Ok, I found what's happening. I probably did some change on the wiki to start reflect the new configuration, without having time enough to check the configuration. There is a mistake in your dialplan configuration. You should put this instead:
wrong line: <action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/> please correct with: <action application="set" data="ROUTE_GW=${sip_redirect_contact_host_0}"/> This should match PEER_01 in dialplan instead of trying matching fra...@peer_01. Let me know if this is right now. rod Hristo Benev a écrit : > Hello Rod, > > I did the change. > > Here is extract of console: > ----------------------------------------------------------------------------------------- > variable_continue_on_fail: [true] > variable_sip_h_X-ROUTE: [LOOKUP] > variable_export_vars: [sip_h_X-ROUTE] > variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20] > variable_sip_redirect_contact_0: [sip:fra...@peer_01] > variable_sip_redirected_to: [sip:fra...@peer_01] > variable_sip_redirect_contact_user_0: [France] > <------------------------------------------- > variable_sip_redirect_contact_host_0: [PEER_01] > variable_sip_redirect_dialstring_0: [sofia/internal/sip:fra...@peer_01] > variable_sip_redirect_dialstring: [sofia/internal/sip:fra...@peer_01] > variable_proto_specific_hangup_cause: [sip:503] > variable_sip_hangup_phrase: [DNS Error] > variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE] > variable_ROUTE_GW: [France] > variable_AREA: [France] > variable_current_application: [info] > ------------------------------------------------------------------------------------------ > I have different value it is actually the description field as shown here: > > -------------------------------- > /opt/kamailio/sbin/kamctl cr show > cr carrier names > +----+---------+ > | id | carrier | > +----+---------+ > | 1 | default | > +----+---------+ > cr domain names > +----+---------+ > | id | domain | > +----+---------+ > | 1 | default | > +----+---------+ > cr routes > +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ > | id | carrier | domain | scan_prefix | flags | mask | prob | strip | > rewrite_host | rewrite_prefix | rewrite_suffix | description | > +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ > | 1 | 1 | 1 | 1000 | 0 | 0 | 1 | 0 | PEER_01 > | | | France | > +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ > ----------------------------------------- > > And here is what I have in kamailio: > ------------------------------------------------------------------------------------- > ####### Routing Logic ######## > > > # main request routing logic > > route{ > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > > t_check_trans(); > > if ($rU==NULL) { > # request with no Username in RURI > sl_send_reply("484","Address Incomplete"); > exit; > } > > # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP > if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){ > if(!cr_route("default", "default", "$rU", "$rU", > "call_id","$avp(s:route_desc)")){ > #xlog("ROUTING FAILED: no route found for $rU"); > sl_send_reply("604", "Unable to route this call"); > exit; > } else { > xlog("LOOKUP FOUND: $rd $avp(s:route_desc)"); > avp_pushto("$ru/username", "$avp(s:route_desc)"); > sl_send_reply("302", "$rd"); > exit; > } > } > } > ---------------------------------------------------------------------------------- > > Another question... > In that part of FreeSwitch dialplan.xml > > ----------------------------------- > <context name="ROUTING"> > > <extension name="PEER_01"> > <condition field="${sip_h_X-ROUTE}" expression="PEER_01"> > <action application="set" data="hangup_after_bridge=true"/> > > ----------------------------------- > > X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the > regex is true. > > > As for thanks - for sure by default they are also for the developers of both > apps. > I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning > so it will be nice to have something working to start with. > > > > >-------- Оригинално писмо -------- > >От: rod > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route > >До: freeswitch-users@lists.freeswitch.org > >Изпратено на: Петък, 2009, Август 28 09:54:00 EEST > > >Hello, > > > >the trace seems good. > >If you check the answer from Kamailio, you'll see that Kamailio answers > >with "302 PEER_01". > > > >As Michael Collins stated before, you can get the variable containing > >"PEER_01", then this variable is stored in a custom variable. > > > >In your dialplan, may you please add: > > > >, just before the transfer line, eg: > > > > > > > > > > > > > > > > > > > > > > > > > > > >Using application Info, you'll see on the console (or CDR) the list of > >variables used for this call. You should see the content of > >"${sip_redirect_contact_user_0}" that should contain the value > >"PEER_01". Please check this and let me know. > > > >For the new configuration file, no problem for sharing. But as I wrote > >on the wiki page, I worked on this setup cause LCR module was not > >available when I start working on FS, nor I'm a good programmer to write > >a server side HTTP script (used by xml_curl) that could scale to my needs. > > > >I enhanced a bit this configuration with support for fallback routing, > >almost realtime graph (every minutes using www.cacti.net and some > >functions in FS like limit_hash) of number of concurrent calls per AREA, > >PEER... > > > >Thanks for "the good tutorial", but don't forget the dev team who did > >this great product ;-) > > > >rod. > > > > > >Hristo Benev a écrit : > >> I assume you asked for port 5062 since I do not have any traffic on 5060 > (I have one IP and my internal sip port is 5090 and external 5080). > >> > >> If you need additional info I'll provide it. > >> > >> Here is trace: > >> > >> ngrep -d any -nn -i '1000' port 5062 -W byline > >> interface: any > >> filter: (ip or ip6) and ( port 5062 ) > >> match: 1000 > >> # > >> U 10.10.10.10:5090 -> 127.0.0.1:5062 > >> INVITE sip:1...@127.0.0.1:5062 SIP/2.0. > >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj. > >> Max-Forwards: 69. > >> From: "Extension 1001" ;tag=978g69jZaFpBD. > >> To: . > >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. > >> CSeq: 119574771 INVITE. > >> Contact: . > >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > >> Supported: timer, precondition, path, replaces. > >> Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > >> Content-Type: application/sdp. > >> Content-Disposition: session. > >> Content-Length: 429. > >> X-ROUTE: LOOKUP. > >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. > >> . > >> v=0. > >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10. > >> s=FreeSWITCH. > >> c=IN IP4 10.10.10.10. > >> t=0 0. > >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:115 G7221/32000. > >> a=fmtp:115 bitrate=48000. > >> a=rtpmap:107 G7221/16000. > >> a=fmtp:107 bitrate=32000. > >> a=rtpmap:9 G722/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:3 GSM/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=rtpmap:13 CN/8000. > >> a=ptime:20. > >> > >> # > >> U 127.0.0.1:5062 -> 10.10.10.10:5090 > >> SIP/2.0 302 PEER_01. > >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj. > >> From: "Extension 1001" ;tag=978g69jZaFpBD. > >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8. > >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. > >> CSeq: 119574771 INVITE. > >> Contact: sip:fra...@peer_01. > >> Server: Kamailio (1.5.2-notls (i386/linux)). > >> Content-Length: 0. > >> . > >> > >> # > >> U 10.10.10.10:5090 -> 127.0.0.1:5062 > >> ACK sip:1...@127.0.0.1:5062 SIP/2.0. > >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj. > >> Max-Forwards: 69. > >> From: "Extension 1001" ;tag=978g69jZaFpBD. > >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8. > >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. > >> CSeq: 119574771 ACK. > >> Content-Length: 0. > >> . > >> > >> > >> > >> > >> > >> >-------- Оригинално писмо -------- > >> >От: Hristo Benev > >> >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no > route > >> >До: freeswitch-users@lists.freeswitch.org > >> >Изпратено на: Четвъртък, 2009, Август 27 14:58:53 EEST > >> > >> > > >> >Bojnour, > >> > > >> >I'll send a trace ASAP. > >> > > >> >What I see is that SIP header does not get updated -> regex is not true > then it does not go to the peer. (I assume that is coming from kamailio > config) > >> > > >> >I'm really interested to see the updates of the project. > >> > > >> >Thank you for the good tutorial. > >> > > >> >Hristo > >> > > >> > > >> > >-------- Оригинално писмо -------- > >> > >От: rod > >> > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no > route > >> > >До: freeswitch-users@lists.freeswitch.org > >> > >Изпратено на: Четвъртък, 2009, Август 27 09:41:05 EEST > >> > > >> > >Hi Hristo, > >> > > > >> > >I'm the author of this setup and wiki page. I did a lot of > modifications > >> > >on this setup (alternative routing if failure essentially) but don't > >> > >have too much time to update the wiki. > >> > > > >> > >May you please send me an ngrep trace when you call 1000: > >> > > > >> > >ngrep -d any -nn -i '1000' port 5060 -W byline > >> > > > >> > >I will check what's happening. > >> > >Do you have an entry for 1000 in your mysql database ? > >> > > > >> > >regards, > >> > >rod > >> > > > >> > >Hristo Benev a écrit : > >> > >> > >> > >> It seems that the problem is on kamailio configuration. > >> > >> Will ask on their list. > >> > >> > >> > >> But to test i try to connect to my asterisk server and i receive > 407 proxy authentication required. > >> > >> > >> > >> I have it setup as friend in asterisk, but still ??? > >> > >> > >> > >> Any ideas? > >> > >> > >> > >> Thanks, > >> > >> > >> > >> > >> > >> >-------- Оригинално писмо -------- > >> > >> >От: Hristo Benev > >> > >> >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - > no route > >> > >> >До: freeswitch-users@lists.freeswitch.org > >> > >> >Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST > >> > >> > >> > >> > I think that the problem is here: > >> > >> >------------------------- > >> > >> >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 > Processing 1001->1000 in context ROUTING > >> > >> >Dialplan: sofia/internal/1...@209.71.254.33 parsing > [ROUTING->PEER_01] continue=false > >> > >> >Dialplan: sofia/internal/1...@209.71.254.33 Regex (FAIL) > [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false > >> > >> >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 > No Route, Aborting > >> > >> >-------------------------- > >> > >> > > >> > >> >Actually Regex FAIL > >> > >> > > >> > >> >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} > should be PEER_01 for success? > >> > >> >Here is my default.xml: > >> > >> >---------------- > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> > > >> > >> >-------------------------- > >> > >> > > >> > >> > > >> > >> > > >> > >> > >-------- Оригинално писмо -------- > >> > >> > >От: Brian West > >> > >> > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio > - no route > >> > >> > >До: freeswitch-users@lists.freeswitch.org > >> > >> > >Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST > >> > >> > > >> > >> > >We do not blindly follow 302's as that is a dangerous thing to > do. > >> > >> >You have to process all 302's in the dialplan. > >> > >> >Set this on your sofia profile > >> > >> >You can set these variables sip_redirect_profile, > >> > >> >sip_redirect_context, > >> > >> >sip_redirect_dialplan, > >> > >> >When a redirect happens you get these variables - > sip_redirect_contact_%d, > >> > >> >sip_redirected_to, > >> > >> >sip_redirect_contact_user_%d, > >> > >> >sip_redirect_contact_host_%d, > >> > >> >sip_redirect_contact_params_%d, > >> > >> >sip_redirect_dialstring_%d, > >> > >> >sip_redirect_dialstring, > >> > >> >sip_redirected_byThen its up to you to process the redirect in > your dialplan, If you don't set the > >> > >> >sip_redirect_context then it'll default to redirected context and > XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI > followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have > following problem when I dial 1000 Kamalio reports 302, but freeswitch does > not routeWhere to look for problems? > >> > >> > > > >> > >> > > >> > >> >_______________________________________________ > >> > >> >FreeSWITCH-users mailing list > >> > >> >FreeSWITCH-users@lists.freeswitch.org > >> > >> >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> >http://www.freeswitch.org > >> > >> > > >> > >> > >> > >> _______________________________________________ > >> > >> FreeSWITCH-users mailing list > >> > >> FreeSWITCH-users@lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> > > > >> > >_______________________________________________ > >> > >FreeSWITCH-users mailing list > >> > >FreeSWITCH-users@lists.freeswitch.org > >> > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >http://www.freeswitch.org > >> > > > >> > > >> >_______________________________________________ > >> >FreeSWITCH-users mailing list > >> >FreeSWITCH-users@lists.freeswitch.org > >> >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >http://www.freeswitch.org > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users@lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > >_______________________________________________ > >FreeSWITCH-users mailing list > >FreeSWITCH-users@lists.freeswitch.org > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org