Hi,<br>
<br>
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<br>
<br>
Carrier<->OpenSER<->Asterisk1<->Asterisk2<br>
<br>
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.<br>
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.<br>
<br>
The Invite message sent to OpenSER and 200 OK received are shown below:<br>
<br>
INVITE sent<br>
-----------<br>
Session Initiation Protocol<br>
Request-Line: INVITE <a href="mailto:sip:1234@192.168.0.1">sip:1234@192.168.0.1</a>;transport=udp SIP/2.0<br>
Method: INVITE<br>
Resent Packet: False<br>
Message Header<br>
Via: SIP/2.0/UDP <a href="http://192.168.0.3:5060">192.168.0.3:5060</a>;branch=z9hG4bK67156992;rport<br>
Route: <sip:<a href="http://192.168.0.2">192.168.0.2</a>;ftag=as04d1d0dc;lr=on><br>
From: "16477239819" <<a href="mailto:sip:16477239819@192.168.0.3">sip:16477239819@192.168.0.3</a>>;tag=as04d1d0dc<br>
To:
<<a href="mailto:sip:12133411419@192.168.0.2">sip:12133411419@192.168.0.2</a>>;tag=d12f2182-140a6d
<br>
Contact: <<a href="mailto:sip:16477239819@192.168.0.3">sip:16477239819@192.168.0.3</a>><br>
Call-ID: <a href="mailto:7d1f99f5735cdec8743ed3d244a05c99@192.168.0.3">7d1f99f5735cdec8743ed3d244a05c99@192.168.0.3</a><br>
CSeq: 104 INVITE<br>
User-Agent: Asterisk<br>
Max-Forwards: 70<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Type: application/sdp<br>
Content-Length: 245<br>
"SDP not shown"<br>
<br>
200 OK received<br>
---------------<br>
<br>
Session Initiation Protocol<br>
Status-Line: SIP/2.0 200 OK<br>
Message Header<br>
Call-ID: <a href="mailto:7d1f99f5735cdec8743ed3d244a05c99@192.186.0.3">7d1f99f5735cdec8743ed3d244a05c99@192.186.0.3</a><br>
Contact: <<a href="mailto:sip:666251612133411419@192.168.0.1">sip:666251612133411419@192.168.0.1</a>;transport=udp><br>
Content-Length: 232<br>
Content-Type: application/sdp<br>
CSeq: 103 INVITE<br>
From: "16477239819"<<a href="mailto:sip:16477239819@192.186.0.3">sip:16477239819@192.186.0.3</a>>;tag=as04d1d0dc<br>
Record-Route: <sip:<a href="http://192.168.0.2">192.168.0.2</a>;ftag=as04d1d0dc;lr=on><br>
To: <<a href="mailto:sip:12133411419@192.168.0.2">sip:12133411419@192.168.0.2</a>>;tag=d12f2182-140a6d<br>
User-Agent: Quintum/1.0.0<br>
Via: SIP/2.0/UDP <a href="http://192.186.0.3:5060">192.186.0.3:5060</a>;branch=z9hG4bK0f664853;rport=5060<br>
"SDP not shown"<br>
<br>
<br>
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<br>
<br>
Session Initiation Protocol<br>
Status-Line: SIP/2.0 200 OK<br>
Status-Code: 200<br>
Resent Packet: False<br>
Message Header<br>
Call-ID: <a href="mailto:11d8858b42cf83725641484d0f63289d@192.168.0.3">11d8858b42cf83725641484d0f63289d@192.168.0.3</a><br>
Contact: <<a href="mailto:sip:666251614168404385@192.168.0.1">sip:666251614168404385@192.168.0.1</a>><br>
Content-Length: 232<br>
Content-Type: application/sdp<br>
CSeq: 103 INVITE<br>
From: "16477239819"<<a href="mailto:sip:16477239819@192.168.0.3">sip:16477239819@192.168.0.3</a>>;tag=as41da20f1<br>
To: <<a href="mailto:sip:666251614168404385@192.168.0.1">sip:666251614168404385@192.168.0.1</a>>;tag=d12f2182-140d2e<br>
User-Agent: Quintum/1.0.0<br>
Via: SIP/2.0/UDP <a href="http://192.168.0.3:5060">192.168.0.3:5060</a>;branch=z9hG4bK0e1703c7;rport<br>
<br>
The differences I notice are <br>
1. OpenSER modifies "rport" at the end of Via to "rport=5060".<br>
2. Openser appending "transport=udp" in Contact.<br>
<br>
I am using Asterisk 1.2-18, canreinvite is set to yes and nat is set to no.<br>
<br>
I will really appreciate if someone can shed some light on this issue and help me fix it.<br>
<br>
Regards,<br>
Danish