I am not familiar with your freeswitch config and what freeswitch should do. Also, I don't deal much with freeswitch in order to assist you with it, maybe other people here can help, if not, you can eventually ask on freeswitch mailing list.

Cheers,
Daniel

On 13/01/16 18:15, malik sherif wrote:

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@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@AbdulKamailioSIP.com to XML[kb-7632689993@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@10.22.52.2 [d52b6ef9-c4f6-4edf-aff9-8a8da3761788]
2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/7632689993@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@10.22.52.2) Ended
2016-01-13 05:37:39.632245 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/7632689993@10.22.52.2 [CS_DESTROY]
2016-01-13 05:37:39.632245 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/7632689991@AbdulKamailioSIP.com!
2016-01-13 05:37:39.653182 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/7632689991@AbdulKamailioSIP.com] has been answered
2016-01-13 05:37:50.532203 [NOTICE] sofia.c:952 Hangup sofia/internal/7632689991@AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING]
2016-01-13 05:37:50.552949 [NOTICE] switch_core_session.c:1641 Session 11 (sofia/internal/7632689991@AbdulKamailioSIP.com) Ended
2016-01-13 05:37:50.552949 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/7632689991@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@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

-- 
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