similar to: Faster Sound Files

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'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

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

(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

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

I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug.

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

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

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

A playback of a mono vorbis track on a system using the i810 audio driver via ogg123 results in a double speed playback (chipmunk sound). It seems that this chipset doesn't support mono and while xmms seems to compensate for it, ogg123 doesn't I have a bugzilla report at redhat.com filed on this with more details https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=90433 Regards, Yusuf

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

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: > > On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: > >> Is it possible to switch the Left / Right channel when encoding in FLAC? >> (I have some flac files with incorrect left/right channel assignment that I would like to re-encode with flac) >> _____________________________

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading