Hi,
The company i work for has been active with asterisk for more than a year
and have done plenty of * installations .
Their VOIP team is based in delhi.
Navnit
- Original Message -
From: "Kannaiyan Natesan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, August 14, 2004 2:07
Hi,
1. When an agent is active on a call, i need the ablity for a third person
to join the conversation. Basically barge in by a supervisor, participate in
the conversation and then leave.
2. As an extension to the above, while on call, can the agent request a
conference from another agent and late
$AGI->exec('Record',"$vmfile:wav 30");
- Original Message -
From: "Tom Daly" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 28, 2004 8:05 PM
Subject: [Asterisk-Users] AGI->Exec Problem
> Hello,
> I am having some trouble with the Asterisk::AGI perl library. It seems
> t
Hi,
I have agents on IAX.
When a call gets transferred to an agent, ast_channel_supports_html(peer)
returns 0 even though the agent is on an iax client so the url does not get
sent.
If i dial the same client via Dial, the URL gets sent.
Can anybody help?
Thanx in advance
Navnit
Hi,
I am not sure whether this is the right forum but anyway am posting my woes.
I am trying to compile iaxclient on win2k.
Using cygwin,The lib compiles fine but when i try to make any client, i get
the following errors
/iaxclient_lib.c:424: undefined reference to `__beginthreadex'
eg. when i c
en
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan
Gesendet: Samstag, 19. Juni 2004 10:29
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Big problem with Flash
Let asterisk dial 7 and connect you to the door phone. You can then
che Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan
Gesendet: Samstag, 19. Juni 2004 10:01
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Big problem with Flash
small point. * will not go to priority 3 unless Zap/1/7 hangs up
You can send DTMF direct
small point. * will not go to priority 3 unless Zap/1/7 hangs up
You can send DTMF directly by the dial command by using the option D(digits)
but am not sure about flash
- Original Message -
From: Thorsten Gehrig
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004 1:04 PM
Subject: [Asteris
Sounds Interesting.
We have 3 * servers for a variety of configurations and are willing to test
the app
Thanx
Navnit
- Original Message -
From: "Kyle Hagan" <[EMAIL PROTECTED]>
To: "Asterisk" <[EMAIL PROTECTED]>
Sent: Friday, May 28, 2004 10:02 PM
Subject: [Asterisk-Users] Asterisk Recepti
> I rather think this is a soundcard issue - try a different brand in
> *both* computers or call a hardphone with your softphone.
>
> Cheers, Philipp
I think you are right because this lag is different when using different
sound cards.
Navnit
___
Aster
Hi,
IF i use a sip softphone or a iax softphone with
asterisk, i get a lag of about 1 second.
The two phones were on 2 different pc's near me.
When I speak on one, i hear it on the other after about 1 second.
I tried using iaxComm, Xten Xlite, etc.
Same.
FYI: The codec used was GSM.
Using
Hi,
How do I detect an Answering Machine in Asterisk.
I saw a post by Francois Lambert on 19 Jan. but am unable to get his email
id.
http://www.mail-archive.com/[EMAIL PROTECTED]/msg02388.html
Can somebody please help?
Thank you
Navnit
___
Asterisk-Use
Hi,
I am a newbie and am currently writing an
application for making outbound calls for a reminder service.I am using AGI for
that
Problem is for SIP calls. As soon as the call
goes through (even ringing), Asterisk says that the call is answered.
Checking CHANNEL STATUS gives me 6 ev
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