Hello Everyone,
I have a new problem where when placing the call, asterisk will
automatically go into music on hold until the call is connected (ie,
no ringing). It was kind of confusing because sometimes `SESSION
PROGRESS` takes longer than others, during this time we are in MOH.
The call does
Nick Cameo wrote:
There is two way audio, it's just during ringing that this happens.
If you can put the SIP signaling and Asterisk console output up
somewhere then we can have a better idea of what Asterisk is being told,
and what it is doing.
--
Joshua Colp
Digium, Inc. | Senior Software
Does your Dial() command include the m option?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
There is two way audio, it's just during ringing that this happens.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hello Jg,
Thank your for your response. No m option on dial. I think it's a RTP
relay issue however, do not know how to diagnose the SDP payload. Any
help would be appreciated.
N.
--
_
-- Bandwidth and Colocation Provided by
Nick Cameo wrote:
I ran a test call with trace can be found here:
http://pastebin.com/f8MuxaFV
I also wanted to mention that yes we have * setup with disallow=all
and allow=g729 for testing,
maybe permanently if we can successfully setup G729 pass through. That
being said, the same problem is
Maybe the ringtone from downstream is not
reaching asterisk, and thus a2billing is appending the `m` to the dial
command?
With digital systems (starting with ISDN, or so), ringing is signaled, or indicated. The
ringtone is produced locally, either by the PBX or by the SIP phone itself. Since
Yeah of course. Still digging into it :). Will post the solution if I
find it. a2billing forum takes for ever to answer...
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
I have no idea where the `m` is coming from. I even looked into the
A2Billing script. Still digging
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Yes of course, I just did not want to overwhelm you guys with SIP
trace. Before that though, I realized something:
[Sep 10 12:03:30] WARNING[8178]: res_musiconhold.c:802 set_moh_exec:
SetMusicOnHold application is deprecated and will be removed. Use
Set(CHANNEL(musicclass)=...) instead
-- AGI
Hope this helps someone save a day of running around.
So my issue was with a2billing. The warning `No remote address on RTP
instance '0xb6d16a28' so dropping frame`
was not related to the music on hold coming on during ringing.
The Problem:
We have a script that loads rates into
Did you the a2billing settings for a music on hold setting
I remember seeing some setting
-Original Message-
From: Nick Cameo sym...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 10 Sep 2013 12:46:54
To: Asterisk Users Mailing List - Non-Commercial
Nick Cameo wrote:
Yeah!!! The Dial command setting:
http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704
I know this is not an a2billing mailing list, and I am sorry however,
I do think that the No remote address on RTP instance may have
something to do with it. Maybe the ringtone from
Oh scandalous Instead of playing the MOH, I would like to play the
ringtone that is on the machine. Ummm, where is it? :)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
I ran a test call with trace can be found here:
http://pastebin.com/f8MuxaFV
I also wanted to mention that yes we have * setup with disallow=all
and allow=g729 for testing,
maybe permanently if we can successfully setup G729 pass through. That
being said, the same problem is still there using
i have used a2billing some time ago maybe there is somthing new .
you can try shoot up loglevel to 4 and see the verbose of agi that may give
you some hint.
On Tue, Sep 10, 2013 at 7:34 PM, jg webaccou...@jgoettgens.de wrote:
Maybe the ringtone from downstream is not
reaching asterisk,
Yeah!!! The Dial command setting:
http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704
I know this is not an a2billing mailing list, and I am sorry however,
I do think that the No remote address on RTP instance may have
something to do with it. Maybe the ringtone from downstream is not
17 matches
Mail list logo