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 >> >> >
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