similar to: The efficient way to add MeetMe to pure SIP install ?

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#

So historically I've done one of two things on systems where I've needed to use MeetMe * used a real Digium card, and I've only ever used a TE400 or a TE420 for that purpose, and I know they have the timing chip * used dahdi_dummy, which works well with light load, but I had it running on a very overloaded server and had audio quality issues. I may have had quality issues even with a

Running * version 1.6.1.17. My meetme conferences automagically disconnect users approximately 5-15 seconds after the user is connected. This occurs regardless of whether music on hold is active or not. [Mar 23 11:34:36] -- Executing Macro("SIP/SDN_TMCKEE-000000e9", "confroom,1808") [Mar 23 11:34:36] -- Executing [s at macro-confroom:1]

I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the

Hi, I was switching from zaptel to dahdi and got latest SVN from everything. Compiling works fine. kernel module dahdi_dummy is loaded. /dev/dahdi/pseudo exists Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [8001 at client_int_sgmobile:1] MeetMe("SCCP/6000-00000001", "444|dcIM") in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775

I updated Asterisk to 1.6.2.7 and now the user introduction in the meetme application is no longer working: [May 27 09:26:51] WARNING[2407]: channel.c:4034 ast_request: No channel type registered for 'DAHDI' -- Created MeetMe conference 1023 for conference '800' [May 27 09:26:51] WARNING[2407]: app_meetme.c:3640 find_conf: No DAHDI channel available for conference, user

Hi, I am running Fedora 7 VM. On an earlier configuration with zaptel and Asterisk 1.4.21 , meetme worked alright. I upgraded to Dahdi and Asterisk 1.4.26, and the result is choppy sound via Meeme, while a simple Musiconhold works OK with descent audio quality. So I am sure its a Dahdi_dummy problem. Running dahdi_test gave me very poor results. Opened pseudo dahdi interface, measuring

Usually I used real Digium cards in asterisk systems, so I'm running into this for the first time. dahdi_tool reports that dahdi_dummy is in state UNCONFIGURED. This isn't super surprising, as it seems like the configuration files for DAHDI are really intended only for configuring real physical cards. But that begs the question: Is there any legal DAHDI configuration for dahdi_dummy such

Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules

Howdy, I am playing around with asterisk within an LXC container on Ubuntu 11.04. I have asterisk (1.4.42) running fine, but want access to dahdi_dummy for timing (meetme). I have dahdi installed on the "host", and dahdi_dummy is loaded: root at astnorth:/# ls -ltr /dev/dahdi total 0 crw-rw---- 1 root root 196, 250 2011-06-08 13:59 transcode crw-rw---- 1 root root 196, 253

Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I

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

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. extensions.conf: [meetme] exten => 2663,1,MeetMe(,De) 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 a PIN. Hangup. Then users would dial

I want to have a "personal meetme conference", so when on a call I can transfer the other party to my personal conference with "#7". (I can then make other calls, and dump them into the conference using "#7" as well, then join myself by dialing "7"). Using: exten => 7,1,MeetMe(${CALLERIDNUM}|Mpd) this works as long as I originate the call. However,

Hello, I have a problem with the meetme application, but I'm not sure if it's a bug or just a misuse. I'm trying to get a 3 way call system working as follow : A calls C B calls C C who's speaking with A or B, presses one keypad (only one) and the 2 incoming SIP (A, B) and C are redirected into a conference room. Therefore, I created an entry in the applicationmap

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

Hi when i use MeetMe, i have this errors: app_meetme.c: Unable to open pseudo device Where is the problems ? i have too warning and error into my logs: [Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. [Nov 1 07:26:17] WARNING[18544] config.c: Realtime mapping for 'iaxpeers' found to engine 'mysql', but

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

Hi, My SIP service provider terminates calls in meetme in my Asterisk PBX and am getting delay on those channels. I found following link to measure delay in meetme and to decrease it eventually. http://lists.digium.com/pipermail/asterisk-dev/2005-August/014958.html It says, enable USE_RTC for dahdi_dummy. I have been using virtual server for hosting Asterisk and I had it disabled as per one had

Hi, Vinicius, did you actually solve the choppy audio issue by compiling Gordon's kernel? I have the same problem on the exact same Alix platform (using kernel 2.6.31, though). regards, Darko ps. Sorry everyone if this mail does not get threaded right. I've just joined the mailing list and not sure how to reply to an existing thread I found on the web. -------------- next part