similar to: channels change names

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

I've just started switching my project to use confbridge instead of meetme and app_conference (because of audio glitches that kept appearing in those applications). However, I can't find any way to interact with an existing confbridge conference. Surely there's some equivalent to meetme's 'meetme list' command? Anything else I can use through the cli or manager API? I

(this was also posted to the asterisk forum, but received no replies... Maybe someone here can help) I'm using the ices command to stream a conference to an icecast server. This is working nicely, for the most part, but the volume is very low. The streamed ogg vorbis audio is much quieter than what I hear in a SIP client, for example (on the same machine with the same audio hardware, of

I'm trying to set up a system so that I can record a conversation over SIP. Monitor and the like don't work so well for me, because I need to pipe the conversation to other programs in realtime, rather than record to a file, so I've been trying to use EAGI instead. (if anyone has any other suggestions about this, it would be greatly appreciated!) At this point, I'm a little

Is there a way to have an extension match on a sip address? I've tried the obvious - _.@. but it seems to behave just like _. which is no good. Is there a better way? -- Jon-o Addleman - http://redowl.dyndns.org

Hello, I made a post to the forums (http://forums.digium.com/viewtopic.php?f=1&t=72901&sid=3d5c2717ca5ab7ad676957ae436d4b51) but haven't received any replies, so thought I'd try here. On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been noticing that there's a problem with conferences (using both meetme and app_conference) and the audio sent out to an

I'm having trouble getting an IE client to hear mp3 streams through a flash player. It appears to be the same problem as described at http://icecast.imux.net/viewtopic.php?t=2039 - the flash player connects to the icecast server and begins downloading the audio data, but never actually plays it. I've tried all the suggestions from that thread - I patched icecast so that the Content-length

I have a project here that involves streaming conversations out to an icecast server, and it would be great if asterisk were able to do this nicely. So far, I've got it working by using a simple dialplan like this: exten => 22,1,MixMonitor(test.wav) exten => 22,2,Dial(SIP/blabla@blabla.com) No problems at all if I record to a file, but then I made test.wav a fifo, and had oggenc read

All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far, a piece of

I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a bit slow. I can use sox to increase the speed, but then the pitch changes and she starts to sound like a chipmunk. Any audio experts out there know how I can increase the speed a little bit, and change the pitch accordingly so that she sounds ... normal? Thanks Doug.

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

I have just finished implementing an Asterisk system for my place of business (first one), and after three days of flawless usage, Asterisk seems to have crashed. I wasn't running with '-g', so I don't have a core dump. Here's the sequence of events leading up to the crash: 1. call comes in on our TDM2400P 2. all of our phones (about 26 Polycoms) ring. (it's after

Hi, I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

I have a very old server that is used only for conferences on Meetme. To manage the conference rooms we use Web Meetme. Now it is time to upgrade everything but since Meetme is no longer available I need to find a replacement GUI to manage the conference rooms. Anyone know a solution that works with Confbridge? I found "Asterisk Confbridge Manager" from a russian company but it

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017