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(a)hotmail.com To: sr-users(a)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(a)lists.sip-router.org> on behalf of malik sherif <asherif74(a)hotmail.com> Sent: Monday, January 11, 2016 5:03 PM To: sr-users(a)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(a)lists.sip-router.org> on behalf of malik sherif <asherif74(a)hotmail.com> Sent: Friday, January 8, 2016 7:39 PM To: sr-users(a)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 Thank you very much and your help is appreciated. Thanks Abdul freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc [Asipto ... 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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Any body has kamailio/freeswitch SBC working? ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Tuesday, January 12, 2016 6:00 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC 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 &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of Abdul Basit &lt;basitstar(a)hotmail.com&gt; 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(a)hotmail.com To: sr-users(a)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 &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Monday, January 11, 2016 5:03 PM To: sr-users(a)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 &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Friday, January 8, 2016 7:39 PM To: sr-users(a)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/_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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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 &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of Daniel-Constantin Mierla &lt;miconda(a)gmail.com&gt; 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 <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: 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://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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 &lt;miconda(a)gmail.com&gt; 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

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 &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Wednesday, January 13, 2016 8:11 PM To: Kamailio (SER) - Users Mailing List; miconda(a)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 &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Wednesday, January 13, 2016 5:15 PM To: Kamailio (SER) - Users Mailing List; miconda(a)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(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 &lt;miconda(a)gmail.com&gt; 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

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 &lt;miconda(a)gmail.com&gt; 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@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: Wednesday, January 13, 2016 8:11 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@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@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: Wednesday, January 13, 2016 5:15 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@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@AbdulKamailioSIP.com<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>! 2016-01-13 05:37:39.653182 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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<mailto: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<mailto: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<mailto: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><mailto: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 [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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

Any idea as to how to correct this problem? Also wireshark trace shows that unrecognized SIP header for invite from freeswitch to kamailio and the header from freeswitch is X-FS-support: update _display ,send_info. How can I disable this header? any other solution beside disabling the header? Thanks Abdulmalik Sherif 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 ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Wednesday, January 13, 2016 9:21 PM To: Kamailio (SER) - Users Mailing List; miconda(a)gmail.com Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC 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 &lt;miconda(a)gmail.com&gt; 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@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: Wednesday, January 13, 2016 8:11 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@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@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: Wednesday, January 13, 2016 5:15 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@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@AbdulKamailioSIP.com<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>! 2016-01-13 05:37:39.653182 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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<mailto: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<mailto: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<mailto: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><mailto: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 [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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

Hello Daniel, Thanks again for responding. Should I attached tcmpdump file or just copy and paste the 404 response? Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of Daniel-Constantin Mierla &lt;miconda(a)gmail.com&gt; Sent: Tuesday, January 19, 2016 10:51 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC I think I asked what do you mean by "freeswitch is generated UN-allocated number" -- is it sending a SIP reply to Kamailio? Maybe you can paste here the sip capture with ngrep for such call. Cheers, Daniel On 19/01/16 23:44, malik sherif wrote: Any idea as to how to correct this problem? Also wireshark trace shows that unrecognized SIP header for invite from freeswitch to kamailio and the header from freeswitch is X-FS-support: update _display ,send_info. How can I disable this header? any other solution beside disabling the header? Thanks Abdulmalik Sherif 2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/7632689993@10.22.52.2<mailto: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 ________________________________ 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: Wednesday, January 13, 2016 9:21 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC 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@gmail.com><mailto:miconda@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@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: Wednesday, January 13, 2016 8:11 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@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@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: Wednesday, January 13, 2016 5:15 PM To: Kamailio (SER) - Users Mailing List; <mailto:miconda@gmail.com> miconda@gmail.com<mailto:miconda@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 <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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 <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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 <mailto:sofia/internal/7632689993@10.22.52.2> sofia/internal/7632689993@10.22.52.2<mailto:sofia/internal/7632689993@10.22.52.2> [d52b6ef9-c4f6-4edf-aff9-8a8da3761788] 2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup <mailto:sofia/internal/7632689993@10.22.52.2> sofia/internal/7632689993@10.22.52.2<mailto: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<mailto: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<mailto:sofia/internal/7632689993@10.22.52.2> [CS_DESTROY] 2016-01-13 05:37:39.632245 [NOTICE] sofia_media.c:92 Pre-Answer <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>! 2016-01-13 05:37:39.653182 [NOTICE] mod_dptools.c:1268 Channel [<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>sofia/internal/7632689991@AbdulKamailioSIP.com<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>] has been answered 2016-01-13 05:37:50.532203 [NOTICE] sofia.c:952 Hangup <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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<mailto: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<mailto: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><mailto: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 [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul .4.........INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com> Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a User-Agent: CALIX 711/10.6.100.2 Accept: application/sdp Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,INFO,OPTIONS Contact: <sip:7632689991@10.22.50.23:5060> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Type: application/SDP Content-Length: 170 v=0 o=7632689991 611732 611732 IN IP4 10.22.50.23 s=CALIX SIP CALL c=IN IP4 10.22.50.23 t=0 0 m=audio 49152 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 a=sendrecv 10:17:24.730003 IP (tos 0x10, ttl 64, id 43742, offset 0, flags [none], proto UDP (17), length 443) 10.22.52.2.5060 > 10.22.50.23.5060: [udp sum ok] UDP, length 415 E.......@.S. .4. .2........?SIP/2.0 100 trying -- your call is important to us From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com> Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 10:17:24.762891 IP (tos 0x10, ttl 64, id 43743, offset 0, flags [none], proto UDP (17), length 1236) 10.22.52.2.5060 > 10.22.50.23.5060: [udp sum ok] UDP, length 1208 E.......@.P. .4. .2....... #SIP/2.0 200 OK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Record-Route: <sip:10.22.52.2;lr=on> From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Contact: <sip:7632689993@10.22.52.2:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 160 Remote-Party-ID: "kb-7632689993" <sip:kb-7632689993@AbdulKamailioSIP.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1453275680 1453275681 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 30964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.787780 IP (tos 0x0, ttl 62, id 207, offset 0, flags [none], proto UDP (17), length 734) 10.22.50.23.5060 > 10.22.52.2.5060: [udp sum ok] UDP, length 706 E.......>... .2. .4........HACK sip:7632689993@10.22.52.2:5090;transport=udp SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 ACK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-421561a4 Max-Forwards: 70 Contact: <sip:7632689991@10.22.50.23:5060> Route: <sip:10.22.52.2;lr> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Length: 0 10:17:24.754176 IP (tos 0x0, ttl 64, id 56102, offset 0, flags [none], proto UDP (17), length 1103) 10.22.52.2.5090 > 10.22.52.2.5060: [bad udp cksum 0x807c -> 0x95df!] UDP, length 1075 E..O.&..@..H .4. .4......;.|INVITE sip:7632689993@10.22.52.2 SIP/2.0 Via: SIP/2.0/UDP 10.22.52.2:5090;rport;branch=z9hG4bKU80S4QFy1t4pr Max-Forwards: 14 From: "7632689991" <sip:7632689991@10.22.52.2>;tag=X6jgFceZ05SSQ To: <sip:7632689993@10.22.52.2> Call-ID: 21f9cf32-3a34-1234-32a7-0004e203079c CSeq: 86317258 INVITE Contact: <sip:ufs@10.22.52.2:5090> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 160 X-FS-Support: update_display,send_info Remote-Party-ID: "7632689991" <sip:7632689991@10.22.52.2>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1757856541 1757856542 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 20800 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.754709 IP (tos 0x10, ttl 64, id 56103, offset 0, flags [none], proto UDP (17), length 397) 10:17:24.762891 IP (tos 0x10, ttl 64, id 43743, offset 0, flags [none], proto UDP (17), length 1236) 10.22.52.2.5060 > 10.22.50.23.5060: [udp sum ok] UDP, length 1208 E.......@.P. .4. .2....... #SIP/2.0 200 OK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Record-Route: <sip:10.22.52.2;lr=on> From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Contact: <sip:7632689993@10.22.52.2:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 160 Remote-Party-ID: "kb-7632689993" <sip:kb-7632689993@AbdulKamailioSIP.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1453275680 1453275681 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 30964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.787780 IP (tos 0x0, ttl 62, id 207, offset 0, flags [none], proto UDP (17), length 734) 10.22.50.23.5060 > 10.22.52.2.5060: [udp sum ok] UDP, length 706 E.......>... .2. .4........HACK sip:7632689993@10.22.52.2:5090;transport=udp SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 ACK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-421561a4 Max-Forwards: 70 Contact: <sip:7632689991@10.22.50.23:5060> Route: <sip:10.22.52.2;lr> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Length: 0 10:17:24.762891 IP (tos 0x10, ttl 64, id 43743, offset 0, flags [none], proto UDP (17), length 1236) 10.22.52.2.5060 > 10.22.50.23.5060: [udp sum ok] UDP, length 1208 E.......@.P. .4. .2....... #SIP/2.0 200 OK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Record-Route: <sip:10.22.52.2;lr=on> From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Contact: <sip:7632689993@10.22.52.2:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 160 Remote-Party-ID: "kb-7632689993" <sip:kb-7632689993@AbdulKamailioSIP.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1453275680 1453275681 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 30964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.787780 IP (tos 0x0, ttl 62, id 207, offset 0, flags [none], proto UDP (17), length 734) 10.22.50.23.5060 > 10.22.52.2.5060: [udp sum ok] UDP, length 706 E.......>... .2. .4........HACK sip:7632689993@10.22.52.2:5090;transport=udp SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 ACK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-421561a4 Max-Forwards: 70 Contact: <sip:7632689991@10.22.50.23:5060> Route: <sip:10.22.52.2;lr> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Length: 0 10:17:24.762891 IP (tos 0x10, ttl 64, id 43743, offset 0, flags [none], proto UDP (17), length 1236) 10.22.52.2.5060 > 10.22.50.23.5060: [udp sum ok] UDP, length 1208 E.......@.P. .4. .2....... #SIP/2.0 200 OK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Record-Route: <sip:10.22.52.2;lr=on> From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Contact: <sip:7632689993@10.22.52.2:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 160 Remote-Party-ID: "kb-7632689993" <sip:kb-7632689993@AbdulKamailioSIP.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1453275680 1453275681 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 30964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.787780 IP (tos 0x0, ttl 62, id 207, offset 0, flags [none], proto UDP (17), length 734) 10.22.50.23.5060 > 10.22.52.2.5060: [udp sum ok] UDP, length 706 E.......>... .2. .4........HACK sip:7632689993@10.22.52.2:5090;transport=udp SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 ACK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-421561a4 Max-Forwards: 70 Contact: <sip:7632689991@10.22.50.23:5060> Route: <sip:10.22.52.2;lr> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Length: 0 10:17:24.762891 IP (tos 0x10, ttl 64, id 43743, offset 0, flags [none], proto UDP (17), length 1236) 10.22.52.2.5060 > 10.22.50.23.5060: [udp sum ok] UDP, length 1208 E.......@.P. .4. .2....... #SIP/2.0 200 OK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Record-Route: <sip:10.22.52.2;lr=on> From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Contact: <sip:7632689993@10.22.52.2:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 160 Remote-Party-ID: "kb-7632689993" <sip:kb-7632689993@AbdulKamailioSIP.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1453275680 1453275681 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 30964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.787780 IP (tos 0x0, ttl 62, id 207, offset 0, flags [none], proto UDP (17), length 734) 10.22.50.23.5060 > 10.22.52.2.5060: [udp sum ok] UDP, length 706 E.......>... .2. .4........HACK sip:7632689993@10.22.52.2:5090;transport=udp SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 ACK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-421561a4 Max-Forwards: 70 Contact: <sip:7632689991@10.22.50.23:5060> Route: <sip:10.22.52.2;lr> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Length: 0 10.22.52.2.5060 > 10.22.52.2.5090: [bad udp cksum 0x7dba -> 0x1ef6!] UDP, length 369 E....'..@.!. .4. .4......y}.SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.22.52.2:5090;rport=5090;branch=z9hG4bKU80S4QFy1t4pr From: "7632689991" <sip:7632689991@10.22.52.2>;tag=X6jgFceZ05SSQ To: <sip:7632689993@10.22.52.2>;tag=0a5403eae2c268b22e9b4df60792062f-2584 Call-ID: 21f9cf32-3a34-1234-32a7-0004e203079c CSeq: 86317258 INVITE Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 10:17:24.755026 IP (tos 0x0, ttl 64, id 56104, offset 0, flags [none], proto UDP (17), length 382) 10.22.52.2.5090 > 10.22.52.2.5060: [bad udp cksum 0x7dab -> 0xe34a!] UDP, length 354 E..~.(..@.". .4. .4......j}.ACK sip:7632689993@10.22.52.2 SIP/2.0 Via: SIP/2.0/UDP 10.22.52.2:5090;rport;branch=z9hG4bKU80S4QFy1t4pr Max-Forwards: 14 From: "7632689991" <sip:7632689991@10.22.52.2>;tag=X6jgFceZ05SSQ To: <sip:7632689993@10.22.52.2>;tag=0a5403eae2c268b22e9b4df60792062f-2584 Call-ID: 21f9cf32-3a34-1234-32a7-0004e203079c CSeq: 86317258 ACK Content-Length: 0 10:17:24.762627 IP (tos 0x0, ttl 64, id 56105, offset 0, flags [none], proto UDP (17), length 1319) 10.22.52.2.5090 > 10.22.52.2.5060: [bad udp cksum 0x8154 -> 0xc720!] UDP, length 1291 E..'.)..@..m .4. .4........TSIP/2.0 200 OK Via: SIP/2.0/UDP 10.22.52.2;branch=z9hG4bK2722.d24afa3ffbe9f3c64de0a40e16fc57f9.0 Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Record-Route: <sip:10.22.52.2;lr=on> From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Contact: <sip:7632689993@10.22.52.2:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 160 Remote-Party-ID: "kb-7632689993" <sip:kb-7632689993@AbdulKamailioSIP.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1453275680 1453275681 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 30964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.762891 IP (tos 0x10, ttl 64, id 43743, offset 0, flags [none], proto UDP (17), length 1236) 10.22.52.2.5060 > 10.22.50.23.5060: [udp sum ok] UDP, length 1208 E.......@.P. .4. .2....... #SIP/2.0 200 OK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d57d2-278ad05a Record-Route: <sip:10.22.52.2;lr=on> From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 INVITE Contact: <sip:7632689993@10.22.52.2:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 160 Remote-Party-ID: "kb-7632689993" <sip:kb-7632689993@AbdulKamailioSIP.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1453275680 1453275681 IN IP4 10.22.52.2 s=FreeSWITCH c=IN IP4 10.22.52.2 t=0 0 m=audio 30964 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:10 10:17:24.787780 IP (tos 0x0, ttl 62, id 207, offset 0, flags [none], proto UDP (17), length 734) 10.22.50.23.5060 > 10.22.52.2.5060: [udp sum ok] UDP, length 706 E.......>... .2. .4........HACK sip:7632689993@10.22.52.2:5090;transport=udp SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 ACK Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-421561a4 Max-Forwards: 70 Contact: <sip:7632689991@10.22.50.23:5060> Route: <sip:10.22.52.2;lr> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Length: 0 10:17:24.788186 IP (tos 0x10, ttl 64, id 56106, offset 0, flags [none], proto UDP (17), length 789) 10.22.52.2.5060 > 10.22.52.2.5090: [bad udp cksum 0x7f42 -> 0x5179!] UDP, length 761 E....*..@. n .4. .4........BACK sip:7632689993@10.22.52.2:5090;transport=udp SIP/2.0 From: 7632689991 <sip:7632689991@AbdulKamailioSIP.com>;tag=1378608-a163217-13c4-17e5-662d0efc-17e5 To: <sip:7632689993@AbdulKamailioSIP.com>;tag=vXSQDHXU3v36B Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-17e5(a)AbdulKamailioSIP.com CSeq: 2 ACK Via: SIP/2.0/UDP 10.22.52.2;branch=z9hG4bK2722.5b81c7f3e4a61ce57604a60cda694802.0 Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-421561a4 Max-Forwards: 16 Contact: <sip:7632689991@10.22.50.23:5060> Proxy-Authorization: Digest username="7632689991",realm="AbdulKamailioSIP.com",nonce="Vp+0QFafsxToZFRGg9d5/pBf0jLCNurV",uri="sip:7632689993@AbdulKamailioSIP.com",response="78d437422bee35a1b1c21a1cef932169",algorithm=MD5 Content-Length: 0 ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Wednesday, January 20, 2016 9:39 PM To: miconda(a)gmail.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Hello Daniel, Thanks again for responding. Should I attached tcmpdump file or just copy and paste the 404 response? Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of Daniel-Constantin Mierla &lt;miconda(a)gmail.com&gt; Sent: Tuesday, January 19, 2016 10:51 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC I think I asked what do you mean by "freeswitch is generated UN-allocated number" -- is it sending a SIP reply to Kamailio? Maybe you can paste here the sip capture with ngrep for such call. Cheers, Daniel On 19/01/16 23:44, malik sherif wrote: Any idea as to how to correct this problem? Also wireshark trace shows that unrecognized SIP header for invite from freeswitch to kamailio and the header from freeswitch is X-FS-support: update _display ,send_info. How can I disable this header? any other solution beside disabling the header? Thanks Abdulmalik Sherif 2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/7632689993@10.22.52.2<mailto: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 ________________________________ 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: Wednesday, January 13, 2016 9:21 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC 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@gmail.com><mailto:miconda@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@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: Wednesday, January 13, 2016 8:11 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@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@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: Wednesday, January 13, 2016 5:15 PM To: Kamailio (SER) - Users Mailing List; <mailto:miconda@gmail.com> miconda@gmail.com<mailto:miconda@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 <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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 <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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 <mailto:sofia/internal/7632689993@10.22.52.2> sofia/internal/7632689993@10.22.52.2<mailto:sofia/internal/7632689993@10.22.52.2> [d52b6ef9-c4f6-4edf-aff9-8a8da3761788] 2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup <mailto:sofia/internal/7632689993@10.22.52.2> sofia/internal/7632689993@10.22.52.2<mailto: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<mailto: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<mailto:sofia/internal/7632689993@10.22.52.2> [CS_DESTROY] 2016-01-13 05:37:39.632245 [NOTICE] sofia_media.c:92 Pre-Answer <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>! 2016-01-13 05:37:39.653182 [NOTICE] mod_dptools.c:1268 Channel [<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>sofia/internal/7632689991@AbdulKamailioSIP.com<mailto:sofia/internal/7632689991@AbdulKamailioSIP.com>] has been answered 2016-01-13 05:37:50.532203 [NOTICE] sofia.c:952 Hangup <mailto:sofia/internal/7632689991@AbdulKamailioSIP.com> sofia/internal/7632689991@AbdulKamailioSIP.com<mailto: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<mailto: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<mailto: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><mailto: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 [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

Hello Daniel, Thanks again for your help Packet 208 invite coming from Freeswitch using IP address while Kamailio using domain name. Do you think this matter? Also, the header "X-FS-Support: update_display,send_info" responded with UN-recognized SIP header. Please, filter the Wireshark trace with 10.22.52.2 IP address and this IP address is the SIP server IP address. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla &lt;miconda(a)gmail.com&gt; Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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

Any hint? ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda(a)gmail.com Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla &lt;miconda(a)gmail.com&gt; Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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

Thanks Samy for responding, I didn't know about FOSDEM. Kamailio is responsible for registration, this is how it suppose to work, the lines are always registered, and I get dial tone all the time. After the placed the call, it goes to voicemail after it plays announcement that the person is not available and then it send 404. Thanks Abdul * kamailio * user authentication * user registration * user location * call routing * instant messaging and presence * freeswitch * voicemail * conference * SBC - this can be used for topology hiding, transcoding, prepaid or playing audio messages within calls * other media services (announcement, ivr, a.s.o) ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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

Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif &lt;asherif74(a)hotmail.com&gt; wrote:

Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data=&quot;{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1(a)domain.org&quot;/&gt; <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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

Thanks Sammy again, I will make adjustment let you know, thank you again for your help. Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Wednesday, February 10, 2016 10:23 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hi Abdul, Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this: <action application="bridge" data=&quot;sofia/external/$1(a)AbdulkamailioSIP.com&quot;/&gt; If you have saved your kamailio as a gateway then you can alternatively dial it as following: <action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/> Where GOOD_GATEWAY is the gateway name from an xml file. Here is how. FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/ FreeSWITCH-A:~# vim kamailio.xml Insert these Lines in this file: <include> <gateway name="GOOD_GATEWAY"> <param name="username" value="nothing"/> <param name="password" value="doesn't_matter"/> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --> <param name="register" value="false"/> <param name="retry-seconds" value="10"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="25"/> <param name="inbound-late-negotiation" value="true"/> <param name="context" value="default"/> </gateway> </include> Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/ Regards, Sammy On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org<mailto:1@domain.org>"/> <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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 _______________________________________________ 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

This is what I have for FS log debug if need be I will post it in FS site. ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Thursday, February 11, 2016 5:28 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hello Sammy, I used both the gateway method and external, the result is the same it goes the voicemail. I enabled debug on FS an should I post my question to FS? I followed the steps that was in kamailio to integrate kamailio and FS to setup SBC and that way I posted on kamailio site. Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Wednesday, February 10, 2016 10:23 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hi Abdul, Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this: <action application="bridge" data=&quot;sofia/external/$1(a)AbdulkamailioSIP.com&quot;/&gt; If you have saved your kamailio as a gateway then you can alternatively dial it as following: <action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/> Where GOOD_GATEWAY is the gateway name from an xml file. Here is how. FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/ FreeSWITCH-A:~# vim kamailio.xml Insert these Lines in this file: <include> <gateway name="GOOD_GATEWAY"> <param name="username" value="nothing"/> <param name="password" value="doesn't_matter"/> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --> <param name="register" value="false"/> <param name="retry-seconds" value="10"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="25"/> <param name="inbound-late-negotiation" value="true"/> <param name="context" value="default"/> </gateway> </include> Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/ Regards, Sammy On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org<mailto:1@domain.org>"/> <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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 _______________________________________________ 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

Thanks Sammy again, I just post the log debug. Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Thursday, February 11, 2016 5:41 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Share logs here as well, might help update the integration guide. Following are the major reasons why you'll fall into the voicemail application: 1 - FS failed to Dial to Kamailio, probably unable to reach Kamailio or syntax problem in the originate/bridge etc 2 - FS dialled to Kamailio but the route file is not properly setup to handle calls from FS and lookup() the user. 3 - Kamailio is setup correctly but the user is not online, or the lookup() don't have the user as FS required in uesrlocation table, or the end user doesn't accept the codecs. I mentioned the mismatch in domain part in RURI in one of my previous emails looking at your sip traces, you've already modified the packet but I still need to take a look at the sip captures to verify this. Thanks, Sammy On Thu, Feb 11, 2016 at 12:28 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello Sammy, I used both the gateway method and external, the result is the same it goes the voicemail. I enabled debug on FS an should I post my question to FS? I followed the steps that was in kamailio to integrate kamailio and FS to setup SBC and that way I posted on kamailio site. Thanks Abdul ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Wednesday, February 10, 2016 10:23 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hi Abdul, Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this: <action application="bridge" data=&quot;sofia/external/$1(a)AbdulkamailioSIP.com&quot;/&gt; If you have saved your kamailio as a gateway then you can alternatively dial it as following: <action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/> Where GOOD_GATEWAY is the gateway name from an xml file. Here is how. FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/ FreeSWITCH-A:~# vim kamailio.xml Insert these Lines in this file: <include> <gateway name="GOOD_GATEWAY"> <param name="username" value="nothing"/> <param name="password" value="doesn't_matter"/> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --> <param name="register" value="false"/> <param name="retry-seconds" value="10"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="25"/> <param name="inbound-late-negotiation" value="true"/> <param name="context" value="default"/> </gateway> </include> Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/ Regards, Sammy On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org<mailto:1@domain.org>"/> <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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 _______________________________________________ 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

While the full debug log is being approved, I just copy and paste some of the log. 2016-02-11 11:38:42.469315 [DEBUG] switch_core_codec.c:246 sofia/internal/102(a)newkama.AbdulKamailioSIP.com Restore previous codec PCMU:0. 2016-02-11 11:38:42.549341 [DEBUG] mod_voicemail.c:2806 Deliver VM to 101(a)10.22.52.2 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI: Processing for 101(a)10.22.52.2 in inbox 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1946 Update MWI: Messages Waiting yes 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1947 Update MWI: Update Reason NEW 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1948 Update MWI: Message Account 101(a)10.22.52.2 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1949 Update MWI: Voice Message 12/0 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:2901 sofia/internal/102(a)newkama.AbdulKamailioSIP.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State EXECUTE going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) Running State Change CS_HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) Callstate Change ACTIVE -> HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State HANGUP 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 sofia/internal/102(a)newkama.AbdulKamailioSIP.com Overriding SIP cause 480 with 904 from the other leg 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:413 Channel sofia/internal/102(a)newkama.AbdulKamailioSIP.com hanging up, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:60 sofia/internal/102(a)newkama.AbdulKamailioSIP.com Standard HANGUP, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State HANGUP going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State Change CS_HANGUP -> CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102(a)newkama.AbdulKamailioSIP.com [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) Running State Change CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104 sofia/internal/102(a)newkama.AbdulKamailioSIP.com Standard REPORTING, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State REPORTING going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State Change CS_REPORTING -> CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102(a)newkama.AbdulKamailioSIP.com [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) Locked, Waiting on external entities 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1641 Session 7 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) Ended 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)newkama.AbdulKamailioSIP.com [CS_DESTROY] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) Running State Change CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State DESTROY 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/102(a)newkama.AbdulKamailioSIP.com SOFIA DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111 sofia/internal/102(a)newkama.AbdulKamailioSIP.com Standard DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102(a)newkama.AbdulKamailioSIP.com) State DESTROY going to sleep ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Thursday, February 11, 2016 5:41 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Share logs here as well, might help update the integration guide. Following are the major reasons why you'll fall into the voicemail application: 1 - FS failed to Dial to Kamailio, probably unable to reach Kamailio or syntax problem in the originate/bridge etc 2 - FS dialled to Kamailio but the route file is not properly setup to handle calls from FS and lookup() the user. 3 - Kamailio is setup correctly but the user is not online, or the lookup() don't have the user as FS required in uesrlocation table, or the end user doesn't accept the codecs. I mentioned the mismatch in domain part in RURI in one of my previous emails looking at your sip traces, you've already modified the packet but I still need to take a look at the sip captures to verify this. Thanks, Sammy On Thu, Feb 11, 2016 at 12:28 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello Sammy, I used both the gateway method and external, the result is the same it goes the voicemail. I enabled debug on FS an should I post my question to FS? I followed the steps that was in kamailio to integrate kamailio and FS to setup SBC and that way I posted on kamailio site. Thanks Abdul ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Wednesday, February 10, 2016 10:23 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hi Abdul, Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this: <action application="bridge" data=&quot;sofia/external/$1(a)AbdulkamailioSIP.com&quot;/&gt; If you have saved your kamailio as a gateway then you can alternatively dial it as following: <action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/> Where GOOD_GATEWAY is the gateway name from an xml file. Here is how. FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/ FreeSWITCH-A:~# vim kamailio.xml Insert these Lines in this file: <include> <gateway name="GOOD_GATEWAY"> <param name="username" value="nothing"/> <param name="password" value="doesn't_matter"/> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --> <param name="register" value="false"/> <param name="retry-seconds" value="10"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="25"/> <param name="inbound-late-negotiation" value="true"/> <param name="context" value="default"/> </gateway> </include> Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/ Regards, Sammy On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org<mailto:1@domain.org>"/> <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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 _______________________________________________ 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

Thanks Sammy, I will use pastebin.com next as you recommended. Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Thursday, February 11, 2016 11:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Use pastebin.com<http://pastebin.com> or something ? [http://pastebin.com/i/facebook.png]<http://pastebin.com/> Pastebin.com - #1 paste tool since 2002!<http://pastebin.com/> pastebin.com Pastebin.com is the number one paste tool since 2002. Pastebin is a website where you can store text online for a set period of time. On Feb 11, 2016 18:32, "malik sherif" <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: While the full debug log is being approved, I just copy and paste some of the log. 2016-02-11 11:38:42.469315 [DEBUG] switch_core_codec.c:246 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Restore previous codec PCMU:0. 2016-02-11 11:38:42.549341 [DEBUG] mod_voicemail.c:2806 Deliver VM to 101@10.22.52.2<mailto:101@10.22.52.2> 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI: Processing for 101@10.22.52.2<mailto:101@10.22.52.2> in inbox 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1946 Update MWI: Messages Waiting yes 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1947 Update MWI: Update Reason NEW 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1948 Update MWI: Message Account 101@10.22.52.2<mailto:101@10.22.52.2> 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1949 Update MWI: Voice Message 12/0 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:2901 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State EXECUTE going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Callstate Change ACTIVE -> HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State HANGUP 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Overriding SIP cause 480 with 904 from the other leg 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:413 Channel sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> hanging up, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:60 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard HANGUP, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State HANGUP going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State Change CS_HANGUP -> CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard REPORTING, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State REPORTING going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State Change CS_REPORTING -> CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Locked, Waiting on external entities 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1641 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Ended 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [CS_DESTROY] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State DESTROY 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> SOFIA DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State DESTROY going to sleep ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, February 11, 2016 5:41 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Share logs here as well, might help update the integration guide. Following are the major reasons why you'll fall into the voicemail application: 1 - FS failed to Dial to Kamailio, probably unable to reach Kamailio or syntax problem in the originate/bridge etc 2 - FS dialled to Kamailio but the route file is not properly setup to handle calls from FS and lookup() the user. 3 - Kamailio is setup correctly but the user is not online, or the lookup() don't have the user as FS required in uesrlocation table, or the end user doesn't accept the codecs. I mentioned the mismatch in domain part in RURI in one of my previous emails looking at your sip traces, you've already modified the packet but I still need to take a look at the sip captures to verify this. Thanks, Sammy On Thu, Feb 11, 2016 at 12:28 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello Sammy, I used both the gateway method and external, the result is the same it goes the voicemail. I enabled debug on FS an should I post my question to FS? I followed the steps that was in kamailio to integrate kamailio and FS to setup SBC and that way I posted on kamailio site. Thanks Abdul ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Wednesday, February 10, 2016 10:23 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hi Abdul, Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this: <action application="bridge" data=&quot;sofia/external/$1(a)AbdulkamailioSIP.com&quot;/&gt; If you have saved your kamailio as a gateway then you can alternatively dial it as following: <action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/> Where GOOD_GATEWAY is the gateway name from an xml file. Here is how. FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/ FreeSWITCH-A:~# vim kamailio.xml Insert these Lines in this file: <include> <gateway name="GOOD_GATEWAY"> <param name="username" value="nothing"/> <param name="password" value="doesn't_matter"/> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --> <param name="register" value="false"/> <param name="retry-seconds" value="10"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="25"/> <param name="inbound-late-negotiation" value="true"/> <param name="context" value="default"/> </gateway> </include> Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/ Regards, Sammy On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org<mailto:1@domain.org>"/> <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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 _______________________________________________ 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 _______________________________________________ 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

Hello Sammy, Thanks again, here is my join me number: 685-184-730 Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Friday, February 12, 2016 8:01 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hey Abdul, Let me get this all done in my Virtual environment, use your cfg script. Make this work on my environment and get back to you on how to get this done. Alternatively you can just share screen via teamviewer or joinme and I may take a quick look and fix it for you. Regards, Sammy. On Fri, Feb 12, 2016 at 2:33 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Thanks Sammy, I will use pastebin.com<http://pastebin.com> next as you recommended. Thanks Abdul ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, February 11, 2016 11:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Use pastebin.com<http://pastebin.com> or something ? [http://pastebin.com/i/facebook.png]<http://pastebin.com/> Pastebin.com - #1 paste tool since 2002!<http://pastebin.com/> pastebin.com<http://pastebin.com> Pastebin.com is the number one paste tool since 2002. Pastebin is a website where you can store text online for a set period of time. On Feb 11, 2016 18:32, "malik sherif" <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: While the full debug log is being approved, I just copy and paste some of the log. 2016-02-11 11:38:42.469315 [DEBUG] switch_core_codec.c:246 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Restore previous codec PCMU:0. 2016-02-11 11:38:42.549341 [DEBUG] mod_voicemail.c:2806 Deliver VM to 101@10.22.52.2<mailto:101@10.22.52.2> 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI: Processing for 101@10.22.52.2<mailto:101@10.22.52.2> in inbox 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1946 Update MWI: Messages Waiting yes 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1947 Update MWI: Update Reason NEW 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1948 Update MWI: Message Account 101@10.22.52.2<mailto:101@10.22.52.2> 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1949 Update MWI: Voice Message 12/0 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:2901 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State EXECUTE going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Callstate Change ACTIVE -> HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State HANGUP 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Overriding SIP cause 480 with 904 from the other leg 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:413 Channel sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> hanging up, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:60 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard HANGUP, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State HANGUP going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State Change CS_HANGUP -> CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard REPORTING, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State REPORTING going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State Change CS_REPORTING -> CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Locked, Waiting on external entities 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1641 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Ended 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [CS_DESTROY] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State DESTROY 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> SOFIA DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State DESTROY going to sleep ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, February 11, 2016 5:41 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Share logs here as well, might help update the integration guide. Following are the major reasons why you'll fall into the voicemail application: 1 - FS failed to Dial to Kamailio, probably unable to reach Kamailio or syntax problem in the originate/bridge etc 2 - FS dialled to Kamailio but the route file is not properly setup to handle calls from FS and lookup() the user. 3 - Kamailio is setup correctly but the user is not online, or the lookup() don't have the user as FS required in uesrlocation table, or the end user doesn't accept the codecs. I mentioned the mismatch in domain part in RURI in one of my previous emails looking at your sip traces, you've already modified the packet but I still need to take a look at the sip captures to verify this. Thanks, Sammy On Thu, Feb 11, 2016 at 12:28 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello Sammy, I used both the gateway method and external, the result is the same it goes the voicemail. I enabled debug on FS an should I post my question to FS? I followed the steps that was in kamailio to integrate kamailio and FS to setup SBC and that way I posted on kamailio site. Thanks Abdul ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Wednesday, February 10, 2016 10:23 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hi Abdul, Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this: <action application="bridge" data=&quot;sofia/external/$1(a)AbdulkamailioSIP.com&quot;/&gt; If you have saved your kamailio as a gateway then you can alternatively dial it as following: <action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/> Where GOOD_GATEWAY is the gateway name from an xml file. Here is how. FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/ FreeSWITCH-A:~# vim kamailio.xml Insert these Lines in this file: <include> <gateway name="GOOD_GATEWAY"> <param name="username" value="nothing"/> <param name="password" value="doesn't_matter"/> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --> <param name="register" value="false"/> <param name="retry-seconds" value="10"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="25"/> <param name="inbound-late-negotiation" value="true"/> <param name="context" value="default"/> </gateway> </include> Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/ Regards, Sammy On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org<mailto:1@domain.org>"/> <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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 _______________________________________________ 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 _______________________________________________ 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

Thanks for your help Sammy, the call is working, and I will share later about the change that fix this problem ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Friday, February 12, 2016 9:13 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hello Sammy, Do you have my kamailio cfg file? Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of malik sherif &lt;asherif74(a)hotmail.com&gt; Sent: Friday, February 12, 2016 9:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hello Sammy, Thanks again, here is my join me number: 685-184-730 Thanks Abdul ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.sip-router.org&gt; on behalf of SamyGo &lt;govoiper(a)gmail.com&gt; Sent: Friday, February 12, 2016 8:01 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hey Abdul, Let me get this all done in my Virtual environment, use your cfg script. Make this work on my environment and get back to you on how to get this done. Alternatively you can just share screen via teamviewer or joinme and I may take a quick look and fix it for you. Regards, Sammy. On Fri, Feb 12, 2016 at 2:33 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Thanks Sammy, I will use pastebin.com<http://pastebin.com> next as you recommended. Thanks Abdul ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, February 11, 2016 11:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Use pastebin.com<http://pastebin.com> or something ? [http://pastebin.com/i/facebook.png]<http://pastebin.com/> Pastebin.com - #1 paste tool since 2002!<http://pastebin.com/> pastebin.com<http://pastebin.com> Pastebin.com is the number one paste tool since 2002. Pastebin is a website where you can store text online for a set period of time. On Feb 11, 2016 18:32, "malik sherif" <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: While the full debug log is being approved, I just copy and paste some of the log. 2016-02-11 11:38:42.469315 [DEBUG] switch_core_codec.c:246 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Restore previous codec PCMU:0. 2016-02-11 11:38:42.549341 [DEBUG] mod_voicemail.c:2806 Deliver VM to 101@10.22.52.2<mailto:101@10.22.52.2> 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI: Processing for 101@10.22.52.2<mailto:101@10.22.52.2> in inbox 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1946 Update MWI: Messages Waiting yes 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1947 Update MWI: Update Reason NEW 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1948 Update MWI: Message Account 101@10.22.52.2<mailto:101@10.22.52.2> 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1949 Update MWI: Voice Message 12/0 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:2901 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State EXECUTE going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Callstate Change ACTIVE -> HANGUP 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State HANGUP 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Overriding SIP cause 480 with 904 from the other leg 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:413 Channel sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> hanging up, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:60 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard HANGUP, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State HANGUP going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State Change CS_HANGUP -> CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State REPORTING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard REPORTING, cause: NORMAL_CLEARING 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State REPORTING going to sleep 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State Change CS_REPORTING -> CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [BREAK] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Locked, Waiting on external entities 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1641 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Ended 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> [CS_DESTROY] 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) Running State Change CS_DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State DESTROY 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> SOFIA DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111 sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com> Standard DESTROY 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com<mailto:102@newkama.AbdulKamailioSIP.com>) State DESTROY going to sleep ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, February 11, 2016 5:41 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Share logs here as well, might help update the integration guide. Following are the major reasons why you'll fall into the voicemail application: 1 - FS failed to Dial to Kamailio, probably unable to reach Kamailio or syntax problem in the originate/bridge etc 2 - FS dialled to Kamailio but the route file is not properly setup to handle calls from FS and lookup() the user. 3 - Kamailio is setup correctly but the user is not online, or the lookup() don't have the user as FS required in uesrlocation table, or the end user doesn't accept the codecs. I mentioned the mismatch in domain part in RURI in one of my previous emails looking at your sip traces, you've already modified the packet but I still need to take a look at the sip captures to verify this. Thanks, Sammy On Thu, Feb 11, 2016 at 12:28 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello Sammy, I used both the gateway method and external, the result is the same it goes the voicemail. I enabled debug on FS an should I post my question to FS? I followed the steps that was in kamailio to integrate kamailio and FS to setup SBC and that way I posted on kamailio site. Thanks Abdul ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Wednesday, February 10, 2016 10:23 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Hi Abdul, Kindly share the whole FS console logs (enable sip debug inside the logs too) , can you modify the bridge statement as this: <action application="bridge" data=&quot;sofia/external/$1(a)AbdulkamailioSIP.com&quot;/&gt; If you have saved your kamailio as a gateway then you can alternatively dial it as following: <action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/> Where GOOD_GATEWAY is the gateway name from an xml file. Here is how. FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/ FreeSWITCH-A:~# vim kamailio.xml Insert these Lines in this file: <include> <gateway name="GOOD_GATEWAY"> <param name="username" value="nothing"/> <param name="password" value="doesn't_matter"/> <param name="proxy" value="192.168.30.3"/> <!--SET IP OF KAMAILIO HERE --> <param name="register" value="false"/> <param name="retry-seconds" value="10"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="25"/> <param name="inbound-late-negotiation" value="true"/> <param name="context" value="default"/> </gateway> </include> Also, if you don't use gateway approach can you make sure that from your FS the domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server. I've a feeling that this email should be in Freeswitch mailing list, not in Kamailio's/ Regards, Sammy On Wed, Feb 10, 2016 at 5:00 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Hello, I am using Kamailio and freeswitch to setup SBC but the I attempted to make a call it just goes to the voice mail. Here is what freeswitch is displaying. Thanks for your help in advance Abdul freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulKamailioSIP.com [12f87c10-f3be-43ee-b038-f6647e5af373] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context public 2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer sofia/internal/102(a)AbdulKamailioSIP.com to XML[kb-102@default] 2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 <102>->kb-102 in context default 2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/102(a)AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102(a)AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102(a)AbdulkamailioSIP.com) Ended 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulkamailioSIP.com [CS_DESTROY] 2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/102(a)AbdulKamailioSIP.com! 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102(a)AbdulKamailioSIP.com] has been answered 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING] 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102(a)AbdulKamailioSIP.com) Ended 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102(a)AbdulKamailioSIP.com [CS_DESTROY] Any idea as to how to implement this command on freeswitch dial plan, I am not sure what to use for gw1 <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org<mailto:1@domain.org>"/> <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="call_timeout=50"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=AbdulkamailioSIP.com"/> <action application="export" data="sip_contact_user=ufs"/> <action application="bridge" data=&quot;sofia/$${domain}/$1(a)AbdulkamailioSIP.com&quot;/&gt; <action application="answer"/> <action application="voicemail" data="default ${domain_name} $1"/> </condition> </extension> ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Friday, January 29, 2016 5:02 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC Sorry for last email: if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } That is where you get 404 Not Found. What I see is that you're registering users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call to Kamailio the RURI becomes: INVITE sip:7632689993@10.22.52.2<mailto:sip%3A7632689993@10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com SIP/2.0 So, you need to go in your FS dialplan and make sure you set the proper Domains before sending call out, there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP header from FS to contain a domain name, and in Kamailio set headers as you require; something like this: Attach a SIP Header in FS dialplan before sending call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I receive call in Kamailio.cfg I detect this header if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + $hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it before executing record_route() functions, so possibly need to do this inside your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom SIP headers in sky blue, that doesn't mean there is any error in there. Regards, Sammy On Fri, Jan 29, 2016 at 11:47 AM, SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote: Hi Abdul, This is where you are getting your 404 NOT Found from Kamailio: On Thu, Jan 28, 2016 at 4:30 PM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: I will also run the commands that suggested. ________________________________ From: sr-users <sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>> on behalf of SamyGo <govoiper@gmail.com<mailto:govoiper@gmail.com>> Sent: Thursday, January 28, 2016 6:08 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC I believe Daniel is busy with FOSDEM , Abdul can you confirm that you're still getting this output in FS console: 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<mailto: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<mailto:7632689993@10.22.52.2> [CS_ROUTING] [UNALLOCATED_NUMBER] Please paste your complete dialplan here as well, though this clearly states that the number it tried to dial is not registered or unable to dial to. please paste out the content of the following command just before dialing: fs_cli> show registrations Also, it will help you find out useful info about why it shows you UNALLOCATED NUMBER if you enable the sofia sip debug by using the following command. fs_cli> sofia global siptrace on Once you execute the above command make a call to destination and see what FreeeSWITCH is trying to do. Thanks, Sammy. On Thu, Jan 28, 2016 at 11:23 AM, malik sherif <asherif74@hotmail.com<mailto:asherif74@hotmail.com>> wrote: Any hint? ________________________________ 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: Tuesday, January 26, 2016 11:35 PM To: Kamailio (SER) - Users Mailing List; miconda@gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Thanks again and here is the pcap file. Thanks Abdul ________________________________ From: Daniel-Constantin Mierla <miconda@gmail.com<mailto:miconda@gmail.com>> Sent: Friday, January 22, 2016 8:46 AM To: malik sherif; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Can you attach the pcap file - copy&paste inline makes it imposible to read and digest it with a traffic analyzer (e.g., wireshark). Cheers, Daniel On 21/01/16 18:31, malik sherif wrote: ________________________________ 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: Wednesday, January 20, 2016 9:55 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the server IP address Thanks again Abdul [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswit… -- 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 _______________________________________________ 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 _______________________________________________ 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 _______________________________________________ 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

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: <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