Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
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:1000@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:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@127.0.0.1:5062>. Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. CSeq: 119574771 INVITE. Contact: <sip:mod_sofia@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:1001@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:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8. Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. CSeq: 119574771 INVITE. Contact: sip:France@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:1000@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:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@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/1001@209.71.254.33 parsing [ROUTING->PEER_01] continue=false Dialplan: sofia/internal/1001@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 ${sip_h_X-ROUTE} should be PEER_01 for success? Here is my default.xml: ----------------
--------------------------
-------- Оригинално писмо -------- От: Brian West Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route До: freeswitch-users@lists.freeswitch.org Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST
We do not blindly follow 302's as that is a dangerous thing to do. You have to process all 302's in the dialplan. Set this on your sofia profile You can set these variables sip_redirect_profile, sip_redirect_context, sip_redirect_dialplan, When a redirect happens you get these variables - sip_redirect_contact_%d, sip_redirected_to, sip_redirect_contact_user_%d, sip_redirect_contact_host_%d, sip_redirect_contact_params_%d, sip_redirect_dialstring_%d, sip_redirect_dialstring, sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
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Hello, the trace seems good. If you check the answer from Kamailio, you'll see that Kamailio answers with "302 PEER_01". As Michael Collins stated before, you can get the variable containing "PEER_01", then this variable is stored in a custom variable. In your dialplan, may you please add: <action application="info"/>, just before the transfer line, eg: <condition field="destination_number" expression="(\d+)$"> <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/${destination_number}@127.0.0.1:5062"/> <action application="set" data="ROUTE_GW=${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:1000@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:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@127.0.0.1:5062>. Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. CSeq: 119574771 INVITE. Contact: <sip:mod_sofia@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:1001@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:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8. Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. CSeq: 119574771 INVITE. Contact: sip:France@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:1000@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:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@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/1001@209.71.254.33 parsing [ROUTING->PEER_01] continue=false Dialplan: sofia/internal/1001@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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participants (2)
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Hristo Benev
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rod