Displaying 20 results from an estimated 40000 matches similar to: "asterisk-users Digest, Vol 73, Issue 58"
2010 Aug 26
2
Use of AGISIGHUP
Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn't seem to be doing anything as the script is still exiting on a
hangup and not completing properly. I am using 1.4.35 and have tried
various combinations. Can anyone shed any light on this?
Regards
Lee
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2010 Aug 27
0
Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround() but no audio is heard from the phone
First off, let me first say that this is not a one-way audio problem.
Sometimes I can get 'her' to speak to me, other times I can't for a
long time.
I'm just using a very simple system to dump people into MeetMe().
Nothing fancy.
I do have the following in my modules.conf:
preload => format_mp3.so
preload => codec_ulaw.so
preload => format_pcm.so
My extensions.conf
2010 Aug 27
0
Asterisk 1.6 Displaying BackGround() in call trace but no audio is heard from caller
Thought a different succinct subject line must drum up an answer or two...
Also, this has been tested from two different carriers: We're getting
an average of 2/10 call success rate.
---------- Forwarded message ----------
From: Joe Wood <schmoe at gmail.com>
Date: Thu, Aug 26, 2010 at 6:58 PM
Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw
file using BackGround()
2010 Jun 13
2
bug with Moh on MeetMe ?
Hello,
The MeetMe application refuses MusicOnHold personalized and skip always in
the default!
Have you any idea how to fix this?
-- Executing [028883899 at default:1] Set("SIP/109.10.214.1-00000002",
"CHANNEL(language)=fr") in new stack
-- Executing [028883899 at default:2] Answer("SIP/109.10.214.1-00000002",
"") in new stack
-- Executing
2006 May 25
1
PAP-2 Conferencing Problems
Just come across a problem - we have sent out heaps of PAP-2 ATA's and
just discovered that when joined in a conference they are choppy on the
up leg (so the other users in the conference will hear them with a
choppy sound) but the down leg is perfectly fine (so the end user can
hear the conference participants perfectly).
I have tested the same setup with different brands of ATA's
2014 Oct 21
1
[asterisk-user] Confbridge Kick Action
Hi All,
I am working on Asterisk 12.6.0 with ConfBridge module, when there are
multiple user like admin and normal participant running with conference.
When I try to kicked 2 user (Normal User), it play file "conf-kicked" and
again join conference
My scenario in confbridge like.
1] Admin User (e.g. SIP/8484-00000000)
2] Normal User (e.g. SIP/8484-00000001)
3] Admin User (e.g.
2005 Aug 18
1
Unable to transfer external calls to MeetMe conference
I have a peculiar situation, and am hoping someone on the list can offer
assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server
has no Zap hardware, but is configured to use ztdummy. All incoming calls
are via IAX2.
Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also
able to transfer calls among my SIP devices, voice mail, IVR, etc. All of
my SIP
2004 Jul 01
2
Grandstream HT286 1.0.4.63 & Meetme
Good day!
Have a weird problem with HT-286 and Conference room. I use Asterisk
CVS-HEAD-06/04/04.
Here it is:
When HT-286 get into the conference room first and nobody in that room
everything seems ok (with any codec HT286 allowed), but when HT-286 get
into conference room when somebody already there, have got different HT
behavior:
1. When HT use GSM codec => it connects to conference room,
2005 May 23
1
SendDTMF into a conference room
I have been trying to figure a way to SendDTMF into a MeetMe room using
the Manager API.
I can't redirect everyone into another context and then bring them back
because that would mess up my logic.
I am trying to use local channels and the originate Action to accomplish
this.
Exten: 3441115
Priority: 1
ActionID: actid-00000001
Context: senddtmftones
Action: Originate
Channel:
2009 May 15
1
meetme dies looking for conf-getconfno
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
cat meetme.conf
[rooms]
conf => 600
extensions.conf:
[meetme]
exten => 2663,1,MeetMe(,D)
exten => 2663,n,Hangup()
exten => 2666,1,MeetMe()
exten => 2666,n,Hangup()
What I'm expecting is to dial 2663, get a conference room number ( 600,
I suppose since it's the only room ), and set
2010 Aug 27
2
dynamic MeetMe, min. digits
Hi All,
Is there a way to use the dynamic feature of the meetme application (D)
and to set an option to configure the minimum length of the numbers for
the conference and the associated pin.
In my case, I'd like them to be at least four digits.
Thanks in advance !
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2006 Mar 18
2
Jittery meetme conference using Linksys 942 phones
We have two Linksys 942 phones which sound great when they call each other
directly through Asterisk. But when they both dial in to a meetme conference
room, the sound is very jittery. Other phones like Polycom 501 and Snom 360
sound fine when using meetme.
Both Linksys phones are set to use the default g711u (ulaw) codecs.
Adjusting the jitter buffer and jitter level settings to various values
2004 Dec 09
6
Horrible MeetMe performance
Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
conference, the audio is very delayed, choppy and segmented -- totally
unusable.
At the
2005 Jul 09
1
MeetMe problem - some parameters ignored
Hi All...
I set up a conference bridge using MeetMe. It works nicely, except that it
seems that certain parameters I give it are ignored or else don't work.
Here is the line from my dial plan:
exten => 6500,1,absolutetimeout,0
exten => 6500,2,MeetMe,100|ciMpPs|1234
The MOH and * work, but users are not announced when they join or leave and
the pin is not requested. Maybe I am
2007 Feb 08
2
requesting real world meetme capacity numbers
Hi All,
I'm very interested in real world experience of double digit number of
users sustaining good quality audio in a single meetme conference.
Personally, I have seen 23 users in one conf room, all coming in SIP,
ULAW. Server is 3.2GHz proc, 1Gig RAM, 1-2 % proc utilization under
23 user load, perfect audio.
I'm working on a conf bridge for 150+ users, could use some advice, if
2009 May 07
1
How to get meetme participants in dialplan?
The meetmeadmin() dialplan function lets you specify a user to mute,
un-mute or kick. But how do you get a list of users in your dialplan?
When a user joins a conference, the user number assigned is "the last user
number +1." If you have a long running conference with callers joining and
leaving all the time, this can grow to be a large number.
I want to be able to
2006 Jun 03
4
Meetme versus app_conference
As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.
Does the app_conference application works the same way?
Or if i have SIP/g729 users and i create a conference with other users
also at g729 asterisk will not transcode (when using app_conference)?
2008 Jan 15
3
Meetme recording
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
********************************************************************
This email and any attachments
2007 Mar 22
3
ChanSpy and MeetMe
I have been successful using ChanSpy on a standard Dial call but when
attempting to ChanSpy on an incoming call that has been added to a
MeetMe conference (attempting to coach an agent that is speaking to a
conference of callers) it seems to fail to connect to the channel.
Here's the console dump:
-- Accepting call from '2154182700' to '3399' on channel 0/18, span
4
2007 May 24
3
meetme sounds
I am playing around with dynamic meetme conferences, and wanted to have
one person constantly in the conference, with calls "popping in and out".
Is there an option / any way of playing enter / leave sounds to the
person who created the conference only, and not the people leaving /
joining ?
TIA
Julian.