search for: voicefile

Displaying 11 results from an estimated 11 matches for "voicefile".

2008 Feb 08
Upgrade 1.2 -> 1.4 voice files
Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian
2009 Jun 02
Call quality - how to debug
...well. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is < 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, with...
2004 Dec 02
Agent Login "Play a file"
...f this is not an available feature, any ideas on the difficulty in making this feature? Example: Extensions.conf exten?=>?700,1,AgentCallbackLogin(${CALLERIDNUM}|?AnnounceCAllQue-TechSu pport?); ....... exten => s,6,Queue(queue1) Agents.conf agent => 2204,1234,Ron Hartmann VoiceFile... AnnounceCAllQue-TechSupport contains Allison saying "This call is from the Technical Support Call Queue Please press # to accept the call" Ron Hartmann dials 700, enters the agentid and password and extension he is working at. A Call comes into the office... the call is transfe...
2013 May 24
Pri-Debug-Log / Is Early Media supported by provider?
Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the following: (I replaced originating caller id by 123456) PRI Span: 1 &l...
2006 Feb 20
automatically start application from thecommandprompt
...bject: [Asterisk-Users] automatically start application from the commandprompt Hello, Is it possible to start an asterisk application from the command prompt? This application has to dial to a number. When the calling party picks up the phone, the asterisk application had to play certain voicefiles. Kind Regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: internet: -------------- next part -------------- An H...
2009 May 20
inbound SIP funnies
...r(0) ; Pick up phone instantly exten => 555,n,Playback(vq51) ; Let them know what's going on exten => 555,n,Playback(vq20) exten => 555,n,Goto(default,555,3) ; repeat So as far as I can tell, we should be accepting the connection and playing the voicefile (yup - I know this would be open to the internet, that's the intention). Sip.conf also has: allowexternalinvites=yes allowexternaldomains=yes so it should be working I think... This is a 1.4.15 based asterisk Thanks Adrian -------------- next part -------------- An HTM...
2013 Oct 01
Direct DAHDI documentation
...Either way, I'm not sure how the D-channel data is flowing. 3) I got the idea that B-channel data is collected by the kernel module in 8 sample blocks (1 ms). Does this mean I need to be reading it out/writing it in at that rate? I saw some buffering code, but wasn't sure if that was voicefile type playback/record or if all audio is treated without regard to its source/destination. I guess I could lock onto it at 1ms using Linux's HPET timer, although that sounds clumsy. 4) I can certainly convert between ulaw/linear to sum for conferencing, but it seems the kernel modu...
2013 Mar 07
Recording with MixMonitor and AGI
Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten =>
2006 Nov 28
Call recording filename
I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who
2010 Jun 23
50 mantis issues marked 'Ready for Testing' searching for free devices in a group of phones [patch] Opening voice channel on FastStartAcknowledged before Answer. Remove H245inSetupOptions for better capability. [patch] Add voicefile and dtmf options to res/res_agi.c [patch] MGCP Business Phone Packages patch [patch] chan_mgcp new feature: digitmaps definitions
2013 Sep 26
Asterisk / SIP-Call / AGI-Script / SIGHUP after Answer
Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find my SIP-Debug-Output. I can see a "BYE"-Message. But why? AGI-Debug-Messages: (yes - I can the result is -1 > but why? Normally it...