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/066480588134@from-internal' (thanks to SIP/tpsiptestproxyu01-00d0a0b8) -- Executing [066480588134@from-internal:1] NoOp("Local/066480588134@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 [066480588134@from-internal:2] NoOp("Local/066480588134@from-internal-d69e;2", "") in new stack -- Executing [066480588134@from-internal:3] Dial("Local/066480588134@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/066480588134@from-internal-d69e;2 -- Local/066480588134@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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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..r..@.@.[0..,N..,L.....^.aSIP/2.0 100 Trying Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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..a..@.@.XA..,N..,L.....M+LINVITE sip:066480588134@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572949012@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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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
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.Meister@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/066480588134@from-internal' (thanks to SIP/tpsiptestproxyu01-00d0a0b8) -- Executing [066480588134@from-internal:1] NoOp("Local/066480588134@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 [066480588134@from-internal:2] NoOp("Local/066480588134@from-internal-d69e;2", "") in new stack -- Executing [066480588134@from-internal:3] Dial("Local/066480588134@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/066480588134@from-internal-d69e;2 -- Local/066480588134@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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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..r..@.@.[0..,N..,L.....^.aSIP/2.0 100 Trying Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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..a..@.@.XA..,N..,L.....M+LINVITE sip:066480588134@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572949012@tpseru01.tele.netsip%3A435572949012@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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@ [--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@ [--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@ [--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Hi Florian!
1. As Asterisk detects the "spiral" and handles it internally, there is nothing you can do in Kamailio.
2. Just a warning: In general, Remote-Party-ID is used for CLI indication, not for billing purposes. In your setup it might be possibly to use RPID for billing to, but it would be better to use the proper headers, like the obsolete (as you are using RPID I suspect you like obsolete headers ;-) Diversion header, or the History-Info header. (another warning: make sure that your system ignores/removes these headers when sent by your customers - otherwise they can fool your billing system and signal arbitrary CLIs)
3. You can for example easily bypass the problem by making the DB lookup also in Asterisk. Use e.g. func_odbc to query your call-forwarding table from within Asterisk, set the CDR(whatever) variables to fulfill you billing requirements, and then send the call to the PSTN again.
regards Klaus
Florian Meister 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/066480588134@from-internal' (thanks to SIP/tpsiptestproxyu01-00d0a0b8) -- Executing [066480588134@from-internal:1] NoOp("Local/066480588134@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 [066480588134@from-internal:2] NoOp("Local/066480588134@from-internal-d69e;2", "") in new stack -- Executing [066480588134@from-internal:3] Dial("Local/066480588134@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/066480588134@from-internal-d69e;2 -- Local/066480588134@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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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..r..@.@.[0..,N..,L.....^.aSIP/2.0 100 Trying Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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..a..@.@.XA..,N..,L.....M+LINVITE sip:066480588134@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572949012@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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--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:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--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:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" sip:435572501134@[--ASTERISK--];tag=as27658014 To: sip:+435572949012@[--KAMAILIO--];tag=45db18648893e7acabf725621374d382-4ddb Contact: sip:435572501134@[--ASTERISK--] Call-ID: 4bbee84339a9e2d30850185317983625@[--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
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
- As Asterisk detects the "spiral" and handles it internally, there is
nothing you can do in Kamailio.
That's bad. So I have to do the diversion check at two places ...
- Just a warning: In general, Remote-Party-ID is used for CLI
indication, not for billing purposes. In your setup it might be possibly to use RPID for billing to, but it would be better to use the proper headers, like the obsolete (as you are using RPID I suspect you like obsolete headers ;-) Diversion header, or the History-Info header. (another warning: make sure that your system ignores/removes these headers when sent by your customers - otherwise they can fool your billing system and signal arbitrary CLIs)
Thanks for the info. What are the headers which should be used for CLI, the not obsolete ones ? ;-)
- You can for example easily bypass the problem by making the DB lookup
also in Asterisk. Use e.g. func_odbc to query your call-forwarding table from within Asterisk, set the CDR(whatever) variables to fulfill you billing requirements, and then send the call to the PSTN again.
Ok. That's what I have to do.
Thanks for your help,
Ciao, Florian
On Tuesday 01 December 2009 15:07:48 Florian Meister wrote:
Thanks for the info. What are the headers which should be used for CLI, the not obsolete ones ? ;-)
RFC 3323, 3324 and 3325
PAI: P-Asserted-Identity PPI: P-Prefered-Identity
Hi all,
I got table of subscribers attached to a domain "sip.domain.com". I wish to support receiving "username@sip.antisip.com" as well as "username@antisip.com".
I understand I could disable the check for domain in modules, but that won't fit my needs as I have several separate domains. Is there any possibility to define alias for one specific domain?
sip.domain.com alias of domain.com sip.domain2.com alias of domain2.com etc...
Tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/
El Martes, 1 de Diciembre de 2009, Aymeric Moizard escribió:
Hi all,
I got table of subscribers attached to a domain "sip.domain.com". I wish to support receiving "username@sip.antisip.com" as well as "username@antisip.com".
It sounds to DNS SRV usage with clients not supporting it so they need the DNS A type.
If this is the case, I thin a better approach is setting the outbound proxy for those clients (sip.antisip.com) and setting the user's domain to be "antisip.com".
Don't try to use a multidomain environment with clients using domain aliases as it makes impossible the authentication in the case the user uses "user@domain" in the credentials username.
I understand I could disable the check for domain in modules, but that won't fit my needs as I have several separate domains. Is there any possibility to define alias for one specific domain?
sip.domain.com alias of domain.com sip.domain2.com alias of domain2.com etc...
As I've said above, I think this is the wrong way.
Regards.
Hi Aymeric!
AFAIK there is no built-in feature in Kamailio 1.5 to solve this issue.
But maybe you can solve it manually - for each task separately: - for call routing, just replace sip.domain with domain.com if detected, e.g.: if ($rd=="sip.domain") { $rd="domain"; }
Also add sip.domain to domain table (if you use the domain module).
More complicated is authentication: I think it can be solved by checking for "sip.domain" and putting "domain" into PV and use pv_[www|proxy]_authorize() functions.
I think the hardest part is registration. I do not know a solution to save() user@sip.domain as user@domain.
Probably the easiest solution would be - as indicated by Inaki - ignoring clients which are wrong configured or do not allows to configure outboundproxy and SIP domain separately.
Maybe it can be solved also by useng sip-router and ser's userloc/registrar/auth module. AFAIK in ser's DB schema, the SIP AoR was decoupled from the credentials (subscriber table) - but you would need to migrate your DB schema.
regards klaus
Aymeric Moizard wrote:
Hi all,
I got table of subscribers attached to a domain "sip.domain.com". I wish to support receiving "username@sip.antisip.com" as well as "username@antisip.com".
I understand I could disable the check for domain in modules, but that won't fit my needs as I have several separate domains. Is there any possibility to define alias for one specific domain?
sip.domain.com alias of domain.com sip.domain2.com alias of domain2.com etc...
Tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Florian Meister wrote:
Thanks for the info. What are the headers which should be used for CLI, the not obsolete ones ? ;-)
See Raul's email for the standardized headers. These are often used for provider interconnect and also supported by recent PSTN gateways.
IIRC there is no explicit support for PAI, PPI in Asterisk, so you would have to use the SIP_HEADER function and manually parse the headers.
So, as long as you use it only internally, you can use the obsolete RPID header too, actually you could any header.
For example, internally (between proxies and Asterisk-Gateways) I use a lot of proprietary headers for signaling for internal data, for example:
X-AU: +436991234567 (user provided CLI)
X-AN: +431234567 (network provided CLI)
X-C: +437201234567 (number which did the forwarding)
X-BID: 00046346 (billing ID (internal customer identity))
X-Hangupafter: 1000 (terminate call after x seconds)
regards Klaus
Hi Klaus and Raul,
Thanks for your answers ... I think I'm more clear now.
ciao, flo
See Raul's email for the standardized headers. These are often used for provider interconnect and also supported by recent PSTN gateways.
IIRC there is no explicit support for PAI, PPI in Asterisk, so you would have to use the SIP_HEADER function and manually parse the headers.
So, as long as you use it only internally, you can use the obsolete RPID header too, actually you could any header.
-
Aymeric Moizard -
Florian Meister -
Iñaki Baz Castillo -
Klaus Darilion -
Raúl Alexis Betancor Santana -
Uriel Rozenbaum