similar to: MOH and conference calls

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

Hi all, I've got an AGI script that launches the conference bridge with a line like: "$main::agi->exec(ConfBridge,$conf,default_bridge,default_user,$menu_profile)" The $conf variable contains the room number. I'm trying to configure it so that when only one person is in the conference, they hear moh. My /etc/asterisk/confbridge.conf looks like:

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having

Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 Would anyone have an idea on how I would be able to take the mic in on the computer and put it as the "talking party" for a conference room. I would then be able to set up a "listen only" profile for others to get in on. Reason for doing this is for 'shut-in's' for my Church.

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

>The combination of applications CBMysql and MeetMe2 seem to >address our goals. I have MeetMe2 working. CBMysql is >another story, the code looks simple enough and has been >modified to leverage MeetMe2, but * restarts everytime it >tries to launch CBMysql. I cannot find any examples of how >to launch it from the dial plan, nor have I been able to >get any meaningful

A couple comments. I'm not a programmer, my C is passable, but my web development would have to grow by leaps and bounds to be considered poor. I pulled the Meetme2 from here: http://www.areski.net/asterisk-meetme/about.php?s=0 The app needs a minor tweak to compile against 1.0.5. I stumbled down a false path or two, so the diff shows some lines being deleted and re-added that are

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

Struggling with an IP650 and 7 IP335s this morning. I have the following hints defined (courtesy of FreePBX 2.9): extensions_additional.conf:exten => 300,hint,SIP/300 extensions_additional.conf:exten => 301,hint,SIP/301 extensions_additional.conf:exten => 302,hint,SIP/302 extensions_additional.conf:exten => 303,hint,SIP/303 extensions_additional.conf:exten => 304,hint,SIP/304

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.

Cool! I'm still away from the office, but I was starting to work towards syching meetme2 up to the version of meetme in * 1.0.7. It is over a 2000 line diff, ignoring the database integration code, so it was looking like a not too trivial task. One question though, how difficult will it be to extend your latest version with the scheduling features I built on top of you previous version? I

Hi, I have a data file with columns separated by ";" I read this file without any problem using read.csv2( ) but I had problems trying to read it with read.table( ... sep=";"). So it is not a problem for me, but I wonder if there is a bug here. drt <- read.csv2("t.txt", header=TRUE) # ok dcs <- read.table("t.txt", header=TRUE,

That's what I thought it used to be but it isn't working now Here is what I have in my op_buttons.conf [801] ; Meetme must be defined by its room number Position=15 Label="MeetMe 801" Extension=801 Context=rooms Icon=5 [802] Position=16 Label="Meetme 802" Extension=802 Context=rooms Icon=5 This is in the meetme.conf [rooms] #include

Hello! Do somebody knows how to compile meetme2 with 1.0.6. I readed wiki, applied patches, but no luck ;-( Me be someone can give me working meetme2.c ? :-)

r> <h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010= ) </h2>With SIP 3.2.X firmware (available on the Polycom download site)=20 and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20 showing statuses of Ringing, Inuse and Online and one touch directed=20 call pickup. <br>On the asterisk side all that needs to be done is to add a hint

Actually I can start and receive SIP calls (PC client, iphone client) but?I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this ?error when making a call: *CLI> ? == Using SIP RTP CoS mark 5 ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", "DAHDI/1/4384019357,10") in new stack [Jun 23 16:18:09]

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

Hi, I'm currently working with Asterisk 1.4.17 under ubuntu server 8.04.2. MOH stopped working suddenly a few days ago with no apparent reason. I already checked the wiki and tried different things. I already verified the following items from the wiki: 1. Make sure your asterisk user has read access to the files/folder 2. Set your moh conf up as mentioned above 3. Go into asterisk -r and do

do you have selinux enabled? It should not be. p p.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything. From: "Lachek Butalek" <lachek@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date:

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the