similar to: Asterisk 1.6.1 Won't Play Default ULAW Files

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 - 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 =>

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

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

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

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:

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: > >

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

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

thanks for your response i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue the server asterisk and the ip-phone where the number is configured are in the same network 192.168.1.X Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording

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

Justin Killen wrote: <snip> > > Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ > I get these 3 lines repeating over and over (I?m not 100% sure which > entry is first): > > [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to (slin) > > [2015-02-16

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 =>

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack --

Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]),

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

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 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

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, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile: