[Freeswitch-users] freeswitch as SBC and kamailio - no route

Hristo Benev foxb at abv.bg
Thu Aug 27 08:23:00 PDT 2009


 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 at 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 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: <sip:mod_sofia at 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 at 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 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:France at 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 at 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 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 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 at 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 at 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 at 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 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 > >>  >Dialplan: sofia/internal/1001 at 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:
 > >>  >----------------
 > >>  >
 > >>  >
 > >>  >
 > >>  >  
 > >>  >
 > >>  >     
 > >>  >      
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >      
 > >>  >    
 > >>  >
 > >>  >   
 > >>  >
 > >>  >  
 > >>  >
 > >>  >    
 > >>  >      
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
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 > >>  >
 > >>  >--------------------------
 > >>  >
 > >>  >
 > >>  >
 > >>  > >-------- Оригинално писмо --------
 > >>  > >От:  Brian West 
 > >>  > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >>  > >До: freeswitch-users at 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
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 > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >>  >http://www.freeswitch.org
 > >>  >
 > >>
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 > >>   
 > >
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