For 2 different hosts. SIP/voxbone.com and SIP/4420
From: RSCL Mumbai
Sent: Thu 5/19/2011 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropping incompatible voice frame
Processor: Intel Dual Core Xeon 3.0GHz
- Host: CentOS 5.6 (64
But why does *our *native format keep changing :)
Going by layman terms, if native format is alaw and someone speaks to me in
uLaw, I will say *format changed*.
But if native format is alaw and someone is talking with me in alaw, I
should be happy.
On Thu, May 19, 2011 at 10:28 PM, Terry
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Thursday, May 19, 2011 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dropping
Possibly or possibly not. Most (IMO) calls are placed initially with the
choice 2-3 or more codecs. Normally one codec is negotiated and life goes
on, but IAX is a little different from a SIP/DAHDI call. The most certain
remedy I can think of for this it to just unallow the alaw codec on IAX
Richard Kenner wrote:
I have a SIP phone calling an AGI application. It starts out this way:
-- Executing [...@macro-call-agi:2] AGI(SIP/151-b414f0c8,
computer-temp.sh,darwin,) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh
Then I get a dozen or
What version of Asterisk are you running? This sounds similar to an
issue with AGI's I saw a while ago, but I can't quite remember
exactly what the issue (or issue number) was.
1.6.2.0-rc2
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-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven J.
Douglas
Sent: Wednesday, January 28, 2009 9:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dropping incompatible voice frame
Don't use g729
IMO it is a bridging problem. The evidence of this is:
SIP - Analog - no outgoing audio connection
Analog - SIP (actually Analog - * - SIP - everything ok.
Try putting an Answer() in front of Dial() in your dialplan
(extensions.conf) and see if this goes away.
-Original Message-
From:
Don't use g729 in the iax.conf for the IAXY device. It doesn't support it.
Regards,
Steve
Adam Robins wrote:
I am using a Polycom SIP phone (ext 2042) to call an analog phone
connected via an IAXY (ext 2120). The analog phone rings, and when I
answer, I can hear the person speaking on the
-Commercial Discussion
Subject: Re: [Asterisk-Users]
Dropping incompatible voice frame
This is known issue, we fixed it by putting an answer() in the
dial plan before it gets forwarded,thefix
transcode_via_sln=no (detailed in the bug tracker)didn't work for me.
YMMV.
http
This is known issue, we fixed it by putting an answer() in the dial plan before it gets forwarded,thefix transcode_via_sln=no (detailed in the bug tracker)didn't work for me. YMMV.
http://bugs.digium.com/view.php?id=4101
On 6/28/06, Kevin Savoy [EMAIL PROTECTED] wrote:
Sorry if this has been
Joe
Sorry I did not get back to you on this. Set your redirect event to never.
Andrew
On 1/19/06, Joseph Rothstein [EMAIL PROTECTED] wrote:
I am now getting these messages on a second box running a different version
of Asterisk. If anyone has any idea what is causing these, or how to avoid
On Thursday 19 January 2006 13:48, Joseph Rothstein wrote:
I am now getting these messages on a second box running a different version
of Asterisk. If anyone has any idea what is causing these, or how to avoid
them I would be very grateful.
157 Jan 19 11:59:50 NOTICE[6070]: Dropping
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