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
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
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[1895]:
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 one local phone to
On 24 Aug 2010, at 04:30, Tim Nelson wrote:
- Tim Nelson tnel...@rockbochs.com 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
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.com
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
[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 at
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.
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
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 problem
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_full: File
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
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
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 jonas.kell...@telenet.be
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
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 command
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 Sat, Aug 28, 2010 at 9:52 AM, Tilghman Lesher tles...@digium.com 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
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
I
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 dchannel
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),
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
the
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
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
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)
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 all means configure
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