1 Dec
2009
1 Dec
'09
4:47 p.m.
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(a)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.net<sip%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
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