[Asterisk-Users] 183 Session in Progress

charles at fmctel.com charles at fmctel.com
Fri Jun 18 13:30:41 MST 2004


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


> On Wed, 2004-05-05 at 04:11, Radius wrote:
>> Hi all,
>>
>> From Cisco 7960 I made outgoing calls through Cisco AS5300 to PSTN
>> by   exten => _XXXXXXXX,1,Dial(SIP/${EXTEN:}@150.11.131.2,60,r).
>> 150.11.131.2 is the Cisco AS5300 PSTN gateway.
>>
>> 7960 rings for the first 2 seconds, then display "Session Progress
>> (183)" with no more rings while the phone at the other side of PSTN is
>> ringing. However, calls can be answered and there is no problem for
>> phone converation. The same problem happens on CIsco ATA186. However,
>> it does NOT happen on Grandstream phones. It looks like the call setup
>> problem is only for Cisco products.
>
> 	My guess after my own investigations is that Cisco boxes do honour
> Session Progress, usually when a gateway respond with Session Progress
> it sends also a SDP header signalling the media in that media comes the
> call progress tone as RTP audio I think Granstreams products do not work
> so good with session progress but they do ring because they received a
> 180 Ring and they stay ringing until they receive the connect message.
> 	Probably your AS5300 isn't sending call progress via a RTP stream you
> should use ethereal to see whats happening.
> 	IMHO it's better to honour session progress because it is usual that
> the PSTN puts another tones on the call progress for example voicemail
> prompts in order to start billing after the beep when the call is
> actually answered.
> 	Try calling your cell voicemail with your Granstream and Cisco to see
> whats happening.
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>




More information about the asterisk-users mailing list