Can't make call

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spaldo1973

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Hello everybody,

I set up a generic SIP trunk, as my provider isn't listed. I created some extensions. I downloaded and auto-provisioned 3CXPhone on my desktop. When I try to call my cellphone for testing purposes it says "Destination Not Found."

When I look at the logs on the mgt. console it says

10:41:07.649 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 6.0.20943.0] PBX contact: [sip:[email protected]:5060]

Under ext 100's phone provisioning I don't have a model or mac address listed b/c I don't have a physical phone yet. This is strictly for testing purposes. Is the above error b/c of that or does this have to do w/ the SIP info?
 
You need to include more of the logs, from the point that 3CX receives the digits to the point where it fails to find a destination.

I assume that you created an outbound rule to correctly route the digits that you are dialling?
 
Hey leejor,

Thanks for responding! I will say that I'm having terrible difficultly with this. I'm a newb to voip and pbx systems. I've tried to read some of the 3CX docs and watch videos. I'm certain that some, if not all, of my settings are probably wrong. This whole thing is as clear as mud for somebody who's never done this. Anywho, below is a copy of my server logs from the entire call sequence.

Code:
12:03:45.082|.\StunClient.cpp(356)|Log2|STUN|StunClient::onInitTests:[CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server [ V4 96.9.132.83:3478 UDP target domain=unspecified mFlowKey=0 ] over Transport [ V4 192.168.134.7:5060 UDP target domain=unspecified mFlowKey=0 ]<br>
12:03:45.124|.\StunClient.cpp(141)|Log2|STUN|StunClient::process:[CM506003]: Resolved SIP external IP:port has changed to ([ V4 12.153.20.132:1210 UDP target domain=unspecified mFlowKey=0 ]) on Transport [ V4 192.168.134.7:5060 UDP target domain=unspecified mFlowKey=0 ]<br>
12:03:45.162|.\StunClient.cpp(109)|Error1|STUN|StunClient::process:[CM306003]: SIP IP:port mapping ([ V4 12.153.20.132:1210 UDP target domain=unspecified mFlowKey=0 ]) resolved by STUN server [ V4 96.9.132.83:3478 UDP target domain=unspecified mFlowKey=0 ] differs from the one ([ V4 12.153.20.132:1211 UDP target domain=unspecified mFlowKey=0 ] resolved by STUN server 216.93.246.16<br>
12:03:45.270|.\StunClient.cpp(109)|Error1|STUN|StunClient::process:[CM306003]: SIP IP:port mapping ([ V4 12.153.20.132:1210 UDP target domain=unspecified mFlowKey=0 ]) resolved by STUN server [ V4 96.9.132.83:3478 UDP target domain=unspecified mFlowKey=0 ] differs from the one ([ V4 12.153.20.132:1212 UDP target domain=unspecified mFlowKey=0 ] resolved by STUN server 178.238.134.190<br>
12:03:45.270|.\StunClient.cpp(109)|Error1|STUN|StunClient::process:[CM306003]: SIP IP:port mapping ([ V4 12.153.20.132:1211 UDP target domain=unspecified mFlowKey=0 ]) resolved by STUN server [ V4 216.93.246.16:3478 UDP target domain=unspecified mFlowKey=0 ] differs from the one ([ V4 12.153.20.132:1212 UDP target domain=unspecified mFlowKey=0 ] resolved by STUN server 178.238.134.190<br>
12:11:54.885|.\CallCtrl.cpp(330)|Log2||CallCtrl::onIncomingCall:[CM503001]: Call(5): Incoming call from Ext.100 to <sip:[email protected]:5060><br>
12:11:54.891|.\Extension.cpp(1406)|Log3||Extension::printEndpointInfo:[CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 6.0.20943.0] PBX contact: [sip:[email protected]:5060]<br>
12:11:54.893|.\CallCtrl.cpp(508)|Log3||CallCtrl::onSelectRouteReq:[CM503010]: Making route(s) to <sip:[email protected]:5060><br>
12:11:54.893|.\CallCtrl.cpp(617)|Log2||CallCtrl::onSelectRouteReq:[CM503014]: Call(5): No known route to target: <sip:[email protected]:5060><br>
12:11:54.893|.\Call.cpp(1054)|Log2||Call::RouteFailed:[CM503016]: Call(5): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Found<br>
12:11:54.893|.\Call.cpp(600)|Log2||Call::DoEndCall:[CM503020]: Normal call termination. Reason: Not found<br>
Any help would be greatly appreciated.
 
spaldo1973 said:
12:11:54.893|.\CallCtrl.cpp(617)|Log2||CallCtrl::onSelectRouteReq:[CM503014]: Call(5): No known route to target: <sip:[email protected]:5060><br>
First suggestion is to create an outbound rule which will route call to 1270xxxxxxx

P.S. Please use code tag for the logs. (select quoted logs in the message and press "Code" button[see toolbar])
 
Hey SY,

I will keep the "code" in mind for future reference. I did create an outbound rule that says calls from ext. 100 (which is me) with prefix 270 with 9,10 digit length routes to sip profile I created. Is that not right?

Thanks.
 
spaldo1973 said:
Hey SY,

I will keep the "code" in mind for future reference. I did create an outbound rule that says calls from ext. 100 (which is me) with prefix 270 with 9,10 digit length routes to sip profile I created. Is that not right?

Thanks.
You have dialed 11 digits 1270xxxxxxx.
why do you expect that the rule, which handles 9 or 10 digits started with 270, will be applied?
 
OK. The 9 was a mental lapse on my part. I was thinking 10 digits b/c it says something directly below that about "Strip Digits" and there's a 1 there. I have gone back and put 11 in there w/o success.

I'm still not certain that my trunk settings are correct. Here's what my ISP sent me.

sip-server = bw.voipinternetcable.net
registrar = bw.voipinternetcable.net
domain = bw.voipinternetcable.net

sip identity = 270xxxxxxx
auth-name = 270xxxxxxx
password = xxxxx

Your BTN for this trunk is 270xxxxxxx

It has 3 did's associated with it 270xxxxxxx - 270xxxxxxx - 270xxxxxxx

I've tried to create a port/trunk based on this information. Yet, idk if i've got everything it the right place.

Does anybody know of any available documentation that breaks down each section and what parameters are needed and why and where you get them? I've ordered the 3CX book, but its going to be a few days to arrive. Everything that I've found so far has been geared towards a prior knowledge of PBX/SIP/telephony.

I appreciate your time and attention!
 
spaldo1973 said:
I was thinking 10 digits b/c it says something directly below that about "Strip Digits" and there's a 1 there. I have gone back and put 11 in there w/o success.

The Strip Digits field is the number of digits ( not the actual digits to be removed) that will be removed from the beginning of the original dialled number.

If you don't want any digits taken off, the number sent out as dialled, then put in a 0 (zero).
 
OK, I've got the outbound rule fixed. Now, when I try to make a call the following error appears in the logs.

Code:
12:17:55.776  [CM503003]: Call(2): Call to sip:[email protected]:5060 has failed; Cause: 403 Forbidden; from IP:63.209.193.23:5060

Any ideas?
 
spaldo1973 said:
OK, I've got the outbound rule fixed. Now, when I try to make a call the following error appears in the logs.

Code:
12:17:55.776  [CM503003]: Call(2): Call to sip:[email protected]:5060 has failed; Cause: 403 Forbidden; from IP:63.209.193.23:5060

Any ideas?

The idea is to ask the provider about the reasons of "403 Forbidden; from IP:63.209.193.23:5060"

Thanks
 
SY said:
Call to sip:[email protected]:5060 has failed; Cause: 403 Forbidden;

The majority of providers want some sort of prefix..1,0, 011,01,etc. before the number. Check with yours to see what format they want to see.
 
OK, I changed bw.voipinternetcable.net to trunk.voipinternetcable.net and now I have dialtone!!! But, I still can't make a call.

I did some more poking around and I found "Tunnel Password" under network settings. I said that it needed to match the Tunnel Password of the Tunnel Section of the 3CX VIOP client. Is this the authentication password for my sip trunk? If not, where do I find this so I can make sure it matches?

Here's the log:
Code:
15:29:49.561  [CM503020]: Normal call termination. Reason: Forbidden
15:29:49.561  [CM503016]: Call(6): Attempt to reach <sip:[email protected]> failed. Reason: Forbidden
15:29:49.558  [CM503003]: Call(6): Call to sip:[email protected]:5060 has failed; Cause: 403 Forbidden; from IP:63.209.193.59:5060
15:29:49.407  [CM503025]: Call(6): Calling VoIPline:2705555555@(Ln.10000@BGMU)@[Dev:sip:[email protected]:5060]
15:29:49.275  [MS201000] Use STUN server 'stun3.3cx.com:3478'
15:29:49.267  [MS101005] STUN request failed for ports 9004,9005 on STUN server 'stun2.3cx.com:3478'
15:29:47.458  Currently active calls - 1: [6]
15:29:46.203  [CM503004]: Call(6): Route 1: VoIPline:2705555555@(Ln.10000@BGMU)@[Dev:sip:[email protected]:5060]
15:29:46.202  [CM503010]: Making route(s) to <sip:[email protected]>
15:29:46.200  [CM505001]: Ext.100: Device info: Device Identified: [Man: Cisco;Mod: SPA Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Cisco/SPA504G-7.4.9a] PBX contact: [sip:[email protected]:5060]
15:29:46.197  [CM503001]: Call(6): Incoming call from Ext.100 to <sip:[email protected]>

I'm getting close and I'd love to be able to make a phone call.
 
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