similar to: codec preference

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, 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

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

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:

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

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,

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit <salah.elharit200 at gmail.com> wrote: > 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

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

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

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

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..............

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

hi, i try to convert wav file to gsm format.use following commands; sox net263-welcome.wav -r 8000 -g -c 1 net263-welcome.gsm resample -ql the file is located in /var/lib/asterisk/sounds/net263 but cant' play.do you know what's wrong? -- Executing Playback("SIP/1001-00000091", "net263/net263-welcome") -- <SIP/1001-00000091> Playing

Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -------------------------- [number-to-voice] exten => 8765,1,playback(num:344345,say) exten => 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk.

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but

tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks and regards 2015-03-25 12:59 GMT+00:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit > <salah.elharit200 at gmail.com> wrote: > >

Hi all! In my extensions I have the following: exten => 999,1,Answer() exten => 999,2,PlayBack(beeperr) In /var/lib/asterisk/sounds/ I have both beeperr.gsm & beeperr.ulaw, both with '-rw-r--r--' permissions. when I dial extension 999 I get: ************************************ -- Executing Answer("SIP/asterisk.domain.com-081477a0", "") in new stack