[SR-Users] Kamailio and freeswitch integration for SBC

malik sherif asherif74 at hotmail.com
Wed Jan 13 22:21:32 CET 2016


Kamailio SIP is sending 404 not found since freeswitch is generated UN-allocated number, the call got rejected and goes to voicemail.

Thank you again for your help

Abdul


________________________________
From: Daniel-Constantin Mierla <miconda at gmail.com>
Sent: Wednesday, January 13, 2016 9:06 PM
To: malik sherif; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC

What application is sending the 404?

Cheers,
Daniel

On 13/01/16 21:43, malik sherif wrote:

Any hint as to how to correct this issue?


1 404 UNALLOCATED_NUMBER Unallocated (unassigned) number [Q.850 value 1] This cause indicates that the called party cannot be reached because, although the called party number is in a valid format, it is not currently allocated (assigned).

________________________________
From: sr-users <sr-users-bounces at lists.sip-router.org><mailto:sr-users-bounces at lists.sip-router.org> on behalf of malik sherif <asherif74 at hotmail.com><mailto:asherif74 at hotmail.com>
Sent: Wednesday, January 13, 2016 8:11 PM
To: Kamailio (SER) - Users Mailing List; miconda at gmail.com<mailto:miconda at gmail.com>
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC


is there a new to edit vars.xml file? I haven't touched this file but one of the warning about default password


016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Open /usr/local/freeswitch/conf/vars.xml and change the default_password.

________________________________
From: sr-users <sr-users-bounces at lists.sip-router.org><mailto:sr-users-bounces at lists.sip-router.org> on behalf of malik sherif <asherif74 at hotmail.com><mailto:asherif74 at hotmail.com>
Sent: Wednesday, January 13, 2016 5:15 PM
To: Kamailio (SER) - Users Mailing List; miconda at gmail.com<mailto:miconda at gmail.com>
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC


Thanks again Daniel for replying.

Now the call is failing with 404 not found and goes to voicemail. I was calling between 7632689991 and 7632689993, I looked the extensions on freeswitch, and look OK but it is possible I might have missed something. Freeswitch issues the following errors. Thank you again for your help

Abdulmalik Sherif


2016-01-13 05:37:29.572184 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/7632689991 at AbdulKamailioSIP.com<mailto:sofia/internal/7632689991 at AbdulKamailioSIP.com> [e945266d-8eec-4c0e-80b4-b306f43e18df]
2016-01-13 05:37:29.572184 [INFO] mod_dialplan_xml.c:635 Processing 7632689991 <7632689991>->kb-7632689993 in context public
2016-01-13 05:37:29.572184 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/7632689991 at AbdulKamailioSIP.com<mailto:sofia/internal/7632689991 at AbdulKamailioSIP.com> to XML[kb-7632689993 at default]
2016-01-13 05:37:29.572184 [INFO] mod_dialplan_xml.c:635 Processing 7632689991 <7632689991>->kb-7632689993 in context default
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Open /usr/local/freeswitch/conf/vars.xml and change the default_password.
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Once changed type 'reloadxml' at the console.
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
2016-01-13 05:37:39.632245 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/7632689993 at 10.22.52.2<mailto:sofia/internal/7632689993 at 10.22.52.2> [d52b6ef9-c4f6-4edf-aff9-8a8da3761788]
2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/7632689993 at 10.22.52.2<mailto:sofia/internal/7632689993 at 10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER]
2016-01-13 05:37:39.632245 [INFO] mod_dptools.c:3244 Originate Failed.  Cause: UNALLOCATED_NUMBER
2016-01-13 05:37:39.632245 [NOTICE] switch_core_session.c:1641 Session 12 (sofia/internal/7632689993 at 10.22.52.2<mailto:sofia/internal/7632689993 at 10.22.52.2>) Ended
2016-01-13 05:37:39.632245 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/7632689993 at 10.22.52.2<mailto:sofia/internal/7632689993 at 10.22.52.2> [CS_DESTROY]
2016-01-13 05:37:39.632245 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/7632689991 at AbdulKamailioSIP.com<mailto:sofia/internal/7632689991 at AbdulKamailioSIP.com>!
2016-01-13 05:37:39.653182 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/7632689991 at AbdulKamailioSIP.com<mailto:sofia/internal/7632689991 at AbdulKamailioSIP.com>] has been answered
2016-01-13 05:37:50.532203 [NOTICE] sofia.c:952 Hangup sofia/internal/7632689991 at AbdulKamailioSIP.com<mailto:sofia/internal/7632689991 at AbdulKamailioSIP.com> [CS_EXECUTE] [NORMAL_CLEARING]
2016-01-13 05:37:50.552949 [NOTICE] switch_core_session.c:1641 Session 11 (sofia/internal/7632689991 at AbdulKamailioSIP.com<mailto:sofia/internal/7632689991 at AbdulKamailioSIP.com>) Ended
2016-01-13 05:37:50.552949 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/7632689991 at AbdulKamailioSIP.com<mailto:sofia/internal/7632689991 at AbdulKamailioSIP.com> [CS_DESTROY]

###########################################################################################################################



My extensions are as follow:

<include>
  <user id="7632689991">
    <params>
      <param name="vm-password" value="1001"/>
    </params>
    <variables>
      <variable name="accountcode" value="7632689991"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 7632689991"/>
      <variable name="effective_caller_id_number" value="7632689991"/>
    </variables>
  </user>
</include>

##########################################################################

<include>
  <user id="7632689993">
    <params>
      <param name="vm-password" value="1003"/>
    </params>
    <variables>
      <variable name="accountcode" value="7632689993"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension Sherif"/>
      <variable name="effective_caller_id_number" value="7632689993"/>
    </variables>
  </user>
</include>
############################################################################




________________________________
From: Daniel-Constantin Mierla <miconda at gmail.com><mailto:miconda at gmail.com>
Sent: Wednesday, January 13, 2016 6:34 AM
To: malik sherif; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC

Hello,

the error with creating the SIP UA is most probable because of kamailio listening on 5060 and freeswitch trying to do the same.

To troubleshoot the 408, use ngrep or other network sniffing tool, and look on the network to see where the sip request is sent. Like:

ngrep -d any -qt -W byline port 5060

Cheers,
Daniel

[http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>



--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
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