similar to: extension match sip address

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'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

(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'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

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'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/conf@default Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list 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.

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:

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

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 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

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

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

Hi, Recently We've upgraded some KVM hosts from Fedora 29 to Fedora 30 and now experience broken VM configurations on filesystem after virsh blockcommit. Commands "virsh dumpxml ..." and "virsh dumpxml --inactive ..." is showing diffrent configuration than the one on filesystem. In case of restart libvirtd or system reboot, there are broken VM xml configurations on

Hi I got some strange issues on my samba4.0.1 install yesterday. It happened a while after updating my xubuntu server 13.04 not 13.10. Everything seems to be working fine except shares. Kerberos authentication seem to function properly, also DNS works fine but shares seem semi-broken. I can't mount any shares on my Windows box, including netlogon, profiles. I have one share that is