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