search for: maxduration

Hi All!! Prompt how to record all calls passing through certain span? --- Thanks...

In my context I have: exten => _NXXXXXX,1,Set(CHANNEL(musicclass)=default) exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) but the recorded conversation ended in 10min so it = 600sec I was looking in asterisk configuration file for "600" pertaining recording but

Hello About Record(), ATFT 2nd Edition says that "if the filename contains %d, these characters will be replaced with a number incremented by one each time the file is recorded." Problem is, the documentation doesn't explain how to refer to this filename later in the dialplan :-/ In this particular example, I want to move the file to the web server's /htdocs so users can

Hello. Though I pressed # after I finished recording my voice, but application record() did not stop. dialplan is : exten => 0,1,Record(recordyourmessage.gsm) exten => 0,2,Playback(recordyourmessage) exten => 0,3,Hangup I don't want either silence and maxduration time. Can you help? Regards. Kim.

We are using Record to monitor calls. We use this because it has the option of a max time in it's call. the problem is, and I'm not at all sure it is happening in record, the recordings have sections of the conversation missing, sometimes. there is not significant pattern as to the types of calls that are having this problem. It appears quite random. we are running this on

I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air. I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.com insecure=very qualify=yes type=user context=from-pstn Then

I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS. The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log : ==== Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command' Apr 30

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When