similar to: voicemail problems with CVS-HEAD

Hi List. Don't know if I already posted about this problem but, if I have I apologize for the double post. I am trying to test a time of day extension dialing 80, all I'm trying to test is if is morning I would like asterisk to say "Good Morning" but, when I run the test I get the following error message saying that the file doesn't exist and it does: Night..............

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards

Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

part of extensions.conf: exten => 11,1,Answer() exten => 11,n,NoOp(CallerID : ${CALLERID(all)}) exten => 11,n,Playback(/tmp/welkom-tcs.alaw) exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1) ; wordt doorgerouteerd naar context open, maar indien gesloten : exten => 11,n,NoOp(Oproep tijdens winkel gesloten) exten => 11,n,Playback(/tmp/winkel-gesloten.alaw) exten =>

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

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with "envelope=yes" and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full:

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx at Accueil_HNO:2] BackGround("SIP/SIP000005-0000000c", "Fermeture") in new stack [Jun 16 07:35:06] WARNING[7634][C-00000006]: file.c:701 ast_openstream_full: File Fermeture does

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4

An outside caller started to leave voice mail. The CLI shows: Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: gsm, 0x8295d40 -- x=1, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: wav, 0x829e2c0 -- User cancelled by pressing 0 -- Playing 'vm-saveoper' (language 'en') Later on,

Need some help understanding codec preferences: I have 2 asterisk servers. Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and allow=ulaw in iax.conf Server 2 receives calls and routes them to server 1. It has the same allow lines. We receive calls from a phone co and route them via server 2 to server 1. The calls originate in g729 and everything works fine. Now I want to take

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. cat meetme.conf [rooms] conf => 600 extensions.conf: [meetme] exten => 2663,1,MeetMe(,D) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Hi all! I would like to make a service with recording sounds and playing back to caller. I had wrote the script but it failed at Read statement with file not found error. I have put some file test into script and this is what happen on verbose level 9. -- Executing [8298 at default:8] Record("DAHDI/27-1",

Hi everyone, I'm having an odd issue. I've been doing some testing over the past couple weeks on some Asterisk modules / utilities, but have bumped into a problem which I can't seem to resolve. Asterisk can't seem to play the default sound files (ULAW) in my environment. All necessary debugging information is included below. I'd love to get anyone else's thoughts on this,