[SIPForum-discussion] No Bye Message Generated By the PBX.

[Deb, Nihar Ranjan]

Can u try the same scenario using a soft phone. So that when u send BYE to your application through PBX  you can capture the wireshark trace for softphone to analyze what response PBX sends to soft phone after issuing  BYE message.

 U can post the trace for this.

--Nihar

 

 

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Ranjith Kumar
Sent: Tuesday, September 22, 2009 11:13 AM
To: Matt Vlasach
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] No Bye Message Generated By the PBX.

 

Hi Matt,

 

There is no Authorization involved in this case no challanges for the Bye it is just that there is never a Bye that is generated by the PBX.  The session is completly established as we could see the ACK for the 200 OK with SDP.  But when the User in the PBX wishes to terminate the call there is no Bye Generated and th calls hangs in the system.

 

I would check the PBX any way. If you would wish i would send you the traces that were captured.

 

Regards,

Randy

On Tue, Sep 22, 2009 at 9:32 PM, Matt Vlasach <matt.vlasach at pacificswell.com <mailto:matt.vlasach at pacificswell.com> > wrote:

Randy,

Check to see if authentication is failing.  I have run into an issue where the pbx sends a bye to the carrier but the carrier challenges the bye with an Authorization Required response. The pbx I was using didn't know how to authorize a BYE so it would just dumbly terminate it's rtp stream.

Check you sip traces if this is the case. If so, enable bye authorization on your pbx (if possible) otherwise tell the carrier that your system cannot authorize bye messages.

If absolutely no BYE is being generated by the pbx, it sounds like a pbx issue because the pbx should always send a bye to request to terminate the session.

Good luck!

Matt Vlasach (mobile)
Pacific Swell Networks, Inc.
http://pacificswell.com <http://pacificswell.com/> 

email | matt.vlasach at pacificswell.com <mailto:matt.vlasach at pacificswell.com> 
tel | 310.598.3017
fax | 877.531.6463 



On Sep 22, 2009, at 6:59 AM, Ranjith Kumar <feddy006 at gmail.com <mailto:feddy006 at gmail.com> > wrote:

Hi All,

I am encountering a Problem when we are connecting our PBX to our Public Sip Trunk.  The call flow is as follows

Invite  <--
100 Trying -->
180 Ringing -->
200 Ok / SDP <--
ACK -->

RTP <---->


After this The Extention on the PbX side Hangs up but the PBX does not generate a Bye Request Is there something that I have to check in the Previous Packets that is causing this Behaviour from the PBX.

Regards,
Randy

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