[Asterisk-Users] 183 Session in Progress

[Stewart Nelson]

Hi Charles,

Blocking the 183 is undesirable, because messages
from the PSTN indicating that e.g. a number has been changed,
will be lost.  Instead, do what's necessary to get audio
back to the caller.  On the ATA, set bit 19 of ConnectMode
(see table 5-8 of manual).  On the 5300, see
http://www.ciscopress.com/content/images/1587050757/tips/troubleshooting_tips.doc
and look for "No Ringback on an IP Phone When Calling the PSTN" .

--Stewart

> Date: Fri, 18 Jun 2004 16:30:41 -0400 (EDT)
> Subject: Re: [Asterisk-Users] 183 Session in Progress
> From: charles at fmctel.com
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
> 
> I've the same problem with the Cisco ATA's  and Cisco 5300. The cisco
> sends the: "SIP/SDP Status: 183 Session Progress  , with session
> description", asterisk forwards is to the phone: "SIP/SDP Status: 183
> Session Progress, with session description" after that the SIP Phone stops
> ringing.
> People complains because that Dead Signal while they wait the call to be
> completed, but I don't know what to do, if it's possible to stop asterisk
> forwarding this, or stops cisco sending this.
> 
> 
> Thank you