Thanks Steve,
Not sure how this would allow the caller to ff / rw the file currently
being played - would that portion have to be written in the external
program ?
Are there any examples of how to use externalivr anywhere (I can't
find on google)
TIA
Julian
On 29 August 2010 01:29, Steve Edwar
>
>> language=en
>> context=from-pstn
>> switchtype=national
>> signalling = pri_cpe
>> group=1
>> channel => 1-12
>> ---
>>
> (1-12? not 1-23?)
>
>
Thats what he had originally in the file. I assumed he only wanted the
first 12 channels. If that was an error, then by
On Saturday 28 August 2010 20:27:23 Andrew Joakimsen trolled:
> As recent as 2008 "Asterisk 1.4 is feature frozen" if that is the case
> how come now CallingToken support is added? I don't really know what
> this is but all I know is:
>
> 1) Callingtoken adds new options to the config files
> 2) Ca
As recent as 2008 "Asterisk 1.4 is feature frozen" if that is the case
how come now CallingToken support is added? I don't really know what
this is but all I know is:
1) Callingtoken adds new options to the config files
2) Callingtoken is some new protocol in IAX?
3) Upgrading asterisk 1.4 breaks
I have 2 FXO channels from which I want to route incoming calls to
different contexts in extensions.conf. I edited the context entries in
dahdi-channels.conf and created matching entries in extensions.conf.
One channel is routed to the new context as I want, but the other
channel is stuck going
On Sat, 28 Aug 2010, Julian Lyndon-Smith wrote:
> I want to be able to allow a caller to dial a ddi, system to verify
> identity etc (this is all done)
>
> I then want them to sit listening to music, until an event happens.
> When this (external) event happens, I want to play a certain file to
> t
I want to be able to allow a caller to dial a ddi, system to verify
identity etc (this is all done)
I then want them to sit listening to music, until an event happens.
When this (external) event happens, I want to play a certain file to
the caller, using playback (so that they have ff / rw etc), a
On Sat, Aug 28, 2010 at 01:32:13PM -0400, Andres wrote:
> On 8/28/2010 12:59 PM, jeremy.hellst...@synovate.com wrote:
> >
> > I’m not surprised both the conf file and myself are confused.
> >
> > __
> >
> > I still end up with messages telling me that a d
On 8/28/2010 12:59 PM, jeremy.hellst...@synovate.com wrote:
>
> I’m not surprised both the conf file and myself are confused.
>
> __
>
> I still end up with messages telling me that a dchannel cannot be
> found. Any other suggestions?
>
> Thanks, Jeremy
On Sat, Aug 28, 2010 at 9:52 AM, Tilghman Lesher wrote:
> On Saturday 28 August 2010 10:35:59 Bryant Zimmerman wrote:
> > Thanks for sharing I appericate your insight as this is something I run
> up
> > against as well.
> > What about g729 we use this coded a lot what is the best method to
> > tr
I'm not surprised both the conf file and myself are confused.
I've pared things down in chan_dahdi.conf to ...
_
[channels]
spanmap => 1,1,0,esf,b8zs
#include dahdi-channels.conf
switchtype => national
signalling => pri_cpe
context => default
___
On Saturday 28 August 2010 10:35:59 Bryant Zimmerman wrote:
> Thanks for sharing I appericate your insight as this is something I run up
> against as well.
> What about g729 we use this coded a lot what is the best method to
> transcode it it?
If you load "res_convert.so", you will have a CLI comm
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens
wrote:
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
Hi all,
We're trying to make voice and SMS apps easier and more common. We solved
one part of the problem with pay-as-you-go cloud-scale Asterisk hosting, and
now we're trying to make the app setup easier. With a few exceptions, setup
docs are too rare, and they depend on knowing too much about
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens wrote:
> 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
On Saturday 28 August 2010 04:22:18 Jonas Kellens wrote:
> 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_ful
On Saturday 28 August 2010 04:47:37 Jonas Kellens wrote:
> What I saw was that Asterisk stumbles when putting a comment like this :
>
> ;--> bla bla !!!
>
> It should be :
>
> ; --> bla bla !!!
>
> So with a space between ; and -->
>
>
> The rest of my dialplan came available when doing this... So
Xavier D. wrote:
> Yes but what about the conference number ?
>
You can pass that on via the dial plan. I'm using mysql to setup
dynamic conferences. A snippet below:
; ***
; Get conference room number, if number entered is 5812
; jump to
Oh, and that isn't how a spanmap looks either. It looks like you have mixed
some stuff from system.conf and chan_dahdi.conf here. My guess is your
system.conf is configured at least mostly right, and that is why everything
goes green.
http://svn.asterisk.org/svn/dahdi/tools/branches/2.3/system.con
> [trunkgroups]
> trunkgroup => 1,24
> spanmap => 1,1,0,esf,b8zs
If you're only using one span, is there a reason you are using trunkgroups?
I believe those only get used for NFAS and GR-303
> #include /etc/asterisk/dahdi-channels.conf
Do you have anything defined in this file? Since it comes
I have found it best when doing remarks to not use the ;- combination as I
have seen it cause failuers on dialplan reload.
Bryant
What I saw was that Asterisk stumbles when putting a comment like this :
;--> bla bla !!!
It should be :
; --> bla bla !!!
So with a space between ; and -->
The
What I saw was that Asterisk stumbles when putting a comment like this :
;--> bla bla !!!
It should be :
; --> bla bla !!!
So with a space between ; and -->
The rest of my dialplan came available when doing this... So problem solved.
Jonas.
On 08/28/2010 11:25 AM, kisho...@techroutes.co
On 24 Aug 2010, at 04:30, Tim Nelson wrote:
> - "Tim Nelson" wrote:
>> Greetings all-
>>
>> Here's an odd question. Supposedly, IAX2 now has the ability to
>> operate with signaling and media in separate streams, very much like
>> SIP. I've read about this feature here[1] and there[2], but
becaus your call rules are are mismatched...
kishor
> First of all explan your dial plan and extensions.
>
> i will resolve that...
>
> Regards,
>
> Kishor kumar
>
>
>
>> Hello list,
>>
>> yesterday I finished work having my whole dialplan available...
>>
>> Today I want to make a call from
First of all explan your dial plan and extensions.
i will resolve that...
Regards,
Kishor kumar
> Hello list,
>
> yesterday I finished work having my whole dialplan available...
>
> Today I want to make a call from one local phone to another and I get this
> :
>
> [Aug 28 10:48:57] NOTICE
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
Hello list,
yesterday I finished work having my whole dialplan available...
Today I want to make a call from one local phone to another and I get this :
[Aug 28 10:48:57] NOTICE[1895]: chan_sip.c:15144 handle_request_invite:
Call from 'test2' to extension '60' rejected because extension not fo
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