'yet' but I'm sure since
the API is out it' won't take long :-)
http://www.voip-info.org/tiki-index.php?page=bounty%20skype
Wessel de Roode
Laurent
Bartek Kania a icrit :
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I just noticed that the Skype API for linux seems to be available
Message: 1
Date: Sun, 1 May 2005 19:01:24 +0200
From: Michiel van Baak [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dutch SIP or IAX numbers
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
On 12:23, Sun 01 May 05, Asterisk
de Roode
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 27-03-05
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman
:
[sales]
joinempty = yes
announce-frequency = 30
announce-holdtime = yes
member = Agent/2537
Please help :-)
Thanks in advanced,
Wessel de Roode
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.8.3 - Release Date: 25-03-05
Date: Fri, 4 Mar 2005 17:37:16 -0500
From: John Scully [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is anyone using asterisk in a small call
center
Hello - I have just joined the lists and am considering installing quite a
few * systems.
I am looking for an IP-PBX with both solid standard
For the unknown.
ZAPHFC is a driver that enables the use of a cheap ISDN card to run in TE or
NT mode.
In other words, to run like a standard ISDN terminal to receive and place
calls over a BRI line.
The driver also enables to us a hfc card in NT mode which enables it to
connect to your own
ISDN
carrier via a PRI, they will dictate what
the DID looks like. Some will be the last 4 digits, others
will be all 10. (assuming US). They do this, because it would
be to difficult to maintain your extension mapping on their side.
You purchase a DID. When a call comes in it says, This is the
Message: 1
Date: Sat, 19 Feb 2005 16:20:31 +0100 (CET)
From: Remco Barende [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dutch VOIP-PSTN provider
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
Hi,
I've found the Vertical Service Codes / vservices.inc of Julian in the cache
of google.
It's an very extended extensions include with all the *21 *67 etc services
implemented so it is stored to ODBC or if you replace it to Dbget/put etc.
I'm wondering if somebody has the macro/agi for using
Hi!,
I'm trying to set up a Queue (which works fine now :-)
Sip clients can login in to the Queue with dialing 91 on there phone.
And as soon as there are customers the Queue calls the agents back.
I would like that the queue calls the agents also if it's phone is
call-forwarded.
With agents
Just add a line to your sip.conf:
[general]
videosupport=yes
And to enable video with eyeBeam press the switchon button on the screen :-)
Wessel
-Original Message-
From: Ing. Ignacio Ortega A. [mailto:[EMAIL PROTECTED]
Sent: Friday, January 28, 2005 19:33
To: [EMAIL PROTECTED];
11 matches
Mail list logo
