help how i can do that
Rajeev.
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AUTHENTICATION_7 = 707,1234
TXDTMFOPTION = 3
[VXML Params]
[IPsec Params]
[Audio Staging Params]
;
; *** TABLE DspTemplates ***
; This table contains hidden elements and will not be exposed.
; This table exists on board and will be saved during restarts
;
Rajeev
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
/g0/42121234 but somehow there's a zero that gets prefixed :(
Tried changing zapata.conf to include prilocaldialplan and so on but to no
avail!
Any help appreciated!
thanks
rajeev
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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
)
Presentation: Number not available (67) '' ]
[70 0b a1 39 37 38 39 30 39 31 30 31 31]
Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9789091011' ]
Thanks
Rajeev
On Dec 19, 2007 3:47 AM, Tilghman Lesher [EMAIL
to the macro
3. ** does not** set timelimit
How can I do both - set timelimit and pass call to the Macro - is there
something that is mutually exclusive about the two functions that it does
not let me do this?
thanks
rajeev
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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
will drop the call. Turned off qualify
(removed qualify=yes) and still keeping fingers crossed things seem fine.
Rajeev
On 3/23/07, Edoardo Serra [EMAIL PROTECTED] wrote:
Hi all,
I'm having a problem with some Asterisk servers interconnected
with
each other using IAX (I also tried
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
fine
but none of the above seem to work. The only resolution seems to be to keep
our fingers crossed while we restart the server!
Ideas / thoughts more than welcome!
thanks
rajeev
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Wildcard TDM400
any insights into this greatly appreciated!
Thanks
Rajeev
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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
Any help appreciated
Thanks!
Rajeev
On 2/20/07, Arun Kumar [EMAIL PROTECTED] wrote:
Instead of forwarding to IAX softphone if I'll play some music same thing
is happening in this case also.
On 2/20/07, Mark Phillips [EMAIL PROTECTED] wrote:
Without
GSM phone, I can use 0018775468963 or +18775468963 and
Allison will answer :)
Rajeev
On 12/22/06, Doug Crompton [EMAIL PROTECTED] wrote:
Question... What is the purpose of the + before the number? Does anyone
actually have to enter it? If so how would you do it? It is not used in
the US but do
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 =
rajeev[EMAIL PROTECTED]On 7/19/06, Don
[EMAIL PROTECTED] wrote:
Anyone know a reason why when you jump to a macro
from the dial command the uniqueid of the call changes? Or what a workaround to
that would be? I have tried gettinbg the uniqueid from my AGI script while in
the macro
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]
, the bluetooth device registers with the presence
server and the corresponding phone comes alive.
works great for us (as long as the bloke doesn't leave the cellphone at
home)
rajeev
--
Chief Technology Officer
Gyantec Consulting (I) Pvt. Ltd.
Chennai, INDIA
Phone: +91-44-4205-4446
Mob : +91-944
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
OK... so what you're saying is that I put a diode across the power supply input legs for the DPDT
relay, right?
(sorry, i'm not the best person at electronics...)
Greg Hill wrote:
On Sun, 10 Oct 2004, Rajeev Sharma wrote:
Yeah, thanks, I was thinking of doing something similar to that.
Actually
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