[Asterisk-Users] 183 Session in Progress

[Juan J. Sierralta P.]

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.