Asterisk supports a whole bunch of codecs in the regular install -
ulaw, alaw, gsm,ilbc being the more popular ones. A common paid codec
is g729 - avbl at digium.com
-rajeev
On 1/29/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
Can you please tell me whether Asterisk requires any
All
We have a PRI line setup on an asterisk box using TE110P. Both outbound and
inbound are working fine BUT the provider claims that all our numbers come
prefixed with a '0' (in India a 0 prefix indicates long distance) and that
could become an issue with local calls.
National Numbering Plan
http://astguiclient.sourceforge.net/vicidial.html
- supports both inbound and outbound
http://queuemetrics.com/
- excellent set of metrics to measure your agents' performance!
good luck
-r
On Dec 17, 2007 8:14 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Sat, 2007-12-15 at 19:06 +0200, Dovid
PROTECTED]
wrote:
On Tuesday 18 December 2007 15:22:18 Rajeev Natarajan wrote:
We have a PRI line setup on an asterisk box using TE110P. Both outbound
and
inbound are working fine BUT the provider claims that all our numbers
come
prefixed with a '0' (in India a 0 prefix indicates long
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
What I expected that this will do is:
1. call the number using the string $numtodial2 - works OK
2. Set call limit to $maxcall and play a message $msgtime milliseconds
before
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
What I expected that this will do is:
1. call the number using the string $numtodial2 - works OK
2. Set call limit to $maxcall and play a message $msgtime milliseconds
before
Great! thanks
On Dec 3, 2007 8:31 PM, Mark Michelson [EMAIL PROTECTED] wrote:
Rajeev Natarajan wrote:
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
The problem is that you have one too many pipes ('|') in your Dial
That's what it was... I should have posted :-)
playing with /etc/mactab and nameif to fix it.
-r
On 4/7/07, Nikolai Lusan [EMAIL PROTECTED] wrote:
On Fri, 2007-03-23 at 03:11 +0530, Rajeev Natarajan wrote:
It happened again this evening and when I checked the host-id
in /var/log/asterisk
Well, we have add similar issues - do you use a media gateway /.IP Phones /
softphones as your extensions?
We were running Audiocodes and for some reason (I suspect a poor ethernet
switch), when there are more than 15 people using the line, Audiocodes will
not respond to a qualify and asterisk
Any sip debug you may have?
You might want to check your timing source. if you don't have a digium card,
to see if you have ztdummy installed correctly. Meetme requires a timing
source.
rajeev
On 3/15/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi,
i am using the n-way-call dialplan solution
and eth2 (Yes: i have three
network interfaces) interchange on reboot. Are they related?
thanks
rajeev
On 3/22/07, Rajeev Natarajan [EMAIL PROTECTED] wrote:
All,
I have around 10 opteron 165 servers all running Fedora Core 5 and
Asterisk 1.2.x (mostly Asterisk 1.2.16) with 15-25 channels of g729
All,
I have around 10 opteron 165 servers all running Fedora Core 5 and Asterisk
1.2.x (mostly Asterisk 1.2.16) with 15-25 channels of g729 each. They
register without any problem but I had to use the codec_g729.so
corresponding to the i386 version in all of them (asterisk would not start
if i
All,
Am running asterisk on an Opteron 165 with 4GB RAM and 1x80GB and 1x320GB
SATA for a call center application (running VICIDIAL). Asterisk CLI
(accessed by screen logging asterisk on startup and entering the allocated
screen) gives me 'Warning LSP Low' and the voice quality goes down when
Did some more googling and grep-ping and I found that this message most
likely comes from codec_g729a.so.
Has anybody seen this before? Anything that we should be concerned about?
Thanks
rajeev
On 3/16/07, Rajeev Natarajan [EMAIL PROTECTED] wrote:
All,
Am running asterisk on an Opteron 165
have you tried looking at the CLI to double check on the call flow? do make
sure that you 'set verbose 10' or something like that.
On 2/24/07, Jonathan Solano [EMAIL PROTECTED] wrote:
Hi all, I'm having a problem, with the h extension.
I have an application, when I call it check for the line
I use mime-construct along with the System command - works great.
On 2/26/07, Dovid B [EMAIL PROTECTED] wrote:
- Original Message -
From: Al Bochter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 26, 2007 4:20 AM
Subject: [asterisk-users] Sending Email
Am working with Arun on this project - here's a longer description of the
problem:
We've been fighting with our service provider on this issue - we seem to be
getting a BYE just after we receive an ACK. They claim that it is an
asterisk issue! The service provider provides only IP based
I think the + convention started off because different countries have
different international access codes. Well, on GSM networks, + can be a
part of the number to represent the international access code ( the
traditional access code in India is 00 for international). So to call
Digium, from my
Or you can look at PHP-AGI; use the php to query mysql (probably more scalable than dialplan MYSQL) Take a look at http://www.jivesoftware.org/ - perhaps some way you can use that?
rajeevOn 11/10/06, Andrea Spadaccini [EMAIL PROTECTED] wrote:
Ciao Ondrej, That's why I was more thinking about mysql
Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two...On 11/10/06, Jonathan Borden
[EMAIL PROTECTED] wrote:I have noticed it too and do not use them anymore..
Jon-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL
Asterisk B: Create an extension (just as you would if you want to connect a SIP client)Asterisk A: Have this guy register using the extension (just as you would using a SIP client like SJPhone) - You will probably have to use type=peer though.
Make sure you take care of NAT and stuff like that if
Just a wild idea:Store the filename in a variable before the call enters the queue - say RECFILENAME - and then once you know which agent has taken the call, execute an mv operation (using the system command) something like
system(mv ${RECFILENAME} ${RECFILENAME}-${AGENTNAME})i don't remember the
Tried them for all three - a tad pricey but good service imho.On 10/12/06, Crazy Boy [EMAIL PROTECTED]
wrote:Hi,I want to register with
http://www.inphonex.com VoIP provider. I want to configure my Trixbox and Asterisk servers with inphonex. Anybody using this service? Mainly, I want to do three
try http://astguiclient.sourceforge.netOn 10/7/06, Marnus van Niekerk
[EMAIL PROTECTED] wrote:
Yes, you can easily use asterisk for a call center, start looking here
http://www.voip-info.org/wiki/view/Asterisk+call+queues
M
Imed Imed wrote:
Hi,
I'm a novice in
yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions.On 10/5/06, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:omar parihuana wrote:
Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? )Yup. :P Thanks in
http://www.electronicscience.com/ has a good IAX2 softphone called ESC SoftphoneOn 8/16/06, David Thomas
[EMAIL PROTECTED] wrote:Sorry, poor reply.
Yes I use it on WM5, and have not seen any problems. I admit I don'tuse it a lot, but it does seem to work fine.regards,DaveOn 8/15/06, David Thomas
try www.trixbox.orgasterisk source does not come with any GUIOn 8/1/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi.
[EMAIL
. autoconfiguration for Digium
and Cisco phone hardware, an integrated
text-to-speech system) :)mea culparajeevOn 8/1/06, Alex Robar [EMAIL PROTECTED] wrote:
Trixbox is not a GUI, it's a package that includes the OS, Asterisk, a GUI, etc.
FreePBX is the GUI included in Trixbox.AlexOn 8/1/06, Rajeev Natarajan
So, this is just a wild idea. you want to send all incoming calls to a record prompt. you are probably doing something like[incoming-context]exten = s,1,SetVar(RECFILENAME=)exten = s,2,Record(${RECFILENAME})
what if you did:[incoming-context]exten =
Could you paste your dial plan please or email me off list? The variable space is unique for each channel - are you initiating another call within the macro? you can set the variable with a SetVar(_VARNAME=xxx) to ask the variable to be inherited in the sub-channels created from the initial one
Vidura,you would want to use some kind of IVR + php-agi to do the database operations (of course there are 10 other combinations - like Ruby - on -rails and RAGI). Quick suggestion: if you've played with asterisk before, I recommend that you look at
voip-info.org for php-agi links and
Sure... I would do the following 1. Set qualify =yesBash Script (in a cron) that doesa. asterisk -rx sip show peers foob. grep UNREACHABLE foo | wc -l mime-construct if output of the grep 1
hope this helpsrajeevOn 6/26/06, Roger Workman [EMAIL PROTECTED] wrote:
Is there a way to get asterisk to
http://www.voip-info.org/wiki/view/Asterisk+phonesScroll down and you will see a list of Softphones that you can choose from. best way to test it, imho, use:
1. Echo test - http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Echo2. configure another sip phone on another PC and call!
you may
This worked for me yesterday: -Please replace your actual extension number where it says extensionnumber and password in passwordOn asterisk:[extensionnumber]
If the users have a bluetooth device like a cellphone-with-bluetooth or
their laptop, this might work: http://mundy.org/blog/index.php?p=78 -
you'll have to modify the script in the tutorial a bit.
essentially - you have a presence server at the two offices - when they
enter the building, the
Have you tried using the Trunk Sequence AMP -- Setup -- Outbound Routing
Seems to work for us!
Rajeev
--
Chief Technology Officer
Gyantec Consulting (I) Pvt. Ltd.
Chennai, INDIA
Phone: +91-44-4205-4446
Mob : +91-944-407-2925
Fax : +91-44-4205-4546
VoIP : +1-360-519-5969
Dovid Bender wrote:
http://mundy.org/blog/index.php?p=93
http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki
(Chapter 4 and 7)
The above links have some excellent documentation.
www.voip-info.org specifically has some really good setup examples.
Recommend you go through those...
-R
Sohail Arham
if you are using AAH, please post extensions.conf,
extensions_additional.conf - also send us more info on your phones.
thanks
rajeev
ram wrote:
Hi
all of them thanks for the quick reply
i was tried adding 9 as well as 00
but i get number invalid if i put any of the digits
what kind of
38 matches
Mail list logo
