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<DIV dir=ltr align=left><SPAN class=359411505-26062006><FONT face=Arial
color=#0000ff size=2>Does anyone on this list has idea?</FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Hoa Thai
Duy<BR><B>Sent:</B> Thursday, June 22, 2006 2:50 PM<BR><B>To:</B> 'Asterisk
Users Mailing List - Non-Commercial Discussion'<BR><B>Cc:</B>
asterisk-dev@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] SIP Channel
hangup problem with re-INVITE enabled- ugrent<BR></FONT><BR></DIV>
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<P><FONT face=Arial size=2>Hi List</FONT> </P>
<P><FONT face=Arial size=2>I have UAs registered with Asterisk and make
outbound calls via ITSP1, everything is fine without re-INVITE. When people call
178, the actual number 112233445566 at ITSP1 network will be called.</FONT></P>
<P><FONT face=Arial size=2>When UA or called telephone (112233445566) hang up,
the call and associated channels are cleared.</FONT> </P>
<P><FONT face=Arial size=2>Sip.conf</FONT> </P>
<P><FONT face=Arial size=2>[general]</FONT> <BR><FONT face=Arial
size=2>canreinvite=no</FONT> <BR><FONT face=Arial
size=2>nat=no
</FONT></P>
<P><FONT face=Arial size=2>[ITSP1]</FONT> <BR><FONT face=Arial
size=2>type=peer</FONT> <BR><FONT face=Arial size=2>host=A.B.C.D</FONT> </P>
<P><FONT face=Arial size=2>Extensions.conf</FONT> </P>
<P><FONT face=Arial size=2>exten => 178,1,Answer()</FONT> <BR><FONT
face=Arial size=2>exten =>
178,n,Dial(SIP/112233445566@ITSP1,60)
</FONT><BR><FONT face=Arial size=2>exten => 178,n,Hangup()</FONT> </P><BR>
<P><FONT face=Arial size=2>However, when I enabled re-INVITE like below, the
call still happen, people can talk with each other. If remote called telephone
(112233445566) hang up, then the call is cleared. But if the Asterisk user (US)
Softphone hang up first, the remote telephone still in talking mode (with no
sound, of course).</FONT></P>
<P><FONT face=Arial size=2>Sip.conf</FONT> <BR><FONT face=Arial
size=2>[ITSP1]</FONT> <BR><FONT face=Arial size=2>type=peer</FONT> <BR><FONT
face=Arial size=2>host=A.B.C.D</FONT> <BR><FONT face=Arial
size=2>Canreinvite=yes</FONT> <BR><FONT face=Arial size=2>Nat=yes</FONT>
</P><BR>
<P><FONT face=Arial size=2>In this case, when Asterisk user hang up and remote
phone still not hang up, I do show like this</FONT> </P>
<P><FONT face=Arial size=2>Show channel verbose</FONT> <BR><FONT face=Arial
size=2>0 active channels</FONT> <BR><FONT face=Arial size=2>0 active
calls</FONT> </P><BR>
<P><FONT face=Arial size=2>Sip show channels</FONT> <BR><FONT face=Arial
size=2>Peer
User/ANR Call ID Seq
(Tx/Rx) Form Hold Last Message
</FONT><BR><FONT face=Arial size=2>A.B.C.D 112233445566
14448d41170 00103/00104 unkn No (d) Rx: BYE
</FONT></P>
<P><FONT face=Arial size=2>CLI> sip show channel
14448d41170ac3a66a41602575476d5f@W.X.Y.Z</FONT> <BR><FONT face=Arial
size=2> * SIP Call</FONT> <BR><FONT face=Arial size=2>
Direction:
Outgoing</FONT> <BR><FONT face=Arial size=2>
Call-ID:
14448d41170ac3a66a41602575476d5f@W.X.Y.Z</FONT> <BR><FONT face=Arial
size=2> Our Codec Capability: 256</FONT> <BR><FONT face=Arial
size=2> Non-Codec Capability: 1</FONT> <BR><FONT face=Arial
size=2> Their Codec Capability: 256</FONT> <BR><FONT
face=Arial size=2> Joint Codec Capability: 256</FONT>
<BR><FONT face=Arial size=2>
Format
unknown</FONT> <BR><FONT face=Arial size=2> Theoretical
Address: A.B.C.D:5060</FONT> <BR><FONT face=Arial
size=2> Received Address:
A.B.C.D:5060</FONT> <BR><FONT face=Arial size=2> NAT
Support:
Always</FONT> <BR><FONT face=Arial size=2> Audio
IP:
W.X.Y.Z(local)</FONT> <BR><FONT face=Arial size=2> Our
Tag:
as5436f254</FONT> <BR><FONT face=Arial size=2> Their
Tag:
caba969d04802f1091a1000000000000--558</FONT> <BR><FONT face=Arial size=2>
SIP User agent: Asterisk</FONT>
<BR><FONT face=Arial size=2>
Username:
112233445566</FONT> <BR><FONT face=Arial size=2>
Peername:
112233445566</FONT> <BR><FONT face=Arial size=2> Original
uri:
sip:112233445566@A.B.C.D:5060</FONT> <BR><FONT face=Arial size=2> Need
Destroy: 2</FONT>
<BR><FONT face=Arial size=2> Last
Message: Rx:
BYE</FONT> <BR><FONT face=Arial size=2> Promiscuous
Redir: No</FONT> <BR><FONT face=Arial
size=2>
Route:
sip:112233445566@A.B.C.D:5060;transport=UDP</FONT> <BR><FONT face=Arial
size=2> DTMF
Mode:
rfc2833</FONT> <BR><FONT face=Arial size=2> SIP
Options:
(none)</FONT> </P>
<P><FONT face=Arial size=2>In this case, when Asterisk user hang up and remote
phone still not hang up, there's still active SIP channel, which should be
cleared when BYE received from any of peers.</FONT></P>
<P><FONT face=Arial size=2>In Asterisk Console, I can see BYE from Asterisk user
(UA Softphone) to Asterisk and OK from Asterisk to UA. But Asterisk DO NOT send
BYE to ITSP1, which is wrong?</FONT></P>
<P><FONT face=Arial size=2>Pls. advice</FONT> </P>
<P><FONT face=Arial size=2>Brgds</FONT> </P>
<P><FONT face=Arial size=2>Hoa</FONT> </P><BR><BR><BR></BODY></HTML>