Florian,
I had the same issue using Asterisk and Kamailio. Basically the problem is
that Asterisk interprets the call as a Loop (in the worst case) or might
continue resolving the call locally without taking care of the changes you
made in Kamaili.
I solved my issue using 3XX (redirect) messages.
Example:
sl_send_reply("301", "Go Here");
Try that on teh first place, then you can continue using Asterisk's dialplan
to change message details.
Cheers,
Uriel
On Tue, Dec 1, 2009 at 11:39 AM, Florian Meister <
florian.meis...@teleport.vol.at> wrote:
> Hi,
>
> basically I'm using this structure at the moment:
>
> SIP Users <----> Kamailio <-----> Asterisk <-----> PSTN
>
> I have to add a diversion-functionality at kamailio-level, so to simply
> rewrite $ru with something else defined in the database. That's working
> without problems. For billing issues, I also have to add a Remote-Party-ID
> header, set to the SIP user, which initiated the redirect in the database.
>
> Now to the problem:
>
> When a call is coming from PSTN, it's passing the asterisk server, then at
> kamailio level $ru is rewritten and sent back to asterisk (I'm talking about
> a redirect to a number in PSTN here)
>
> What I've seen from the logs is, that asterisk is seeing that it gets an
> invite back with the same call-id, and therefore it cancels the original
> invite and handles the whole call internally via the Local Channel. The
> Problem is, that in the invite sent from kamailio back to asterisk, I've set
> a Remote-Party-ID header to tell asterisk to set the Callerid correctly for
> billing purposes. Now it seems that asterisk is _ignoring_ this header from
> the second invite.
>
> So is this an expected behavior ? If yes, how to do it correctly ?
>
> Below you can see the verbose output of asterisk. Since the call is handled
> at "Local" Channels the function to read sip headers does not work. The only
> message I get is "thanks to SIP/tpsiptestproxyu01-00d0a0b8".
>
> -- Called tpsiptestproxyu01/+435572949012
> -- Now forwarding DAHDI/2-1 to 'Local/066480588...@from-internal'
> (thanks to SIP/tpsiptestproxyu01-00d0a0b8)
> -- Executing [066480588...@from-internal:1]
> NoOp("Local/066480588...@from-internal-d69e;2", "435572501134") in new
> stack
> [Dec 1 15:14:50] WARNING[20506]: chan_sip.c:15797 func_header_read: This
> function can only be used on SIP channels.
> -- Executing [066480588...@from-internal:2]
> NoOp("Local/066480588...@from-internal-d69e;2", "") in new stack
> -- Executing [066480588...@from-internal:3]
> Dial("Local/066480588...@from-internal-d69e;2", "DAHDI/G0/066480588134")
> in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called G0/066480588134
> -- DAHDI/124-1 is proceeding passing it to
> Local/066480588...@from-internal-d69e;2
> -- Local/066480588...@from-internal-d69e;1 is proceeding passing it to
> DAHDI/2-1
>
> In the SIP debug you can see that asterisk is cancelling the dialog with
> kamailio and doing it itself:
>
> 13:49:30.589054 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length:
> 938
> e.......@..l..,L..,N........INVITE sip:+435572949...@[--kamailio--]
> SIP/2.0
> Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport
> Max-Forwards: 70
> From: "435572501134" <sip:435572501...@[--asterisk--]>;tag=as27658014
> To: <sip:+435572949...@[--kamailio--]>
> Contact: <sip:435572501...@[--asterisk--]>
> Call-ID: 4bbee84339a9e2d30850185317983...@[--asterisk--]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Remote-Party-ID: "435572501134" <sip:435572501134@
> [--ASTERISK--]>;privacy=off;screen=yes
> Date: Tue, 01 Dec 2009 12:49:30 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 263
>
> v=0
> o=root 930830518 930830518 IN IP4 [--ASTERISK--]
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 [--ASTERISK--]
> t=0 0
> m=audio 16924 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 13:49:30.591037 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length:
> 342
> e.....@.@.[0..,N..,L.....^.aSIP/2.0 100 Trying
> Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060
> From: "435572501134" <sip:435572501...@[--asterisk--]>;tag=as27658014
> To: <sip:+435572949...@[--kamailio--]>
> Call-ID: 4bbee84339a9e2d30850185317983...@[--asterisk--]
> CSeq: 102 INVITE
> Server: OpenSER (1.3.2-notls (x86_64/linux))
> Content-Length: 0
>
>
> 13:49:30.594345 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length:
> 1093
> e.....@.@.XA..,N..,L.....M+LINVITE
> sip:066480588...@[--asterisk--]:5060;transport=udp
> SIP/2.0
> Record-Route: <sip:[--KAMAILIO--];lr;ftag=as27658014>
> Via: SIP/2.0/UDP [--KAMAILIO--];branch=z9hG4bK06.05390227.0
> Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060
> Max-Forwards: 69
> From: "435572501134" <sip:435572501...@[--asterisk--]>;tag=as27658014
> To: <sip:+435572949...@[--kamailio--]>
> Contact: <sip:435572501...@[--asterisk--]>
> Call-ID: 4bbee84339a9e2d30850185317983...@[--asterisk--]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Tue, 01 Dec 2009 12:49:30 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 263
> Remote-Party-ID: "435572949012"
> <sip:435572949...@tpseru01.tele.net<sip%3a435572949...@tpseru01.tele.net>
> >;party=caller;privacy=none;screen=yes
>
> v=0
> o=root 930830518 930830518 IN IP4 [--ASTERISK--]
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 [--ASTERISK--]
> t=0 0
> m=audio 16924 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 13:49:30.594605 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length:
> 460
> e.......@..)..,L..,N....... CANCEL sip:+435572949...@[--kamailio--]
> SIP/2.0
> Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport
> Max-Forwards: 70
> From: "435572501134" <sip:435572501...@[--asterisk--]>;tag=as27658014
> To: <sip:+435572949...@[--kamailio--]>
> Call-ID: 4bbee84339a9e2d30850185317983...@[--asterisk--]
> CSeq: 102 CANCEL
> User-Agent: Asterisk PBX 1.6.1.5
> Remote-Party-ID: "435572501134" <sip:435572501134@
> [--ASTERISK--]>;privacy=off;screen=yes
> Content-Length: 0
>
>
> 13:49:30.596307 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length:
> 387
> e.....@.@.[...,N..,L.......KSIP/2.0 200 canceling
> Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060
> From: "435572501134" <sip:435572501...@[--asterisk--]>;tag=as27658014
> To: <sip:+435572949012@
> [--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb
> Call-ID: 4bbee84339a9e2d30850185317983...@[--asterisk--]
> CSeq: 102 CANCEL
> Server: OpenSER (1.3.2-notls (x86_64/linux))
> Content-Length: 0
>
>
> 13:49:30.596801 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length:
> 396
> e.....@.@.Z...,N..,L.......kSIP/2.0 487 Request Terminated
> Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060
> From: "435572501134" <sip:435572501...@[--asterisk--]>;tag=as27658014
> To: <sip:+435572949012@
> [--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb
> Call-ID: 4bbee84339a9e2d30850185317983...@[--asterisk--]
> CSeq: 102 INVITE
> Server: OpenSER (1.3.2-notls (x86_64/linux))
> Content-Length: 0
>
>
> 13:49:30.596842 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length:
> 539
> e..7....@.....,L..,N.....#.oACK sip:+435572949...@[--kamailio--] SIP/2.0
> Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport
> Max-Forwards: 70
> From: "435572501134" <sip:435572501...@[--asterisk--]>;tag=as27658014
> To: <sip:+435572949012@
> [--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb
> Contact: <sip:435572501...@[--asterisk--]>
> Call-ID: 4bbee84339a9e2d30850185317983...@[--asterisk--]
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 1.6.1.5
> Remote-Party-ID: "435572501134" <sip:435572501134@
> [--ASTERISK--]>;privacy=off;screen=yes
> Content-Length: 0
>
>
> Thanks,
>
> Florian
>
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