Displaying 11 results from an estimated 11 matches for "voicefiles".
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voicefile
2008 Feb 08
2
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
3
Call quality - how to debug
Hi All,
I've a 1.4.15 A*k server supporting several users (approx 80 total, but
<10 sim calls usually). I've one user who complains of intermittent bad
calls, though I suspect the bad calls are across the board, but
intermittent.
Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
Asterisk never uses more than 4-5% cpu, systems idle besides that.
Memory seems
2004 Dec 02
1
Agent Login "Play a file"
Good Day list,
Anyone know if there is a way to have the AgentCallBackLogin
function play a voice file after the agent picks up the phone?
If 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)
2013 May 24
0
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
2006 Feb 20
0
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: arjan.kroon@mobillion.nl
internet: www.mobillion.nl
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2009 May 20
0
inbound SIP funnies
Hi,
I've a few working asterisk servers, all seeing the same symptom, but
they are all based on the same configs.
A SIP inbound INVITE message is coming in to an extension (not a peer)
eg 555 at ourserver.com
A tcpdump clearly shows the INVITE coming in, but asterisk seems to be
ignoring it (theres no reply outbound packet). All the source/dest IPs
and ports look good.
A
2013 Oct 01
0
Direct DAHDI documentation
Hello,
I wanted to switch from using Dialogic/Eicon cards to using Digium's T-1 cards. When I purchased a sample card the salesperson assured me there was documentation specific to the DAHDI interface. Now that I'm digging in, I'm finding it's documented a lot like Linux -- one must read the fairly uncommented source code.
I don't have a problem with this generally, but here
2013 Mar 07
2
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 => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})
2006 Nov 28
1
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
dialled. I need to
2010 Jun 23
0
50 mantis issues marked 'Ready for Testing'
List,
Over the last few months we have managed to bring the total number of
issue on the tracker from 610+ to 537 (as of writing). While this is
good news, we still have a number of open issues that require testers
to help move them along. Below, I have posted the oldest 50 issues
that are in the 'Ready for Testing' state.
Basically, we are looking for more people to step-up and test
2013 Sep 26
0
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