SIP Trunk - can't get incoming to work, outgoing is fine.

sniderit
Posts: 6
Member Since:
2010-04-14

I can't figure this one out! I setup a SIP trunk for my office. I am running trixbox CE 2.8.0.4. I have 1 SIP trunk from VoipVoip.com and my outgoing calls work fine. My incoming come through to the trixbox but the caller doesn't get an answer from the IVR if I have the IVR setup, if I send the inbound route straight to my only extension (100) it doesn't ring, if I send it straight to my extensions voicemail it doesn't get through. If I watch asterisk -r it shows it tries to do everything but nothing ever plays on the phone, I have even tried typing 100 on the calling phone to try to send it to my extension and nothing.

When I go to the reports it shows "hang" under Dst and ANSWERED under Disposition.

How I want it to function is if someone calls this number, the inbound route points to a Time Condition I have setup called Incoming. Incoming specifies that if durring business hours (M-F 8-5) it goes to my BusinessHours IVR, if it doesn't match then it goes to my Afterhours IVR which each have a recording they play and go to my extension if during business hours and to my voicemail if not. I have not messed with follow me yet until I can get this working.

The trunk works fine if I use X-lite and enter in my SIP trunk information. Because I can watch the call come in by doing "asterisk -r" in putty and that I can make outgoing calls I would assume that the trunk is working fine and that it is a trixbox problem. What could do this? It's like it keeps hanging instead of transferring to the greeting message or my voicemail.



sniderit
Posts: 6
Member Since:
2010-04-14
Something else that is very

Something else that is very odd to me. If I go to my sip trunk and check disable trunk, submit changes, apply configuration changes, and then call the number, it will play my IVR announcement twice. I can type my extension and it plays my busy voicemail recording and lets me record the message. The message will show up on my office phone and I can listen to it.



obeliks
Posts: 878
Member Since:
2010-03-14
This is 3rd thread about

This is 3rd thread about voipvoip in the last few days. Try looking there to pickup some ideas. I think the solution was to not use g.729 codec.
You will need to provide logs/configs if you want help.



obeliks
Posts: 878
Member Since:
2010-03-14
If I go to my sip trunk and

If I go to my sip trunk and check disable trunk, submit changes, apply configuration changes, and then call the number, it will play my IVR announcement twice

You allowed anonymous SIP calls, didn't you ?



sniderit
Posts: 6
Member Since:
2010-04-14
I tried searching, I did

I tried searching, I did find a thread with a similar problem but I can't find an answer. I tried removing allow=g729 from the sip config and then I would get a recording that said the number was not in service.

Here is my log from the server when I call from my cell phone:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Executing [5551298701@from-sip-external:1] NoOp("SIP/5551298701-b6d11438"
, "Received incoming SIP connection from unknown peer to 5551298701") in new sta
ck
-- Executing [5551298701@from-sip-external:2] Set("SIP/5551298701-b6d11438",
"DID=5551298701") in new stack
-- Executing [5551298701@from-sip-external:3] Goto("SIP/5551298701-b6d11438"
, "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/5551298701-b6d11438", "0?fr
om-trunk,5551298701,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/5551298701-b6d11438", "TIMEOUT
(absolute)=15") in new stack
Channel will hangup at 2010-06-18 14:17:26.000 CDT.
-- Executing [s@from-sip-external:3] Answer("SIP/5551298701-b6d11438", "") i
n new stack
-- Executing [s@from-sip-external:4] Wait("SIP/5551298701-b6d11438", "2") in
new stack
-- Executing [s@from-sip-external:5] Playback("SIP/5551298701-b6d11438", "ss
-noservice") in new stack
-- Executing [s@from-sip-external:6] PlayTones("SIP/5551298701-b6d11438", "c
ongestion") in new stack
== Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/555129870
1-b6d11438'
-- Executing [h@from-sip-external:1] NoOp("SIP/5551298701-b6d11438", "Hangup
") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/5551298701-b6d11438", "DID=s")
in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/5551298701-b6d11438", "s,1")
in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/5551298701-b6d11438", "0?fr
om-trunk,s,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/5551298701-b6d11438", "TIMEOUT
(absolute)=15") in new stack
Channel will hangup at 2010-06-18 14:17:29.000 CDT.
-- Executing [s@from-sip-external:3] Answer("SIP/5551298701-b6d11438", "") i
n new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/555129870
1-b6d11438'



sniderit
Posts: 6
Member Since:
2010-04-14
I did before, I went ahead

I did before, I went ahead and reloaded trixbox and started fresh, and now I have that set to No



obeliks
Posts: 878
Member Since:
2010-03-14
show us your trunk

show us your trunk config
did you put config=from-sip-external instead of context=from-trunk in there ?



nttranbao
Posts: 189
Member Since:
2008-02-16
IMO, all unknown/anonymous

IMO, all unknown/anonymous incoming SIP calls are sent to context from-sip-exernal, which then play ss-noservice then hangs up. This happens if you had disabled that Sip trunk and disabled anonymous incoming calls.

Therefore, enable that SIP trunk, and paste the CLI here

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Slojo
Posts: 28
Member Since:
2009-05-12
Enable anonymous

Under General options enable anonymous calls and see if that work?



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