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
On 13/01/16 00:52, malik sherif wrote:
Hello Daniel,
No I didn't configure freeswitch with loopback but for some reason it was going to the
loopback , it consider it as default network but I am able to point both kamailio and
freeswitch to 10.22.52.2 by disabling IP-v6 for both external-ipv6.xml and
internal-ipv6.xml. Freeswitch was complaining about the following error.
sofia.c:2853 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]
:5060;transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC)
netstat -unlp now shows what I want but call is time out with 408, I might have to check
if port 5090 reachable but I am still wandering why i am getting 408.
udp 0 0 10.22.52.2:5060 0.0.0.0:*
10603/kamailio
udp 0 0 10.22.52.2:5090 0.0.0.0:*
10469/freeswitch
udp 0 0 10.22.52.2:5092 0.0.0.0:*
10469/freeswitch
Thanks again Daniel for responding
Abdul
________________________________
From: sr-users
<sr-users-bounces@lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org>
on behalf of Daniel-Constantin Mierla
<miconda@gmail.com><mailto:miconda@gmail.com>
Sent: Tuesday, January 12, 2016 11:07 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Hello,
did you configure the freeswitch to listen on loopback? You would need to do bridging of
singnaling and eventually rtp between the network interface and loopback if you want this
kind of topology.
Cheers,
Daniel
On 12/01/16 19:00, malik sherif wrote:
Hello Abdul Basit,
I specified that kamailio and freeswitch to share same IP address but different udp port
but netstat -unlp show freeswitch use the loopback IP address (port 5090 and 5092) and
kamailio show the loopback IP and 10.22.52.2 port 5060. Does freeswitch has to go to
loopback IP address?
Thank you very much Abdul Basit for responding.
Abdulmalik Sherif
udp 0 0 10.22.52.2:5060 0.0.0.0:*
31036/kamailio
udp 0 0 127.0.0.1:5060 0.0.0.0:*
31036/kamailio
udp 0 0 127.0.0.1:5090 0.0.0.0:*
30958/freeswitch
udp 0 0 127.0.0.1:5092 0.0.0.0:*
30958/freeswitch
________________________________
From: sr-users
<sr-users-bounces@lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org>
on behalf of Abdul Basit <mailto:basitstar@hotmail.com>
<basitstar@hotmail.com><mailto:basitstar@hotmail.com>
Sent: Tuesday, January 12, 2016 2:55 AM
To: Kamailio SER - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Hi AbdulMalik,
brother this is not kamailio related issue, this is some misconfiguration.
I think, there is some port misconfiguration , kamailio running on 5060 and also
freeswitch running on 5060, make ur kamailio run specfically on 5060, and freeswitch on
different port entirely. please do it and share netstat result again
udp 0 0 10.22.52.2:5060 0.0.0.0:*
9075/kamailio
udp 0 0 ::1:5060 :::*
9002/freeswitch
Regards,
AB
________________________________
From: asherif74@hotmail.com<mailto:asherif74@hotmail.com>
To: sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
Date: Mon, 11 Jan 2016 23:47:41 +0000
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
When I run netstat -unlp command it show the following list. Is it correct for freeswitch
to have the loopback IP address? I think this is maybe a reason the invite is timeout with
408 but I am not sure. How can I fix this problem? I just want to confirm if integrating
Kamailio with freeswitch works as SBC?
Thanks Again
udp 0 0 10.22.52.2:7060 0.0.0.0:*
9075/kamailio
udp 0 0 10.22.52.2:5060 0.0.0.0:*
9075/kamailio
udp 0 0 127.0.0.1:5090 0.0.0.0:*
9002/freeswitch
udp 0 0 127.0.0.1:5092 0.0.0.0:*
9002/freeswitch
udp 0 0 ::1:5060 :::*
9002/freeswitch
udp 0 0 ::1:5080 :::*
9002/freeswitch
________________________________
From: sr-users
<sr-users-bounces@lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org>
on behalf of malik sherif
<asherif74@hotmail.com><mailto:asherif74@hotmail.com>
Sent: Monday, January 11, 2016 5:03 PM
To: sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Hello,
Any hint? How do I get a response? is it through user digest?
Thanks
Abdulmalik Sherif
________________________________
From: sr-users
<sr-users-bounces@lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org>
on behalf of malik sherif
<asherif74@hotmail.com><mailto:asherif74@hotmail.com>
Sent: Friday, January 8, 2016 7:39 PM
To: <mailto:sr-users@lists.sip-router.org>
sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
Subject: [SR-Users] Kamailio and freeswitch integration for SBC
Hello,
I was able to have successful call with kamailio and after integrating kamailio with
freeswitch using the following link, the invite timeout and as a result the call
failed.The status on kamailio show these info. Do I need to download outbound model? I am
running kamailio ver 4.1.1 and freeswitch 1.4.18.
kamailio[19153]: 0(19162) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import
bind_ob
kamailio[19153]: 0(19162) INFO: rr [rr_mod.c:159]: mod_init(): outbound module not
available
<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc
[
http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit…
freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc [Asipto
...<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
kb.asipto.com
The scope of this tutorial is to show how you can use Kamailio (former OpenSER) and
FreeSWITCH to build a complete SIP/VoIP platform for large number of subscribers.
Thank you very much and your help is appreciated.
Thanks
Abdul
<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
[
http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit…
freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc [Asipto
...<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
kb.asipto.com
The scope of this tutorial is to show how you can use Kamailio (former OpenSER) and
FreeSWITCH to build a complete SIP/VoIP platform for large number of subscribers.
_______________________________________________ SIP Express Router (SER) and Kamailio
(OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda<http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio -
http://www.asipto.com
http://miconda.eu
--
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