I am still looking to solve this problem, does anyone have any ideas? Thanks, Andy
-----Original Message----- From: Andy Goss Sent: Friday, October 07, 2005 5:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] call to a particular 800 number never showsanswered on Zap channel Thanks for the reply. Forgive me for being naïve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk server. So, I have a few questions: Is asterisk or the carrier causing the disconnect? Is IBM (the 800 number I am dialing) not passing the answer supervision or is that a function of the carrier? Is there a way to make asterisk not drop the call or to force the answer on this number? Seems like a hard-PBX would have to be able to handle this type of situation. Thanks, Andy -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey Sent: Friday, October 07, 2005 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call to a particular 800 number never showsanswered on Zap channel This one drove me crazy for a while too. I found out that some companies don't exactly play fair and don't pass answer supervision on a call until you are actually speaking with a live person. The person I spoke to about this wasn't sure if that was even legal, but he said it happens quite a bit. I was lucky in that I use multiple carriers (voipjet and broadvoice), voipjet disconnected the call after 60 seconds, but broadvoice did not, so when I find one of those 800 numbers I route it through broadvoice. Hope that helps, G Andy Goss wrote: > Whenever we call IBM, the call counter on the phone never starts and in > the CLI the zap channel never gets the answered signal from the PRI. > See below. > > -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new > stack > -- Requested transfer capability: 0x00 - SPEECH > -- Called g1/18004267378 > > At this point, I am in IBM's menu system. However the call never > indicates that it is answered either on the phone or in the CLI. After > 60 seconds, the call disconnects. > > -- Hungup 'Zap/1-1' > == Spawn extension (main, 18004267378, 1) exited non-zero on > 'SIP/5933-7bff' > -- Executing Hangup("SIP/5933-7bff", "") in new stack > == Spawn extension (main, h, 1) exited non-zero on 'SIP/5933-7bff' > > Does anyone have any ideas? > > Thanks, > Andy > > -- > H. Andy Goss > Network Engineer > Network Advocates Inc. > Main: 502.412.1050 > DID: 502.992.5933 > Mobile: 502.387.8216 > [EMAIL PROTECTED] > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
