similar to: Add Apps to Asterisk?

How do I set up an existing call to dial out to a new terminal which is included in a conference with the two existing legs of the call? When the dialplan executes the Dial(<terminal>) command, control does not return to the dialplan until the terminal disconnects, after which it's obviously too late to conference it. Is there a conference command or option that lets the dialplan dial

I'm trying to use Asterisk (v1.2.11) make a callback that dials both legs of the call into a Meetme() room together, but I keep getting "conf-invalid" messages. I created a callfile (/var/spool/asterisk/outgoing/out.call) that specifies a Local channel (extension) which contains a Dial() command to the "dialer", and an extension which contains a Dial() command to the

hi guys: recently I want to buinding a meeting confence on asterisk and use the meetme application. I have a ztdummy kernel afteri the lsmod commond: ztdummy 5768 0 zaptel 182660 28 zttranscode,ztdummy crc_ccitt 3008 1 zaptel I also configure the meetme.conf conf => 1000; my extensions.conf [default] exten =>

Hi guys, Having a bit of a problem trying to get conference bridges working. In my meetme.conf file I have the following line [rooms] conf => 6000 In my extensions.conf file I have: exten => 1000,1,MeetMe,6000 My problem is that when I dial into extension 1000 it is telling me "this is not a valid conference number". Can anybody telling me what I'm doing wrong here?

Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]:

I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten => 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject:

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

While researching Areski's new Web-MeetMe management gui, I found some odd (from what I expected) behaviour). Using the CLI to set un/mute status works but does not update the flags, or so it appears. Starting with a fresh conference (1 user) *CLI> meetme list 3456 User #: 1 Channel: OH323/R61 Using the CLI to mute the caller (no change in the user status0 *CLI> meetme mute 3456 1

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)

I would us Meetme for conferance SIP-->SIP fist. my Meetme.conf: [rooms] conf => 9999 my extensions.conf: exten => 9999,1,MeetMe(9999) But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable

Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing

Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, asterisk-users-request at lists.digium.com wrote: > Date:

Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing

Upon setting up and configuring the my extension.conf, meetme.conf and following the instruction outlined at this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack -- Executing MeetMe("SIP/216.53.118.2-f41196e0",

Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over

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)?