Andy Goss
2005-Oct-07 12:54 UTC
[Asterisk-Users] call to a particular 800 number never shows answered on Zap channel
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 agoss@ntad.com
Garth Summey
2005-Oct-07 14:17 UTC
[Asterisk-Users] call to a particular 800 number never shows answered 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 > agoss@ntad.com > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 > > >
Stewart Nelson
2005-Oct-11 06:41 UTC
[Asterisk-Users] call to a particular 800 number never shows answered on Zap channel
> 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.First, there is nothing "unfair" or "illegal" going on. Large toll-free users have enough clout that they can negotiate contracts, where they are not billed during the service selection phase of a call. For example, when you call American Airlines, billing doesn't start until an agent answers, or the caller selects "automated flight information" or a similar IVR service. Answer supervision is used to tell the carrier when to start billing. This system is quite common and used by hundreds of companies. With Asterisk, three things might go wrong: You may hear ringing instead of the initial IVR greeting. If your carrier is sending 180 Ringing instead of 183 Progress (SIP) or Alerting without inband audio (PRI), then they must fix the problem; nothing can be done at your end. You may hear the IVR answer, but can't control it, because your outbound DTMF or voice is blocked. Your carrier might be doing the blocking, in which case they obviously must fix it. However, there are also some SIP phones and ATAs that don't send outgoing audio during Progress. If you have such, adjust the configuration if possible. If not, you will need to disable reinvites. You may have two-way communication with the IVR, but the call gets disconnected before answer supervision is received. Find out if it's your carrier or Asterisk that is timing out. If the latter, just put a longer timeout in your Dial statement; 180 seconds should be enough. --Stewart
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