asterisk users - May 2007 - asterisk not sending ACK after reinvite

Hi,

I am faced with this dilema of asterisk not sending an ACK after it receives
200 OK from OpenSER (which is a  response to a reinvite request sent by
asterisk. Here is my setup

Carrier<->OpenSER<->Asterisk1<->Asterisk2

A user is connected with Asterisk1 (through the carrier and OpenSER). On
certain dtmf events the call is forwarded to Asterisk2 using the Dial
command. Canreinvite is set to "yes" in Asterisk1's sip.conf,
therefore it
sends reinvites to both Asterisk2 and OpenSER to release RTP.
OpenSER forwards the reinvite to the carrier and relays the 200 OK received
back to Asterisk1 but Asterisk1 never responds back with an ACK. Finally the
transaction on OpenSER times out and a bye message is sent to Asterisk1,
causing both legs to be hungup. If I reset canreinite to no the scenario
works.

The Invite message sent to OpenSER and 200 OK received are shown below:

INVITE sent
-----------
Session Initiation Protocol
    Request-Line: INVITE sip:1234@192.168.0.1;transport=udp SIP/2.0
        Method: INVITE
        Resent Packet: False
    Message Header
        Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK67156992;rport
        Route: <sip:192.168.0.2;ftag=as04d1d0dc;lr=on>
        From: "16477239819"
<sip:16477239819@192.168.0.3>;tag=as04d1d0dc
        To: <sip:12133411419@192.168.0.2>;tag=d12f2182-140a6d
        Contact: <sip:16477239819@192.168.0.3>
        Call-ID: 7d1f99f5735cdec8743ed3d244a05c99@192.168.0.3
        CSeq: 104 INVITE
        User-Agent: Asterisk
        Max-Forwards: 70
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Content-Type: application/sdp
        Content-Length: 245
    "SDP not shown"

200 OK received
---------------

Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
    Message Header
        Call-ID: 7d1f99f5735cdec8743ed3d244a05c99@192.186.0.3
        Contact: <sip:666251612133411419@192.168.0.1;transport=udp>
        Content-Length: 232
        Content-Type: application/sdp
        CSeq: 103 INVITE
        From:
"16477239819"<sip:16477239819@192.186.0.3>;tag=as04d1d0dc
        Record-Route: <sip:192.168.0.2;ftag=as04d1d0dc;lr=on>
        To: <sip:12133411419@192.168.0.2>;tag=d12f2182-140a6d
        User-Agent: Quintum/1.0.0
        Via: SIP/2.0/UDP 192.186.0.3:5060;branch=z9hG4bK0f664853;rport=5060
    "SDP not shown"


Now the interesting thing is that if I take out OpenSER and forward directly
to the carrier then it works fine. The 200 OK received from the carrier is
shown below

Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
        Status-Code: 200
        Resent Packet: False
    Message Header
        Call-ID: 11d8858b42cf83725641484d0f63289d@192.168.0.3
        Contact: <sip:666251614168404385@192.168.0.1>
        Content-Length: 232
        Content-Type: application/sdp
        CSeq: 103 INVITE
        From:
"16477239819"<sip:16477239819@192.168.0.3>;tag=as41da20f1
        To: <sip:666251614168404385@192.168.0.1>;tag=d12f2182-140d2e
        User-Agent: Quintum/1.0.0
        Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0e1703c7;rport

The differences I notice are
1. OpenSER modifies "rport" at the end of Via to
"rport=5060".
2. Openser appending "transport=udp" in Contact.

I am using Asterisk 1.2-18, canreinvite is set to yes and nat is set to no.

I will really appreciate if someone can shed some light on this issue and
help me fix it.

Regards,
Danish
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Danish Samad wrote:
> Hi,
>
> I am faced with this dilema of asterisk not sending an ACK after it
> receives
> 200 OK from OpenSER (which is a  response to a reinvite request sent by
> asterisk. Here is my setup
Firstly don't cross post.

Olle posted a fix for this yesterday (or the day before):

Author: oej
Date: Fri May 18 13:10:46 2007
New Revision: 65122

URL: http://svn.digium.com/view/asterisk?view=rev&rev=65122
Log:
Not getting an ACK to a 200 OK in the initial invite is critical to
the call.

Modified:
    branches/1.2/channels/chan_sip.c

- --
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com (Great new VoIP end to end solution)
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