On Wed, 18 Apr 2012 06:12:45 -0700 (PDT)
bilal ghayyad wrote:
> Yes, first thing I do is the make all and make install for dahdi,
> then I do ./configure and make and make install for asterisk. But I
> do not find the chan_dahdi under the /usr/lib/asterisk/modules. WHY?
You probably need to run
Dears;
I see this at the /var/log/asterisk/messages:
[Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open
/dev/dahdi/transcode: No such file or directory
Again, I am installing asterisk and dahdi at Ubuntu (uname -a
Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012
AJ-
> On Thursday 19 April 2012, samuel wrote:
>> Just in case it helps:
>>
>> It turned out that from asterisk version 1.8.4 on, the g729 binaries are
>> different from the previous versions so it was a version mismatch between
>> the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
>>
>> Than
Hello, I'm trying to get s extensions to autoanswer to Centos computer
speakers, the computer is a Dell Optiplex 170L embeded sound card. I'm
running Centos 6.2 i386 with Asterisk 1.8.10
Does anybody know how to fix error below?
-- Executing [s@default:1] Dial("SIP/publicip-0001",
"conso
This is your website:
http://www.convergia.com/
Thanks in advanced for any informations.
Best Regards
Josue
Em 19 de abril de 2012 17:11, Josué Conti escreveu:
> Dear all,
> Please let me know if anybody have informations about a company called
> Convergia, like your products, ASR/ACD or mor
Dear all,
Please let me know if anybody have informations about a company called
Convergia, like your products, ASR/ACD or more details.
With Best Regards
Josue
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.
Hi, I am encountering problem making concurrent calls using A sangoma card, It
seems that the 2nd call get a congested or buzy,I connect via
sip-->asterisk-->dahdi attached is the PRI debug messages
-- Making new call for cref 32771> DL-DATA request> Protocol Discriminator:
Q.931 (8) len
On 04/19/2012 11:52 AM, A J Stiles wrote:
On Thursday 19 April 2012, samuel wrote:
Just in case it helps:
It turned out that from asterisk version 1.8.4 on, the g729 binaries are
different from the previous versions so it was a version mismatch between
the g729 (1.8.0_3.1.5) and asterisk (1.8.8
On Thursday 19 April 2012, samuel wrote:
> Just in case it helps:
>
> It turned out that from asterisk version 1.8.4 on, the g729 binaries are
> different from the previous versions so it was a version mismatch between
> the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
>
> Thanks to the Di
On 04/19/2012 10:57 AM, samuel wrote:
Just in case it helps:
It turned out that from asterisk version 1.8.4 on, the g729 binaries are
different from the previous versions so it was a version mismatch
between the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
Ahh, and that is 'documented'
On 04/18/2012 06:43 PM, bilal ghayyad wrote:
Dear Warren;
Yes I am compiling and installing dahdi first and then I start by asterisk
1.4.39 but I do not find chan_dahdi under /usr/lib/asterisk/modules, but if I
used asterisk 1.8, it is working fine.
From the other side: I tried asterisk 1.4.
Just in case it helps:
It turned out that from asterisk version 1.8.4 on, the g729 binaries are
different from the previous versions so it was a version mismatch between
the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
Thanks to the Digium support department that found out the issue.
Samu
UNSubscribe
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
Hi
I'm having a problem with the entirety of a call being recorded in the
following scenario
I'm using asterisk 1.8.7.0
Person A (asterisk peer) calls Person B (not on asterisk, real world
number)
Mixmonitor is invoked by Person A in the outbound context and
AUDIOHOOK_INHERIT(MixMonitor)=yes is a
Dears,
In Asterisk, how do I catch the audio stream in real-time for another
application?
The audio stream can be in the form of samples or RTP Cells , just need
real-time.
Because of that I have the applications myself that need to analysis the
audio in asterisk.
I have read the Asterisk Defin
Hi Alec and Duncan
Thank you both for taking the time to reply.
Firstly regarding Duncan's post.
The asterisk makes a call fine when the FXO card is connected to a analogue
landline and the GSM Gateway works fine when connected to standard phone
handset. The problem only arises when I connect to
--- On Wed, 4/18/12, Warren Selby wrote:
> exten => *280,n,Set(DEVICE_STATE(Custom:lamp)=BUSY)
Thanks!
So in short, it's all about DEVICE_STATE or DEVSTATE for * 1.4.
I've just one last issue and was wondering how to run the following command on
a remote Asterisk server:
Set(DEVSTATE(Custom:
can anyone help me with this case?
thank you,
Leo
Hi there,
I setup realtime asterisk 10.3.0 with backend mysql server. everything seems to
work fine except when I tried to enable the extensions for dialplan to be
obtained from mysql, I got an empty dialplan. I am not sure why this happenin
18 matches
Mail list logo
