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...on the asterisk server which ran the eagi script in the first place. Can the new outgoing connection be attached to that call? I'm not sure if my description will make sense to anyone else, but please let me know if there's any way I can clarify things! Thanks! -- Jon-o Addleman - http://redowl.dyndns.org

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

...'m completely wrong. As a workaround, is there any way that I could boost the volume just for the audio going to ices? I don't want to raise the volume for the conference as a whole if possible, since that's sounding just fine for those talking. Thanks! -- Jon-o Addleman - http://www.redowl.ca

...an'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 just need to list conferences and members. Thanks! -- Jon-o Addleman - http://www.redowl.ca

...Exceptionally long voice queue length queuing to Local/conference at veco-044d;1 [Feb 10 14:14:40] WARNING[15571]: channel.c:1045 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/conference at veco-044d;1 [also repeated many, many times] -- Jon-o Addleman - http://www.redowl.ca

...200 OK Content-Type: audio/mpeg icy-name:no name icy-pub:0 Server: Icecast 2.3.1 Content-Length: 319324133 Surely this is a problem that has been encountered before! Let me know if there's any other wireshark captures that might help, or anything else. Thanks! -- Jon-o Addleman - http://www.redowl.ca

...nsion testmeet at test originate Local/testsound at test extension testmeet at test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca

...lution would be to avoid the use of the FIFO entirely, since asterisk expects to be writing to a real file. I had a quick look at EAGI, but wasn't sure if that would be best either, but I'll have another look if anyone thinks it would be a good approach. Thanks! -- Jon-o Addleman - http://redowl.dyndns.org

...of the connection - it doesn't matter what it is, as long as it doesn't change. As things stand, the conference list isn't accurate, unless I wait about 5 seconds after adding someone before updating the list. Thanks for any suggestions you might have here! -- Jon-o Addleman - http://redowl.dyndns.org

..., tf_count => 50, tf_diff => 950, tf_frequency => 19.0000 There are several lines like this, with tf_diff varying between 912-1083, and tf_frequency from 18.24-21.66. Does anyone know what this might mean? Thanks for any help you might be able to offer! -- Jon-o Addleman - http://www.redowl.ca

I just switched from 1.4.30 to 1.6.2 I initiated a call file - same way in 1.4.30 and nothing happened. I was not aware of changes in the call file to 1.6.2? I was watching the cli and no error showed or anything. In the manager.conf I have things setup. [MyDial] secret=#### permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user Any

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.

Hi there. Anybody knows how can I send metadata updates to a running mountpoint in Icecast? Thanks -- __________________________________________________________ | , , | | / \ | | ((__-^^-,-^^-__)) Octavio Rossell Tabet | | `-_---' `---_-' octavio at

This is exactly what I want. I have a continuous stream to a live mountpoint that needs to be metadata updates. Is this http request you mention any documentation to study? El 02/09/10 08:43, Johann Soukup escribi?: > Hi Octavio, > > I guess you are talking about live mount points. > > We use a http request through the admin/metadata application of the > Icecast admin

I am using the Monitor() application (with soxmix for combining the audios) and the voice connected to the phone network is recorded at a lower volume then the voice connected directory to the Zap analog phone card. How can I get both the audios to be at the same volume on recording? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hallo everyone in the Icecast project. First of all i want to apologize in case this message should not be in this list but i believe this is the place to answer my question. I know only a little about icecast server. I would like to build up a live radio station. The concept is something like : the server is set up in some place and all producers broadcast themselves from their place, so

Hi, I wish to use the manager API to make an outbound call to a sip url,subsequently play a prompt and hangup.Any hints on how to acheive this/feasability will be much appreciated. Regards, Ajit

This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten => 1,1,Set(target=${CHANNEL:4}-) exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) exten => 1,n,VoiceMailMain(${target}) However, every time it runs I get an error in the CLI as follows WARNING[5629]: pbx.c:1366 ast_func_read: Can't

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