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