[asterisk-users] ices low volume
(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 course). I'm using the debian package of asterisk, version 1.2.13~dfsg-2 Any idea what could be happening here? My one guess, which is probably far off the mark, would be that asterisk is outputting the mu-law audio, and ices is expecting plain 16bit pcm. If I'm following the conversion process properly, that could result in the volume reduction I'm seeing, but I'm not sure. 'show channel ___' shows the NativeFormat, ReadFormat and WriteFormat all as 64 for the ices channel though (and all the other channels...), which I gather is supposed to be linear PCM, so maybe I'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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Code parsing error?
On Fri, May 05, 2006 at 03:06:43PM -0600, David L. West spake thusly:
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})
^^ ^
There's definitely a ) missing in this line! A good text editor with
bracket matching (I'm using vim now) makes it a lot easier to find this
sort of thing.
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Jon-o Addleman - http://redowl.dyndns.org
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Re: [Asterisk-Users] Compare to Skype
On Mon, May 01, 2006 at 12:16:18PM +0400, Jean-Michel Hiver spake thusly: This is only an issue if your SIP phone has a poor/nonexistent jitter buffer. I agree with that. Asterisk should just forward any RTP immediately and let endpoints handle the jitter buffer - unless asterisk is the endpoint itself (e.g. with phones plugged in its fxs ports). That makes sense if asterisk is just serving as a gateway, passing on audio to other machines, but if it's processing the audio on its own, that's not so good - it'd mess up recordings, for one. -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outbound calls to sip urls
On Mon, Apr 24, 2006 at 04:46:29PM +0530, Ajit spake thusly: 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. I'm no expert, but it looks simple enough to me - just use the originate action to call with something like this: Action: Originate channel: SIP/[EMAIL PROTECTED] context: testcontext extension: extensiontosendtheprompt priority: 1 So that extension will just send the prompt and then hang up. -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Faster Sound Files
On Mon, Apr 24, 2006 at 09:41:40AM -0600, Douglas Garstang spake thusly: 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? I think the 'stretch' command in sox is what you need. -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extension match sip address
Is there a way to have an extension match on a sip address? I've tried the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is no good. Is there a better way? -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension match sip address
On Fri, Apr 21, 2006 at 02:39:44PM -0600, Alejandro Mejía Evertsz spake thusly: exten = _sipuserX.,1,blah to match sipuser01, sipuser99... ? or exten = sipuser01,1,blah to match sipuser01 only ? Not to usefull when you want to match domain also :S Eg [EMAIL PROTECTED] Looks like I'll just do something like this - I realized I can have the PHP that's starting the call just add SIP to the beginning of the extension, and then asterisk can just use a substring to chop off the beginning of it. Thanks for the suggestions! -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] channels change names
I'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/[EMAIL PROTECTED] Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list of members shows the right channels, etc. The problem I'm having is that if the extension is external, the conference list shows a Local/$extension channel at the start, and then once the call is completed, it changes the channel to whatever was dialed. I'm probably not explaining it properly, but what I'd like to have happen is that I get one consistent channel name from the start 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call recording
On Thu, Apr 20, 2006 at 07:41:48PM -0400, Wai Wu spake thusly: Hi all, Is there a way to record a call conversation starting in the middle of the call? I know I can recording whole conversation with mixmonitor, but I prefer only recording certain part of the conversation. Thnx. From http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial: # w: Allow the called user to start recording after pressing *1 or what # defined in features.conf (Asterisk v1.2.x); requires # Set(DYNAMIC_FEATURES=automon) # W: Allow the calling user to start recording after pressing *1 or what # defined in features.conf (Asterisk v1.2.x); requires # Set(DYNAMIC_FEATURES=automon) See the rest of that page for more about it. I haven't used it myself, but it looks like what you need! -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] placing call with agi
On Thu, Apr 13, 2006 at 01:04:37PM -0400, Jon-o Addleman spake thusly: 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!) So, it turns out there's an Ices application buit in! took forever to find this... asterisk really could use better documentation! Now I need to get meetme working - why I need a kernel driver to do this is a little beyond me, but it should be straightforward to install... Are there any plans to get rid of this requirement anytime soon? -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] placing call with agi
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 stumped about getting the AGI script to dial out though - http://www.voip-info.org/wiki-Asterisk+AGI explains that you can't just dial out, since the script then disconnects from asterisk. I've been attempting the auto-dial call file method that that page links to, but I don't really understand how it's supposed to work. How can I connect the new call to the existing call? It seems that this method is just for starting a call from scratch, but I've already called an extension 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice volume using Monitor application
On Tue, Mar 14, 2006 at 12:22:49PM -0600, Jeff Hoppe spake thusly: 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? I've noticed the same thing. For now, I've just tweaked the volumes with soxmix, but it would be nice for that to be handled automatically, if possible. -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] streaming recordings
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/[EMAIL PROTECTED]) No problems at all if I record to a file, but then I made test.wav a fifo, and had oggenc read it, then pipe it to oggfwd, etc... This does work, but generates a pile of warnings in the asterisk console: WARNING[16235]: format_wav.c:247 update_header: Unable to find our position I can shut off the warnings, of course, using logger.conf, but it still seems kind of messy. I also need to start the commands reading that pipe manually after initiating the call. Can anyone suggest a better way to do this? I figure an AGI script could probably start the streaming commands automatically if necessary, but it seems to me that the ideal solution 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
