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: <action application="info"/>, just before the transfer line, eg: <condition field="destination_number" expression="(\d+)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <action application="bridge" data="sofia/internal/${destination_numb...@127.0.0.1:5062"/> <action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/> <action application="set" data="AREA=${sip_redirect_contact_user_0}"/> <action application="info"/> <action application="transfer" data="${destination_number} XML ROUTING"/> </condition> Using application Info, you'll see on the console (or CDR) the list of variables used for this call. You should see the content of "${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" <sip:1...@10.10.10.10>;tag=978g69jZaFpBD. > To: <sip:1...@127.0.0.1:5062>. > Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. > CSeq: 119574771 INVITE. > Contact: <sip:mod_so...@10.10.10.10:5090>. > 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" > <sip:1...@10.10.10.10>;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" <sip:1...@10.10.10.10>;tag=978g69jZaFpBD. > To: <sip:1...@127.0.0.1:5062>;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" <sip:1...@10.10.10.10>;tag=978g69jZaFpBD. > To: <sip:1...@127.0.0.1:5062>;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? 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