I am having a few problems with my queue. I am using the AgentCallbackLogin
feature. When the call comes to the
user, it does not “announce” the call to the agent. It waits until you enter the “#”.
After you hit #. It will
play the queue-support announcement to the agent and tell them t
I’m not sure about the G711 codec on
the ATA, but I know you need to purchase the g729 from digium.
http://www.digium.com/index.php?menu=asterisk_g729
pretty inexpensive at $10 each.
That’s for “concurrent” connections to the server.
Tim.
-Original Message--
You've got a 50/50 shot.
Try the crossover.
http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note0
9186a00800a3f09.shtml#topic2
It would be more helpful for you to send your /etc/zaptel.conf file and
/etc/asterisk/Zapata.conf file.
You should have something like the following
]/${EXTEN})
exten => _1NXXNXX,306,Dial(IAX2/4th_account_name:[EMAIL PROTECTED]/${EXTEN})
etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Franceen Thompson
Sent: Saturday, November 06, 2004 8:23 PM
To:
[EMAIL PROTECTED]
Subject: [As
I am wondering if there is a way to create a SIP/IAX group
of outgoing lines like Zap groups.
I am currently using the following method, but would like to
use features such as “g2” that would list all the accounts for a
SIP or IAX connection.
exten => _1NXXNXX,1,Dial(SIP/account
I am wondering if anyone is using this combination and has
experienced echo from VOIP clients (IAX & SIP) to a Zap device.
I have zero echo using Zap FXS to
Zap FXO.
I have zero echo from VOIP (SIP or
IAX) to SIP or IAX Telco provider.
I have attempted without any success pla
