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
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
_______________________________________________
--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