Use pastebin.com or something ? On Feb 11, 2016 18:32, "malik sherif" <asheri...@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/ > 1...@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 > 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI: > Processing for 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 > 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/1...@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/1...@newkama.abdulkamailiosip.com) State EXECUTE going to > sleep > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 > (sofia/internal/1...@newkama.abdulkamailiosip.com) Running State Change > CS_HANGUP > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:735 > (sofia/internal/1...@newkama.abdulkamailiosip.com) Callstate Change ACTIVE > -> HANGUP > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 > (sofia/internal/1...@newkama.abdulkamailiosip.com) State HANGUP > 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 sofia/internal/ > 1...@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/ > 1...@newkama.abdulkamailiosip.com hanging up, cause: NORMAL_CLEARING > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/1...@newkama.abdulkamailiosip.com Standard HANGUP, cause: > NORMAL_CLEARING > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 > (sofia/internal/1...@newkama.abdulkamailiosip.com) State HANGUP going to > sleep > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1...@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/1...@newkama.abdulkamailiosip.com [BREAK] > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 > (sofia/internal/1...@newkama.abdulkamailiosip.com) Running State Change > CS_REPORTING > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 > (sofia/internal/1...@newkama.abdulkamailiosip.com) State REPORTING > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/1...@newkama.abdulkamailiosip.com Standard REPORTING, > cause: NORMAL_CLEARING > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 > (sofia/internal/1...@newkama.abdulkamailiosip.com) State REPORTING going > to sleep > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:498 > (sofia/internal/1...@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/1...@newkama.abdulkamailiosip.com [BREAK] > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7 > (sofia/internal/1...@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/1...@newkama.abdulkamailiosip.com) Ended > 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close > Channel sofia/internal/1...@newkama.abdulkamailiosip.com [CS_DESTROY] > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626 > (sofia/internal/1...@newkama.abdulkamailiosip.com) Running State Change > CS_DESTROY > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 > (sofia/internal/1...@newkama.abdulkamailiosip.com) State DESTROY > 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/ > 1...@newkama.abdulkamailiosip.com SOFIA DESTROY > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111 > sofia/internal/1...@newkama.abdulkamailiosip.com Standard DESTROY > 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 > (sofia/internal/1...@newkama.abdulkamailiosip.com) State DESTROY going to > sleep > > > > ------------------------------ > *From:* sr-users <sr-users-boun...@lists.sip-router.org> on behalf of > SamyGo <govoi...@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 <asheri...@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-boun...@lists.sip-router.org> on behalf of >> SamyGo <govoi...@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="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 >> >> > > _______________________________________________ > 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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