similar to: Using a Dial Statement with option m and t

Hey, I've coded an OGG player for Win32 (it uses AL for playback so it's portable to Linux/Mac), but every time the program gets to the 'ov_open()' function, the app completely freezes, and I have to use the task-manager to kill it. I am supplying it with a valid file handle that was just opened (FILE*) and the vorbis file is also a pointer that is not in use (set to null). Any

Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -

Hi, I am searching for the most effective solution for the following scenario: Our users can call into our IVR menu and dial a specific extension and immediately hang up. This event should simply trigger Asterisk to make multiple simultaneous calls through a group of zap channels (5-10 calls). When the called parties answer, Asterisk should simply play a message and hangup. So I was thinking

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] <all.eforums at gmail.com> wrote: > Hello All, > > Probably a silly question, but we're wondering if people have had any > experience and have data to demonstrate if the performance of the Asterisk > system might suffer in terms of latency etc. if we're to have it retrieve > sound files from a database using odbc as

you can do this with MeetMe, but you don't have to. you can also use Parking, which makes things a little simpler. in either case, the strategy is going to be something like: 1. Record the soundfile 2. Park the inbound caller 3. Use a .call file or the manager interface to initiate an outbound call to the mobile phone 4. play soundfile and prompt the mobile phone user to accept/reject the

Hello list, it seems that a sound file is not present on my system, although I have made a standard install... [Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File vm-INBOXs does not exist in any format [Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to open vm-INBOXs (format 0x8 (alaw)): No such file or directory I do not find this particular soundfile

Hi List, I am writing an audio player that exclusively plays FLAC sound files, with CUE sheets. It is written in Python, so it is cross-platform, and it is working very well so far. The soundfile IO is handled by the Audiere library. For metadata (aside from the CUE sheet), I make system calls to metaflac to do things like extract album art for display, and I have a question concerning metaflac.

On 24-Jun-2001 Matt Zimmerman wrote: >> [...] >> ERROR during encoding, state = 15:FLAC__ENCODER_MEMORY_ALLOCATION_ERROR > > This error appeared in the other report as well. It looks like a memory > allocation failure is the cause of the problem. Is the error easily > reproducible given the failed WAV file? Yes, always the same error - but only (no joke) on option -8 ,

Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround("SIP/joa-9def", "tt-weasels") in

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten => s,1,Dial(SIP/50,23,r) exten => s,2,VoiceMail(u50@default) exten => s,3,Playback(vm-goodbye) exten => s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

Hi, in the earlier version there was a sounds.txt with the transcript of the soundfiles. Does this still exist somewhere? Is there a plan to make speech synthesis available the same way as soundfiles, ie. instead of playing language/soundfile.wav, send the text to the speechengine and play the output...? Jay... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm trying to get asterisk to auto-dail out. I created a *.call file with the the top of it being "Channel: Zap/1/2609944", which should have connected to Zap channel 1 and dial out to 2609944, but It did not do so, asterisk would say a call was completed to Zap/1/2609944 but I never heard that phone ring. So I tried just putting "Channel: Zap/1" at the top of

All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help.

Hi, In voicemail.conf: ; Supported values: ; 'filename' filename of a soundfile (single ticks around the filename ; required) ; ${VAR} variable substitution ; A or a Day of week (Saturday, Sunday, ...) ; B or b or h Month name (January, February, ...) ; d or e numeric day of month (first, second, ..., thirty-first) ; Y Year ; I or l

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten

yes, robert, but how do i "join" the two legs inside a call file or in the dialplan? i have experienced that call files can do a call to a channel and if this call is answered it can either be connected to an extension inside a context or call an application with parameters. roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com