My outgoing FXO calls are answered but have no audio in
either direction if I have callprogress=no in
chan_dahdi.conf. If I change to callprogress=yes then the
audio returns. My chan_dahdi.conf file is listed below. Can
anyone point-out why callprogress=no isn't working?
I'm
hi group ,
i am working on dahdi_monitor for measuring voice quality , so i want to
know that on which data i can tell that this PRI
lines are working properly, is there any measurement on basis of that i can
make MOS. i am working from last 2-3 days
but i only get idea about making .raw file and
It seems to be you are using Sangoma T1/E1 card with echo cancellation. If I
am not wrong there is a parameter for echo cancel in the card configuration,
try disabling that because already you have enabled echo cancel in dahdi
file.
Hope it help.:)
On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL
Hi,
The question is can you record the audio to evaluate its quality? There is
intrusive approach when you have a reference file that you can test against the
recorded audio, or non-intrusive approach, which allows you evaluate voice
quality of any call recording (no reference needed). Both
Am 04.02.2011 10:53, schrieb DHAVAL INDRODIYA:
hi group ,
i am working on dahdi_monitor for measuring voice quality , so i want
to know that on which data i can tell that this PRI
lines are working properly, is there any measurement on basis of that
i can make MOS. i am working from last 2-3
Hi Gopal,
i am using *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V* card
with tata PRI lines.
regards
dhaval
On Fri, Feb 4, 2011 at 3:23 PM, Gopalakrishnan A.N sai...@gmail.com wrote:
It seems to be you are using Sangoma T1/E1 card with echo cancellation. If
I am not wrong there
I discussed this with sangoma support in the past. Sangoma says, it
is NOT recommended to disable echo cancellation there.
Am 04.02.2011 10:53, schrieb Gopalakrishnan A.N:
It seems to be you are using Sangoma T1/E1 card with
echo cancellation. If I am not wrong
i have installed asterisk 1.8 following this doc
http://www.asterisk.org/downloads/yum
i installed the package
asterisk18-voicemail-imapstorage-1.8.2.2-1_centos5
in order to store voicemail in imap
but the application voicemail is not available when i type
core show application ?
in the
On Fri, 2011-02-04 at 15:43 +1000, Ryan Tucker wrote:
Hey Guys,
I'm after a way to monitor our sip trunks (peers) and send an email if they
go down.
I know I could use 'asterisk -rx sip show peers' in a shell script but that
seems messy,
especially since I'd like to monitor it fairly
On Fri, Feb 04, 2011 at 03:43:00PM +1000, Ryan Tucker wrote:
I'm after a way to monitor our sip trunks (peers) and send an email if
they go down. I know I could use 'asterisk -rx sip show peers' in a
shell script but that seems messy, especially since I'd like to
monitor it fairly closely (so
Posts untopped.
On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi All,
This posting regarding PRI voice optimization, on dahdi 2.1.0.4.
we have more than 4 machine running on 4 port PRI card with echo
cancellation hardware based.
i have enabled echo
Hi Users,
I have a problem with some of my mp3 files. they crash the system
(Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to
play them. Unfortunately the logs do not give me a clear fault or
cause of crash but i can clearly see that ts because of the MP3 files.
Its the way some
I am a little confused as to what the OP wants the system to do? Call the
proper agent, but when they don't answer, on the next call, it shouldn't
call the same agent? OK, but for how long? 5 minutes? Until they manually
unpause (current option as described by Kevin), 30 minutes? Should it
On Friday 04 Feb 2011, Timothy Smith wrote:
Hi Users,
I have a problem with some of my mp3 files. they crash the system
(Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to
play them.
Some distros used to use mpg321 instead of mpg123 (early versions of which
used to suffer
I found this solution...
In every line that Agent want to make an outgoing call, this call is routed
by my softswitch to Asterisk, in dialplan using func_odbc.conf I could know
if there any agent logged in this line because I have this information in my
DB. Then I set accoutncode field from CDR
On Fri, 4 Feb 2011, Timothy Smith wrote:
I have a problem with some of my mp3 files. they crash the system
(Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to
play them. Unfortunately the logs do not give me a clear fault or
cause of crash but i can clearly see that ts because
Thank you for the pointers.
I have checked my system, I seem to have the real mpg123. see below.
--
[root@ivr2 en]# mpg123
You made some mistake in program usage... let me briefly remind you:
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3
On Fri, 4 Feb 2011 10:54:56 +0330, Pezhman Lali l...@lopl.net wrote:
Meetme is a default conference application, but you can try conference or
konference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
(Putting everything back into the right order, and stripping out unnecessary
bits, for the sake of anybody searching the archives in future.)
On Friday 04 Feb 2011, Timothy Smith wrote:
On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
Try running
$ mpg123
On Fri, Feb 4, 2011 at 12:41 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi All,
This posting regarding PRI voice optimization, on dahdi 2.1.0.4.
we have more than 4 machine running on 4 port PRI card with echo
cancellation hardware based.
i have enabled echo cancel from
I am trying to use SoftHangup in my dialplan, but it's either not
working or I'm not using it correctly.
when i'm on the console, i see:
pbx1*CLI core show channels
Channel Location State Application(Data)
SIP/vgw1-00a2 2156181505@inbound:1 Up AppDial((Outgoing Line))
On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
(Putting everything back into the right order, and stripping out unnecessary
bits, for the sake of anybody searching the archives in future.)
Thanks!
On Friday 04 Feb 2011, Timothy Smith wrote:
On Fri, Feb 4,
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