Displaying 20 results from an estimated 1000 matches similar to: "cpu usage for G.729 codec"
2007 May 18
0
Re: asterisk-users Digest, Vol 34, Issue 82
If I use Asterisk to initiate two call legs with a callfile, dialing
the channel and setting the extension to an AGI that dials another
channel, and both dial by SIP connection to a switch that allows only
G.729, do I need a G.729 codec running on Asterisk? Do I need 2?
And if I use the callfile to connect by SIP to a switch that allows
only G.729, then use the extension AGI to play a file
2005 Mar 03
4
[OT] - Why should I answer a Newbie question, therethick!
If you really want to do this the asterisk list is based off of mailman.
You can learn all about mailman here:
http://list.org/
But really, what are the odds that newbs will know to go there first?
Are you going to moderate it? Someone has to actually answer the
questions you know, if a newb only list is going to exist.
Look, don't answer lame questions if you don't want to. Flaming a
2007 Feb 05
0
Callfiles to Meetme Fails (was: RE: Using Local Channels with Originate)
I have Meetme conferencing compiled for Debian as per
http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the
outgoing dir, and it intitiates a call to a local extension as a
channel, but the call seems to block on the Meetme() command. That
extension completes the outgoing Dial(SIP) command to my phone,
announcing that leg is the only member of the conference, and just
waits. If I
2014 May 15
1
Call file problem, DelayedRetry/retrying spite MaxRetries: 0
I am using Realtime extensions as well, in case that would matter.
Following problem arises from time to time, a call will successfully
terminate:
[May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing
[t at project_init:1] Hangup("SIP/peer-2-00002f7e", "")
[May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init,
t, 1) exited non-zero on
2006 Nov 26
0
Dialout to Meetme Fails?
I'm trying to use Asterisk (v1.2.11) make a callback that dials both
legs of the call into a Meetme() room together, but I keep getting
"conf-invalid" messages.
I created a callfile (/var/spool/asterisk/outgoing/out.call) that
specifies a Local channel (extension) which contains a Dial() command to
the "dialer", and an extension which contains a Dial() command to the
2010 Dec 21
1
SOLVED: Re: Setting `userfield` from within a callfile
On Monday 20 Dec 2010, Olivier wrote:
> 2010/12/20 A J Stiles <asterisk_list at earthshod.co.uk>
>
> > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
> > (written by someone else before me) which sets up calls by creating
> > files of
> > the general form
> >
> > Channel: SIP/$INSIDE_NUMBER
> > Context: $CONTEXT
>
2005 Oct 07
0
Asterisk to CCM Message Waiting Indicator
I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I have just about everything working except for the message waiting indicator.
I have the following setup in context [ccm] in my extensions.conf file:
;MWI
exten => _2807XXX,1,SetCallerID(${EXTEN:3})
exten => _2807XXX,2,Dial(SIP/28888@65.202.115.240)
exten => _2807XXX,3,Answer
exten => _2807XXX,4,Wait,1
2005 Sep 01
0
How to set CLIR when using call files ?
Hi all,
A few days ago I found out with help of some of you guys how to set CLIR.
(Calling line identification restriction) My first idea was to use the
keypad protocol to set the CLIR with dialing *31* before the number but this
was not possible.
So thanks to Damon Estep I got it to work with executing
'SetCallerPres(prohib)' before the dial command. This works perfectly! But
now
2010 Jul 01
1
call file question
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten => s,1,Answer
exten => s,n,Dial(DAHDI/4-1/*717157750)
exten => s,n,Verbose(${DIALSTATUS})
exten => s,n,Hangup
[custom-callfwdcanc]
exten => s,1,Answer
exten
2010 Apr 13
0
Problem with Callfiles
Hi!
I am trying to do a callfiel for autodialing but when I move the callfile to outdialing folder asterisk seems like if did the call but it doesnt.
I put here my callfile and that I get when asterisk begins to do the call
If anybody has idea, pls. Tell me
TIA
;;----CallFile-----
Channel: Zap/g1/8093908270
Callerid: 8093908270
MaxRetries: 2
RetryTime: 300
WaitTime: 45
2005 Jun 09
23
Voicemail and MS Exchange Synchronization
We have a customer considering migrating from a large Nortel PBX with a
third-party voicemail system to Asterisk but one of the features they
really like is the automatic synchronization of voicemail between
Exchange and their voicemail system -- delete a message from the
voicemail system and it is deleted from their email inbox and vice versa.
Searching has not revealed anything like this
2005 Jun 02
0
Call Manager & Asterisk for VM - MWI not working
Like some other people on here, I am trying to integrate Asterisk for VM
with CCM version 3.x. I've got gnugk and Asterisk running, I've got CCM
registering with the GK, I've got the voicemail pilot and profiles
setup. A call comes into a CCM phone, it rings, rolls to the correct VM
on ASterisk and asterisk emails the voicemail and I can check the
voicemail, but I cannot get MWI
2015 Feb 18
1
Callfile problem - Unable to find codec translation path from (nothing)
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]),
2005 May 31
3
Opinions of Sphinx?
I'm planning a system of 120 SIP or PRI channels using speech
recognition (fixed grammar of 500 words) menus.
I could use a Cisco router and VoiceXML, but would prefer not to on cost
grounds.
Has anyone tried Asterisk and Sphinx (bonus points if in a production
environment)? If so, what's your opinion on quality of recognition,
stability, resource usage, etc?
Anyone have any
2009 Sep 02
1
Skype for Asterisk callfile question
Hi list,
To make outgoing calls by skype i would like to have our crm app create
callfiles like we do for normal calls.
If i read the instructions it says this :
---quote---
The syntax for making an outgoing call using Skype for Asterisk is as
follows:
Dial(Skype/[<originator>@]<destination>)
---unquote---
So i create a callfile that looks like this:
---
Channel: SIP/228
2005 Oct 05
0
Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o memory leak when using call files ?
Hi all,
I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P
(EuroISDN cpe)
connected to another similar asterisk box B acting as EuroISDN master.
I'm performing some load tests by contiously feeding up to concurrent 30
call files to /var/spool/asterisk/outgoing/ on box A
which inititate via a dialplan context/extension a outbound call
(redirected via chan_local) to
2010 Jan 17
0
How to escape the Pound-Char in Callfiles
Hello,
I'm using Asterisk 1.6.2.0 and I like to call extension #8 from callfile.
Unfortunately the #-char ist interpreted just as comment.
I got a "Invalid file contents in /var/spool/asterisk/outgoing/callfile,
deleting" from asterisk.
When I try to escape with \ oder use quotes, I got: \#8,1 failed so falling
back to exten 's' or "#8",1 failed so falling back
2009 Oct 09
1
${REASON} not getting set.
Hi all,
I've got a program that creates a callfile and most if it working great.
However, when a call fails, I'm trying to capture the reason, which I'm told
should be in the ${REASON} channel variable.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Here is an excerpt from the callfile:
Channel: local/155555555
Callerid:Tests <155555555>
MaxRetries: 0
RetryTime:
2011 Feb 24
2
[1.4] Still can't get it to call back
Hello
No matter what I try, Asterisk still fails dialing back through a
callfile built through an AGI script.
The whole thing works fine when the original call that triggers
Asterisk is from an internal extension (Xlite), but it fails when it's
from my cellphone ringing through the FXO/Zaptel port and I want to
wait a few seconds and call back through the FXO/Zaptel.
Could it that even
2004 Aug 23
1
H323 outgoing calls
Does asterisk support using an H.323 provider for outgoing calls? From
everything I have found, it looks like it does. However, I have had no
success in getting it to work. I would really appreciate if somebody
could give me a hand. I am using the channel that comes with asterisk.
I have also tried using the channel from inaccessnetoworks but have not
had any more success. My provider