Hi,
I'm planning to use Python to develop applications complementing Asterisk.
Can you share your experience with entries mentioned in voip-info.org (
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples) like :
py-asterisk
pyst
StarPy
Fats
Regards
--
On 08/03/2010 04:21 PM, Philipp von Klitzing wrote:
Also:
There are at least two implementations of the g726 codec, i.e. g726 and
g726aal2. For this also look at the g726nonstandard setting in sip.conf.
It is quite possible that your problem is here.
I have the following setting in
hello,
Although my previous posts in this forum have not received satisfying
answers, here is another question from me.
my question is can i use ChanIsAvail function to get the status of a user in
the format SPI/user-id if i provide user in sip uri like this
ChanIsAvail(SIP/u...@153.18.x.x:5062)
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
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On Thu, Aug 5, 2010 at 11:28 AM, Ron nha...@gmail.com wrote:
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
# Asterisk Manager API Action CoreShowChannels: List currently active
channels
Ron wrote:
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
You could get it to run a command and do 'core show hints' and parse the
result. You will need to
Dear All,
i would like to ask please if someone tried to make a codec conversion from
ilbc to g729, because i did that but the voice quality was too bad and a lot
of disconnection..
Can i get your feedback regarding this issue please?
regards
--
Hi!
Although my previous posts in this forum have not received satisfying
answers, here is another question from me.
You might want to consider to reqest a refund. ;-
my question is can i use ChanIsAvail function to get the status of a user
in the format SPI/user-id if i provide user in
Only when I configure my Grandstream to use only G726 (I have 8
choices), I see that the g726-codec is used.
When I configure 7 x g726 and 1 x alaw, then again alaw is used !
Is it normal that Asterisk has such a great preference for alaw ?! The
moment the peer suggests codec alaw (even if
Thank you. i think i would go for this solution.
On 8/5/10 4:53 PM, Gareth Blades wrote:
Ron wrote:
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
You could
On Wed, Aug 4, 2010 at 10:12 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Ok, here's the challenge:
I would like to be able to find, match - and then react - upon prompts
that are presented by the outbound/remote side of a call. Think mobile
phone and This user is
- michel freiha mich...@gmail.com wrote:
Dear All,
i would like to ask please if someone tried to make a codec conversion from
ilbc to g729, because i did that but the voice quality was too bad and a lot
of disconnection..
Can i get your feedback regarding this issue please?
On Thursday 05 August 2010 06:05:48 Ron wrote:
Thank you. i think i would go for this solution.
On 8/5/10 4:53 PM, Gareth Blades wrote:
Ron wrote:
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Subject: Re: [asterisk-users] AMI Command
Actually, what you probably want is the CoreShowChannels command.
Tilghman Lesher
To second this; core show channels doesn't
Any answers would be appreciated
Thx
UK
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Thursday, July 29, 2010 4:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
Yes. Unless you use make samples while compiling the new Asterisk, you
won't lose your confg files.
I'm afraid there's no 1.7 version of Asterisk. [?]
On Thu, Aug 5, 2010 at 1:01 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote:
Any answers would be appreciated
Thx
UK
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval
Karihaloo
Subject: Re: [asterisk-users] COnfig File question
Hi All:
If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure
all the config file
On Thu, Aug 5, 2010 at 11:14 AM, Felipe Figueiredo
felipe.figueired...@gmail.com wrote:
Yes. Unless you use make samples while compiling the new Asterisk, you
won't lose your confg files.
I'm afraid there's no 1.7 version of Asterisk. [?]
But there is a 1.7 version of AsteriskNow,
Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Subject: Re: [asterisk-users] AMI Command
Actually, what you probably want is the CoreShowChannels command.
Tilghman Lesher
To second this;
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality
Regards
On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson tnel...@rockbochs.com wrote:
- michel freiha mich...@gmail.com wrote:
Dear All,
i would like to ask please if someone tried to make a codec
- michel freiha mich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality
Regards
Again, iLBC is poor quality to begin with. You can't take a poor audio sample
and make it better by converting it to a codec with better
Hi Dan,
I commented locale.php in defines.php and it perfectly worked well.
Now i am wondering what is this invite participants for, while adding
conference. wherein it asks for first name, lastname, emailaddress
telephone number..
Let me brief you how i had done this setup. I had created a
Hi all!
Are someone using a CDR report? I have an Asterisk 1.6 running perfect but I
need a web based report of CDRs.
Nothing big, only the basic. Have anybody a how-to or a link?
Thanks in advance!!
--
Atenciosamente,
---
1.7 for ASteriskNOw
I will investigate..Thx for the ideas!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, August 05, 2010 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Is there a setting to roll over the Master.csv CDR File in
/var/log/asterisk/cdr-csv, from and ZIP the older file once its gets a certain
size?
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New to
Michel-
I tried to convert ilbc to ulaw and get the same
result...Bad Voice Quality
I think you have to be more specific when you say bad voice quality. Like
what? Worse than a cellphone call? Gaps
of audio missing? Robotic or cyborg sound? Static? A background tone or
buzzing?
iLBC
El 05/08/10 14:50, Tim Nelson escribió:
- michel freiha mich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin with. You can't take a poor audio
sample and make it better by
Manmohan wrote:
I commented locale.php in defines.php and it perfectly worked well.
Now i am wondering what is this invite participants for, while adding
conference. wherein it asks for first name, lastname, emailaddress
telephone number..
The 'Invite Others' option is mostly for installs
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in /var/log/asterisk/messages.
... ERROR[25658] codec_dahdi.c: Failed to open
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in /var/log/asterisk/messages.
Miguel-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freiha mich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin with. You can't take a poor audio
sample and make it
logrotate
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
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On Thu, Aug 5, 2010 at 4:26 PM, Ujjval Karihaloo
This just made me remember some comment on the iax.conf sample file...
disallow=lpc10; Icky sound quality... Mr. Roboto.
LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job
with pitch detection so it tends to have a
'robotic' sound. With advent of
Kevin P. Fleming wrote:
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in
On 08/06/2010 05:40 AM, Jeff Brower wrote:
Miguel-
El 05/08/10 14:50, Tim Nelson escribió:
- michel freihamich...@gmail.com wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
Regards
Again, iLBC is poor quality to begin with. You can't
Suddenly a couple days ago all of our SIP registrations are missing the
Mailbox entry. We are using MySQL Add-on for realtime.
Anyone have any idea why? Mailbox is still in the mysql tables.
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Hey, Is there any way to share MySQL connection between different
agi's.Actually when call comes to asterisk box it executes various agi scripts
sequentially. Each script checks various values by making a
new MySQL connection and then execute query and then disconnects.
So, Ideally there
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