Hi,
we have following setup : PBX <-> Parlay <-ISDN PRI-> Asterisk <-SIP-> GSM
Gateway
Call comes from PBX through Parlay to Asterisk and it routes it over SIP to
GSM gateway. GSM gateway gives back call progress (it takes some time to
ring or get through), but this info won't get back to P
Nitin Joshi wrote:
> Hi All,
> I am using Asterisk 1.0.7 with an X101P analog card which is connected
to an
> Alcatel pbx. My problem is that when I place outbound calls on the zap
> channel, Asterisk returns a connect event as soon as the phone start
> ringing. This means that Asterisk is not bein
Hi All,
I am using Asterisk 1.0.7 with an X101P analog card
which is connected to an Alcatel pbx. My problem is that when I place outbound
calls on the zap channel, Asterisk returns a connect event as soon as the phone
starts ringing. This means that Asterisk is not being able to do Call Prog
Title: Call Progress Detection
We have done some answering detecection coding that can differentiate answering machines and live answers. We're having problems with operator intercepts. Asterisk is showing them as No Answers, Does anyone have any suggestions on how to properly differentiate b
Hi to all,
I'm using a TDM22B. When i establish an external call to the PSTN through an
FXO port, I'm not able to know the status of the call (no answer, busy, ...).
If I enable call progress (callprogress=yes) in Zapata.conf, I am able to
detect the no answer state but if the callee on the PST
I have the same problem.
callprogress is very experimental and buggy now.
and i've lost the .call files feature of asterisk.
what do you think about submitting a bug on bugs.digium.com?
regards,
shabanip
> Hello,
>
> I've been experimenting with the call progress analysis features of *,
> with mi
Hello,
I've been experimenting with the call progress analysis features of *, with mixed
success on Zap as well as IAX channels. I've read all the posts about it, including
(but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the
pages it references.
My question is, w
On Sun, 2004-07-18 at 20:38, Stephen David wrote:
> Hello,
>
> I haven't seen any recent posts on call progress detection, so here's
> a question:
>
> How would one accomplish an automated outbound dialing application
> using *, whereby a requirement is to wait for the greeting to complete
> (liv
Hello,
I haven't seen any recent posts on call progress detection, so here's a question:
How would one accomplish an automated outbound dialing application using *, whereby a
requirement is to wait for the greeting to complete (live person, answering machine,
voicemail) before delivering the me
downloaded and compiled today's CVS (04/08/2004)
tried using callprogress on Via mini-itx (running RedHat Linux
9)
if callprogress is set to yes on x100p, an i call the line
connected to x100p, asterisk would execute the first app and
will wait forever.
anyone had success using callprogress?
th
Hi Mark,
I have implemented a procedure for automatically calls
from the client-side (IaxClient - E100P)
What I want to do is to detect the call status from
the client-side.
Meaning, if the line is busy/unavailable/fax log the
status and proceed to next call.
Is this possible with Manager API ?
Just curious as to whether or not there's anyone else out there who is
using iconnecthere behind a completely-NATted asterisk system.
My system, using code from approx a week ago, works just fine with
iconnect, except that I get no "call progress" information (if that's
the correct terminology)
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