[Asterisk-Users] 183 Session in Progress

[charles at fmctel.com]

Hi Stewart, I've tryed all the options for connectionmode, also the
connection progress in cisco, but it didn't work. The way that I solved it
was changing the asterisk source to instead of sending Session Progress
when the call is in progress to send Ringing. Because for the users "In
progress" means "Ringing". I've tested it and it works well. This will be
temporary until we take out the ciscos and do Radius and T1 in our
asterisk.

thank you
Charles Rauber Gomes


> 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
>
>
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