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="sofia/*external*/$1@ AbdulkamailioSIP.com"/> 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 <asheri...@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/1...@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/1...@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/1...@abdulkamailiosip.com > [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3] > 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup > sofia/internal/1...@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/1...@abdulkamailiosip.com) Ended > 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close > Channel sofia/internal/1...@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/1...@abdulkamailiosip.com! > 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel > [sofia/internal/1...@abdulkamailiosip.com] has been answered > 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup > sofia/internal/1...@abdulkamailiosip.com [CS_EXECUTE] [NORMAL_CLEARING] > 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 > (sofia/internal/1...@abdulkamailiosip.com) Ended > 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close > Channel sofia/internal/1...@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/$1...@domain.org"/> > > > > > > From Freeswitch dial plan > > > <extension name="kbridge"> > <condition field="destination_number" expression="^kb-(.+)$"> > <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="sofia/$${domain}/$1...@abdulkamailiosip.com"/> > <action application="answer"/> > <action application="voicemail" data="default > ${domain_name} $1"/> > </condition> > </extension> > > > > > > > ------------------------------ > *From:* sr-users <sr-users-boun...@lists.sip-router.org> on behalf of > SamyGo <govoi...@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 > <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 <govoi...@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 <asheri...@hotmail.com> >> wrote: >> >>> I will also run the commands that suggested. >>> >>> >>> ------------------------------ >>> *From:* sr-users <sr-users-boun...@lists.sip-router.org> on behalf of >>> SamyGo <govoi...@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 >>> [d52b6ef9-c4f6-4edf-aff9-8a8da3761788] >>> 2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/ >>> 7632689993@10.22.52.2 [CS_ROUTING] [UNALLOCATED_NUMBER] >>> >>> 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 <asheri...@hotmail.com> >>> wrote: >>> >>>> >>>> Any hint? >>>> >>>> ------------------------------ >>>> *From:* sr-users <sr-users-boun...@lists.sip-router.org> on behalf of >>>> malik sherif <asheri...@hotmail.com> >>>> *Sent:* Tuesday, January 26, 2016 11:35 PM >>>> *To:* Kamailio (SER) - Users Mailing List; mico...@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 <mico...@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-boun...@lists.sip-router.org> >>>> <sr-users-boun...@lists.sip-router.org> on behalf of malik sherif >>>> <asheri...@hotmail.com> <asheri...@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/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - >>>> http://www.linkedin.com/in/miconda >>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> 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 >>> 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 > 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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users