similar to: One single record file for a meetme monitor?

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

I'm trying to record a Meetme conference to disk, but the Monitor application doesn't seem to play nicely with Meetme. In extensions.conf, I have this: exten => 1000,1,Answer exten => 1000,2,Monitor exten => 1000,3,Meetme This starts up the monitoring OK, and it records the prompts that Meetme gives, but as soon as the user enters the conference, the -out WAV file stops

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package) and

hi,all i install MeetMe module on Asterisk 1.6.2.10. when i use MeetMe to open a conference. even without 'r' option .it will record too. is this the bug of this module? my dialplan is : [95040] exten => 95040263007,1,MeetMe(95040,sM,123) the CLI output is : *CLI> == Using SIP RTP CoS mark 5 -- Executing [95040263007 at 95040:1] MeetMe("SIP/999-00000021",

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package)

Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google (

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use ChanSpy or

I have a conference set up through MeetMe and I can record each call coming in with the Monitor command. What I would like to move away from is having to then generate multiple files for the final output of these calls. On voip-info.org, there is an 'r' option to record the conference. This does not work on my 1.0.7 version of Asterisk. I looked through the app_meetme.c file and the

Hi. I am recording conferences taking place via the meetme application by using the 'r' option. When I start the conference I get the message in the CLI : "Starting recording of MeetMe Conference 8000 into file meetme-conf-rec-8000-1138265171.201.wav." No additional warnings or errors is displayed in the CLI during and after the conference. This tells me everything is fine.

dear all, i facing one big issue in CDR , i have one user connected in meetme and then and after some time one user are joined meetme bu this user was originally dialed by , AMI via Originate and when remote user picked up call he joined meetme room everything is fine , but now 1 st user kicked 2 nd user , via using menu options or by typing "meetme kick 1234 2' user disconnected

In Asterisk 1.2.4 is love being able to recording conferences. However, using the default variables, the files are being written to /var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme. If I change MEETME_RECORDINGFILE variable to something different in works, bit I lose the ability to define CONFNO as part of the file name, which is handy when sorting for users to review. I call meetme

Hello - I am fairly new to Asterisk, but we have a fully operational system with very few hiccups. Much of that is because of this list. Thanks. My question is this - We have assigned MeetMe conference IDs to all of our employees. We then setup a TN to accommodate the MeetMe() app. Everything works fine. In fact, it works great. However, I can't seem to figure out a good way to log

hello All, i have one issue with Asterisk Meetme Application i am recording through Meetme channels through option *'r'* and format for recording a file is '*wav*' lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. i have very strange problem of meetme_recording , once conference starts recording file having a *recording is 2x faster *than normal recording .

I am having problem with rsync 2.5.6 on Solaris 9. It starts to transfer my file, and then it just dumps core. Here's the transaction. path=/usr/local/bin/rsync cdndbsj:/var/opt/myFilename/ /var/opt/myFilename cmd=/usr/local/bin/ssh machine=cdndbsj user= path=/var/opt/myFilename/ cmd=/usr/local/bin/ssh cdndbsj /usr/local/bin/rsync --server --sender -vvvvlogDtprx . /var/opt/myFilename/

I'm trying to get a souped-up test machine(G5 Xserve) from Terrasoft to do some testing in a few weeks. If/when I actually get it I'll certainly post the results here. In theory the G5 should mop the floor with the Intel for high-volume Asterisk Zaptel usage, and I have heard from several Mac-heads that they have run three quad T1 digium cards on the Mac platform with no problems.

Hi all, I an using MeetMe and I would like to use the -i function to annouce the join/leave of the user. However, this require that users record their names. Is there anyway to remove this? I just want MeetMe to annouce somethig like "A new user has joined the conference" and that need not to record user's name. Is there a way to do this?? Pim

I currently have my Asterisk set up to "monitor" (record) all audio in my conference room on meetme. However, Asterisk will record an "____in.wav" and "_____out.wav" file for each user that joins the conference. Is there a way to set my extensions.conf file up so it only records when user when extension 1234 calls, for example? I'm assuming that the

Hi, I'm trying to convert a high(er) quality wav/ulaw/alaw file (captured with Record()) to a gsm file and can't get the bugger to work. The example on the page http://www.voip-info.org/wiki-Asterisk+sound+files says that: $ sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql Should work, but I get the error message: sox: Input and Output rates must be different to use resample