[email protected]:5060

 

[yaca]

Sorry, I just noticed I missed few bits:

01:05:55.877 Call::Terminate [CM503008]: Call(37): Call is terminated
01:05:52.940 Line:rintEndpointInfo [CM505003]: Provider:[TPNA] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:10.163.17.11:5060]
01:05:52.940 CallCtrl:nAnsweredCall [CM503002]: Call(37): Alerting sip:[email protected]:5060
01:05:52.065 CallCtrl:nSelectRouteReq [CM503004]: Call(37): Calling: VoIPline(Ln.10000@TPNA)@[Dev:sip:[email protected]:5060]
01:05:52.049 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
01:05:52.049 Extension:rintEndpointInfo [CM505001]: Ext.201: Device info: Device Identified: [Man: Linksys;Mod: SPA-941;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA942-5.1.15(a)] Transport: [sip:10.163.17.11:5060]
01:05:52.049 CallCtrl:nIncomingCall [CM503001]: Call(37): Incoming call from Ext.201 to [sip:[email protected]]

 

[yaca]

Anybody any thoughts on that? Please help if you can.

Would it be anyhting to do with Source Indentification under VoIP provider settings?

Thanks in advance.

 

[Cjay]

Looks to me like you are trying to dial '8' to be passed out to your VoIP provider. Where's the rest of the digits?

 

[yaca]

Well, it does't get through beacuse straight away I get a voice prompt (I believe it's from my VoIP provider) that I am not allowed to place this call.

I guess it must be something to do with Source indentification settings. The thing is I don-t have much clue how to use those ....

Thanks.

 

[Cjay]

Well, it does't get through beacuse straight away I get a voice prompt (I believe it's from my VoIP provider) that I am not allowed to place this call.

Ok - but what are you actually dialing on your phone? Just the 8 or 8xxxxxx (where xxxx is the real number you want to dial?)

 

[yaca]

I am dialling the full number after 8, so 8xxxxxxxxxxxxxx. I am playing with some settings on the VoIP Provider and here is the latest log. I also tried without 8 but it seems like the rule stripps one 0 by default. Please note I am actually dialling 00529242451437 but it seems like the system stripps off the last 3 digist. That would be the problme or still because Device not Identified?

Thanks for your help.

13:37:53.616 Call::Terminate [CM503008]: Call(113): Call is terminated
13:37:53.600 Call::RouteFailed [CM503015]: Call(113): Attempt to reach [sip:[email protected]] failed. Reason: Forbidden
13:37:53.600 CallLeg:nFailure [CM503003]: Call(113): Call to sip:[email protected]:5060 has failed; Cause: 403 Forbidden; from IP:xx.xx.xx.xx:5060 (here the IP address of my VoIP provider)
13:37:53.366 MediaServerReporting:TMFhandler [MS211000] C:113.1: 10.163.18.53:16388 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
13:37:53.194 Line:rintEndpointInfo [CM505003]: Provider:[VoIPProvider] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:10.163.17.11:5060]
13:37:53.194 CallCtrl:nAnsweredCall [CM503002]: Call(113): Alerting sip:[email protected]:5060
13:37:52.569 CallCtrl:nSelectRouteReq [CM503004]: Call(113): Calling: VoIPline:00529242451@(Ln.10000@VoIPProvider)@[Dev:sip:[email protected]:5060]
13:37:52.554 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
13:37:52.554 Extension:rintEndpointInfo [CM505001]: Ext.206: Device info: Device Identified: [Man: Linksys;Mod: SPA-901;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA901-4.1.11(c)] Transport: [sip:10.163.17.11:5060]
13:37:52.554 CallCtrl:nIncomingCall [CM503001]: Call(113): Incoming call from Ext.206 to [sip:[email protected]]

 

[Cjay]

Please note I am actually dialling 00529242451437 but it seems like the system stripps off the last 3 digist

I wonder if this could be a problem with your phone prematurely passing an incomplete dial string to 3cx? I would expect to see the log entry: Incoming call from Ext.206 to [sip:[email protected]] actually showing the complete number being displayed and not dropping off the last 3 digits. If the number doesn't get into 3cx correctly then it certainly won't dial out correctly!

What is the SPA942 dial plan? Is it the default value - which on a 941 is: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
or has it been changed? I'm thinking something is making/forcing the phone to dial out after 12 digits have been dialed.

Regds,
Cjay.

 

[yaca]

Thanks for the clues. Yes, I had same thought that it may be related to dial plans on Linksys phone. I left it as default on both SPA901 or SPA942 as well as ATA. I even tried the 3CX Softphone client where I guess it does not have any dial plans assigned to it. Still same issue. I also tried to delete the prefix number and the 1 stripped digit on the system... And in this very moment when I am writing it I tried X-Lite softphone and unbelivebly it worked! believe me or not I did try that yesterday and nothing. I must have changed something on the system unconciously. I just dial out without any prefix and it goes through.

But you must be right about the dial plans on the Linksys phones or ATAs. I need to think of adding something to it to allow to dial more than 12 digits. But the funny thing is if I register SPA941 or ATA to the SIP Trunk directly I can make telephone calls without worrying about the dial plans, thats the starnge thing isn-t it?

You are most welcome to give my any ideas about modifying dial plans on Linksys stuff.

Thanks,
yaca

 

[yaca]

Just a quick update, my problem was definitely dial plans related so thanks to you guys I am on the right path to make it all work. Thanks for all your input.