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.
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@fmctel.com > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@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