Problems between an UC520 and Asterisk with sip trunk

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Jun 7th, 2010
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I have an UC520 and Asterisk with a sip trunk created between them, the calls from the UC520 to the Asterisk are ok, but the calls form de Asterisk to the UC520 are always busy.

Logs from the asterisk show that the first part of the call is ok, but the call is not complete, this means that the part where the extensions are with @ipuc520 doesn't appear


I created a sip trunk from de CCA 1.9 and it puts this for incoming calls for the dial peer, if I compare with a CCME, there is no configuration for incoming call there


/* Style Definitions */ table.MsoNormalTable {mso-style-name:"Tabla normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin:0cm; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-fareast-font-family:Calibri; mso-fareast-theme-font:minor-latin; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi; mso-fareast-language:EN-US;}

dial-peer voice 1000 voip

permission term

description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target ipv4:x.y.z.w

incoming called-number .%

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad


And there is no configurarion at all that could block the calls

The x.y.z.w was the sip server ip (asterisk ip)

The comminication between sip and h323 are allowed in the four ways

The allowed codecs are   g711ulaw and g729r8

Asterisk is working now with other CCME and they are ok so I copied the configuration from those CCME to the UC520 and from the other sip trunks in asterisk the new trunk sip for uc520

The sip trunk created from the CCA was replaces for the one from the CCME that is working now

The routes are ok in Asterisk.

There is no translation profile in incoming calls.

There is no ACL applied in all configuration.

There is no log about callres incoming from the asterisk.


Could anyone halp me pls?



Correct Answer by David Trad about 6 years 9 months ago

Hi Rina,


I am glad it is resolved for you


Can you mark the thread as answered please.




Cheers,


David.

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David Trad Mon, 06/07/2010 - 15:02
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Hi Carlos,


Can you please provide a debug CCSIP and debug voip dial-peer on an incoming call from the Asterisk box.


Also what form of Asterisk are you using? Is it elastix, trixbox or a vanilla flavor?




Cheers,



David.

Carlos Montero Mon, 06/07/2010 - 15:31
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Hi,

There is the debug in the atached txt file, the caller number is 3400, the called number is 7129; AA.BB.CC.DD is asterisk public ip, WW.XX.YY.ZZ is UC 520 public ip

And the Asterisk is a TRIXBOX

Thanks for answering!




/* Style Definitions */ table.MsoNormalTable {mso-style-name:"Tabla normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin-top:0cm; mso-para-margin-right:0cm; mso-para-margin-bottom:10.0pt; mso-para-margin-left:0cm; line-height:115%; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi; mso-fareast-language:EN-US;}

David Trad Mon, 06/07/2010 - 16:49
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Hi Carlos,


Great thanks for posting that.


I see the Dial-Peer is a sucessfull match, so the next question(s) to ask.


Where are you sending that incoming call from the Asterisk box? Do you have a Translation rule in place? Are you sending it to a Directory number? Is it supposed to come into the CME and then out a Voice-Port?


I suspect the call is hitting the system, it is processing correctly right up to the point where it asks "Where am i supposed to go".


Any further information you can provide would be appreciated, or if you are willing too maybe post your entire running config with all the sensitive information removed.



Cheers,



David.

Carlos Montero Mon, 06/07/2010 - 19:54
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Where are you sending that incoming call from the Asterisk box?

In UC520, there is no configuration for sending this calls, just a dial peer that was created by the CCA when I created the trunk sip


Do  you have a Translation rule in place? Are you sending it to a Directory  number? Is it supposed to come into the CME and then out a Voice-Port?

yes, I have translation rules in UC520 bur they are not been used in incoming calls, there is no directory number, yes, it is suppoused that.



I am ataching the complete show run without sensible infomation.



Thanks


By the way, I am Rina, and I am using the username of a coworker for publishing my problem



Rina

David Trad Mon, 06/07/2010 - 23:33
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Hi Rina,


Help me to try and understand what you are trying to do.


In this code snippet i see the following:



001808: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=7129, Called Number=7129, Peer Info Type=DIALPEER_INFO_SPEECH
001809: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=7129
001810: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
001811: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=20036


This looks as though you have a call coming in from the Asterisk system to number 7129, which then leads to this according to the config file you provided.


 number 7129
label 7129
description7129
name 7129
call-forward busy 6001
call-forward noan 6001 timeout 10


Which at this point I am going to assume this is ephone-dn  10 (Please confirm). If this is the case then the inbound call is being matched correctly to a DN (Which has its own dial-peer tag "Dial-peer Tag=20036".


But then i see this:


001817: 1w3d: //-1/55940098BA19/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
001818: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
001819: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules Attempt



So the incoming call has been matched to Dial-peer 1000 which is an incoming VoIP dial-peer:


dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad



But then can see it has no where to go. So either I am reading this all wrong and the 7129 number is a result of another call taking place whilst you were debugging the system, or it is part of the debug and I am missing something here.



Rina,  just so I understand this all. Are you trying to do WAN type calling from one system UC-500 (System "A") to the Asterisk system ( System B) and same? And so far calls going from the UC-500 to the Asterisk system are fine, but calls coming in from the Asterisk system to the UC-500 are not?


What happens on the Asterisk side when you try to call an Extension on the UC-500, do you get any ringing? Or is it a fast busy tone?



I am going to look over your configuration and debug a little further when I get home, maybe I am missing something here and can identify it.



Cheers,



David.

Carlos Montero Tue, 06/08/2010 - 10:19
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Hi,

The dial-peer 1000 was the problem, I erased it and now I can call to the UC520.

Before this the busy tone was inmediately.

Thanks a lot for your help

Rina

Correct Answer
David Trad Tue, 06/08/2010 - 14:14
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Hi Rina,


I am glad it is resolved for you


Can you mark the thread as answered please.




Cheers,


David.

luistapia980 Thu, 01/24/2013 - 14:42
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Maybe some time has passed for this post. But I wonder if someone could add a how to to configure a SIP Trunk with Elastix and PBX UC500 using CCA.

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