[Asterisk-Users] advanced audio recording agi help
I'm thinking about doing a project using asterisk that would let someone call in and save the audio to a wav file. I know it can be done using Record() or zapbarge. However, I'd like to be able to do some more complicated/interactive things such as: 1. record some audio 2. play it back 3. pause the playback in the middle 4. start recording again in the middle where I had paused it 5. skip forward or back x number of seconds within the recording 6. maybe even hit a button to delete the audio from this point back or this point forward Kind of like duplicating the basic functionality of windows sound recorder inside asterisk. What we're looking to do is to rewrite our custom built dictation system which is currently windows-only. I was wondering if anyone has done anything like this or if anyone has any ideas on how I would go about doing such a thing? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Frame too large?
I'm occasionally seeing this warning, at this point I'm not sure if it's causing any problems. I did some searches and haven't found any discussions about it. Could anyone help me out and explain what this means and what could be causing it? Thanks. Apr 14 10:39:42 WARNING[7946268]: chan_zap.c:3818 zt_write: Frame too large ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Out of trunk data space on call number 16386, dropping
Actually what he means is we can only get the error to show up while in a conference call no zap involved. Rarely if ever have we seen out of trunk on regular trunked iax calls between servers, but we can get it to happen every time (3 or 4 times every minute or two) when 2 users connected to server 2 hosted conference using sip phones and 1 user from server 1 calls in to the conference on server 2 via an iax trunk using a sip phone or a zap line. Justin Carlson wrote: how did you guys go about diableing it. Is it the threwaycalling directive in zapata.conf ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Out of trunk data space on call number 16386, dropping
I'm having the same kind of issues. We get the out of trunk data space error consistently during conference calls between asterisk servers. And occasionally on regular iax calls. Also while we're on a conference call it seems to cause other calls going out through iax to fail and also give this error. (weather its to another asterisk server or through say oneunified) If you figure this out, please let us know here. I'm pretty much at a loss as to what could be causing it. Justin Carlson wrote: Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] B-channels resetting every 60 minutes?
I'd like to jump in here because we're also experiencing the out of trunk data problem. So is this the only thing that causes the out of trunk data error? Because we are running iax between the boxes and both boxes have trunk=yes in the iax.conf entries and there is a zaptel device in both. James Sharp wrote: Are you running IAX between the boxes? Are you running IAX trunking between the boxes? If so.. Do you have trunking configured identically on both ends (trunk=yes in iax.conf)? Do you have a zaptel device (or ztdummy) in both ends? If either of those questions is no, then you'll get the out of trunk data space error and drop calls. Make sure both ends are configured the same for trunking and you have a zaptel device or ztdummy. Or just don't use trunking (trunk=no). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls from queue
I just updated to latest cvs and the problem remains. I did also notice that when the call coming in on the queue is through a Zap line (from an adtran 750 to an x100p) it logs the following just before the warnings below: pr 7 14:21:21 VERBOSE[60194841]: -- SIP/hrutter-432b answered Zap/13-1 Apr 7 14:21:21 DEBUG[60194841]: Set option TONE VERIFY, mode: MUTECONF/MAX(2) on Zap/13-1 Apr 7 14:21:21 VERBOSE[60194841]: -- Stopped music on hold on Zap/13-1 Tony Buser wrote: We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it seems to happen when the following gets logged: Apr 7 14:53:35 WARNING[60424217]: File 10 does not exist in any format Apr 7 14:53:35 WARNING[60424217]: Unable to open 10 (format G729A): No such file or directory Apr 7 14:53:35 WARNING[60424217]: Agent on SIP/hrutter-c6fa hungup on the customer. They're going to be pissed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls from queue
Hate to reply to my own message again, but I just figured it out. Nothing wrong with asterisk really, just a bad configuration. Somehow the queue line in extensions conf got changed by someone to: exten = 81003,3,Queue(receptionistq|tTH||10) Thats where the 10 was coming from. :) Could this be considered a bug? It shouldn't hang up on someone just because the wav file for an announcement can't be found? All this time we were blaming the poor receptionist. Tony Buser wrote: Apr 7 14:53:35 WARNING[60424217]: File 10 does not exist in any format Apr 7 14:53:35 WARNING[60424217]: Unable to open 10 (format G729A): No such file or directory Apr 7 14:53:35 WARNING[60424217]: Agent on SIP/hrutter-c6fa hungup on the customer. They're going to be pissed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel - Extensions fixed
That works well! Thank you very much. :) Wade J. Weppler wrote: I was able to get the extension/channel problem fixed. In version .03, change the following lines, starting at line 329: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
We're having a problem with transfering calls. Our channels are not the same as the extensions. We use words instead of numbers. So our config looks like this: SIP/HRUTTER,1,81101 Hildegard SIP/JFOLEY-GS, 2,81103 Jerry Consequently when I drag and drop to transfer a call to Jerry, it fails because it tries to transfer to an extension called JFOLEY-GS, but his extension is really 81103. Btw, might want to make the code be a little more forgiving, we could only get it to recognize the channels when we made the names in all capital letters (SIP/HRUTTER). I looked through your code to see if I could make some changes, unfortunatly I can't speak Italian! :) Nicolas Gudino wrote: http://sip.house.com.ar/operator Its a server/client combo that displays the status of your Asterisk PBX in a web browser in real time. You can also perform some actions. Hang-up channels and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so expect some bugs. I appreciate any feedback you can give me. Best regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
by the way, when I start up op_server.pl I get the following, even though everything appears to work ok. Use of uninitialized value in transliteration (tr///) at ./op_server.pl line 67, CONFIG line 35. Use of uninitialized value in string at ./op_server.pl line 68, CONFIG line 35. Use of uninitialized value in string at ./op_server.pl line 69, CONFIG line 35. Use of uninitialized value in substitution (s///) at ./op_server.pl line 78. Use of uninitialized value in concatenation (.) or string at ./op_server.pl line 79. I looked through your code to see if I could make some changes, unfortunatly I can't speak Italian! :) Me neither! I speak spanish..LOL. Woops! In case you hadn't guessed I don't speak spanish either, sorry. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Ah, yes that line was a blank line. Nicolas Gudino wrote: Try removing line 35 on your op_server.cfg, maybe its a blank line and the server does not handle that gracefuly. Its not harmfull anyways. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compiling gastman
I'm trying to compile gastman on Mandrake 9.2 using gcc 3.3.1 and I get the following error: gui.c: In function `gui_init': gui.c:944: warning: passing arg 2 of pointer to function from incompatible pointer type gui.c:944: warning: passing arg 4 of pointer to function makes pointer from integer without a cast gui.c:944: error: too few arguments to function make: *** [gui.o] Error 1 It compiles fine on Mandrake 9.1 with gcc 3.2.2. I'm thinking its a gcc issue. Anyone know what I can do to resolve this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpagi DIAL command not working
Welp I had no luck trying to get this to work. However, I came up with
a workaround that isn't as clean, but at least it works. :) Here's
what I did:
In my agi script:
$agi-agi_exec(SET VARIABLE PHONENUM $strPhone);
$agi-agi_exec(SET PRIORITY 20);
Then in extensions.conf:
exten = 70557,1,AGI(fileno.php)
exten = 70557,20,Dial(Zap/5/${PHONENUM})
exten = 70557,21,Hangup
Someone told me about the paragraph on the wiki
http://www.voip-info.org/wiki-Asterisk+AGI that says:
If the application dials outward, the script returns execution to the
dialplan and looses contact with the asterisk server. The script
continues to run in the background by itself and is free to clean up and
do post-dial processing.
That sounds to me like just execing a DIAL command from within the agi
_should_ work and yet it doesn't. Anyone have any insight?
Tony Buser wrote:
I'm using phpagi to try and call out through our x100p line with the
following code for instance:
$strPhone = 610555;
$agi-agi_exec(DIAL Zap/5/$strPhone);
On the console I see:
fileno.php: DIAL Zap/5/610555
However it doesn't dial. It just continues on to the next line of the
script and eventually times out and hangs up. If I put the following
in extensions and call it, it dials out correctly:
exten = 70558,1,Dial(Zap/5/610555)
Everything else in my code works. (ANSWER, HANGUP, agi_getdtmf,
text2wave, etc) I'm running the latest cvs as of yesterday. What am
I doing wrong?
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[Asterisk-Users] phpagi DIAL command not working
I'm using phpagi to try and call out through our x100p line with the following code for instance: $strPhone = 610555; $agi-agi_exec(DIAL Zap/5/$strPhone); On the console I see: fileno.php: DIAL Zap/5/610555 However it doesn't dial. It just continues on to the next line of the script and eventually times out and hangs up. If I put the following in extensions and call it, it dials out correctly: exten = 70558,1,Dial(Zap/5/610555) Everything else in my code works. (ANSWER, HANGUP, agi_getdtmf, text2wave, etc) I'm running the latest cvs as of yesterday. What am I doing wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival voices
Thanks that works. :) So you use cepstral voices in festival? I thought cepstral was a whole seperate system. I still think the slt_arctic_hts voice from http://festvox.org/voicedemos.html sounds better then the regular cepstral voices. Brian West wrote: (Parameter.set 'Audio_Method 'linux16audio) ;(Parameter.set 'Audio_Method 'esdaudio) ;(Parameter.set 'Audio_Method 'mplayeraudio) ;(Parameter.set 'Audio_Method 'sunaudio) ; American female I'm using the cepstral frank with festival ;) (set! voice_default 'voice_frank) in /root/.festivalrc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] festival voices
Hi, I'm new to both asterisk and festival. I'm trying to figure out how to change the voice festival uses. For example, I've downloaded don_diphone to festival/lib/voices/english. I then edited /etc/asterisk/festival.conf and changed the festival command to: festivalcommand=(voice_don_diphone)(tts_textasterisk %s 'file)(quit)\n Started up festival_server, then started up asterisk, call the extension and it fails to speak with the following message on the console: == Parsing '/etc/asterisk/festival.conf': == Parsing '/etc/asterisk/festival.conf': Found Feb 12 11:27:40 WARNING[409626]: app_festival.c:434 festival_exec: Festival returned LP : don_diphone I'm thinking its because when you change the voice in festival it returns the name of the new voice which may be confusing asterisk because it's expecting something else? Also, if anyone else out there is using festival, whats the best most natural sounding voice? So far the best I've found were from here: http://hts.ics.nitech.ac.jp/download.html Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival voices
Chris Albertson wrote: try adding a set of parens like this: festivalcommand=((voice_don_diphone)(tts_textasterisk %s'file)(quit))\n Unfortunately that results in the following error at the asterisk console: Feb 12 19:45:27 WARNING[409626]: app_festival.c:437 festival_exec: Festival returned ER And the following error in the festival_server: SIOD ERROR: unbound variable : don_diphone SIOD ERROR: unbound variable : \n Have you seen festivox? It's a tool for building voices The key to making festival sound natural is to get the timming and entonation right. The astrisk app uses festivels demo test to speech application which is just that a quick dirty demo. Have you seen the markup language on the CMU site? http://www-2.cs.cmu.edu/~awb/festival_demos/sable.html Sable can do MUCH better then the simple tts application. Thanks, I'll take a look at that. So to use Sable I'd have to use festival from like an AGI script and not inside the Festival() function in extension.conf? I blindly tried pasting sable markup in there and the best I could do was get it to read back the markup and all. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
