similar to: asterisk-users Digest, Vol 73, Issue 58

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:

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

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()

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

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

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.

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

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,

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:

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

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:

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

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

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

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

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

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)?

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

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

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.