Hi,
I don't use asterisk since 1.2.x version and never deployed an big project
with Asterisk, so I don't know if currently Asterisk can replace to Cisco
Unity as Voice Mail, but Cisco Unity is not only for voice mail the main
objective is to be part of all Unified Communications infrastructure.
Hi Guys:
I'm using Asterisk with Addpac 2620 as gateway, internally I'm using
Grandstream BT200, unfortunately when I called to external phones (PSTN),
and I have to choose some extensions, the Phone don't dial the extensions, I
believe that DTMF relay in ADDPAC is not working well. I'm using
Hi Guys,
I'm implementing my Asterisk step by step, so far the communications between
softphones, hardphones with Gateways, voice mail, are working fine. Rightnow
I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer
and AttendXFER, I'm reading features.conf in accordance
Thanks!!!
I forget Tt option! (too basis!!)
On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote:
Your dial string must have either the t or T option set.
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *omar parihuana
*Sent:* Friday
, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Check out voip-info.org, there are quite a few GUIS some even generate nice
graphs!
On 10/31/06, omar parihuana [EMAIL PROTECTED] wrote:
Hi Folks,
I would like to recover all information about the calls, incoming
calls, call time, call
Hi Folks,
I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format, are there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.
Thanks
Rgds.
--
Omar E.P.T
For bandwidth requeriments don't forget Layer 2 overhead. I.e
Frame-relay overhead is lower than Ethernet overhead.
Rgds.
On 10/5/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Oct 05, 2006 at 09:44:56AM +0100, Brian Candler wrote:
A 20ms packet duration means that 20ms of audio is
Unfortunately cRTP is not supported by Asterisk, and normally it is
implemented between routers that support it.
On 10/5/06, Dan Austin [EMAIL PROTECTED] wrote:
J. Oquendo wrote:
Benny Amorsen wrote:
rJ == raphael Jacquot [EMAIL PROTECTED] writes:
rJ ATM cell tax is actually 10% as there's
Hi Folks,
I'm reading about meetme feature, but in accordance to voip-info it
say: A zaptel interface must be installed for conferencing to work.
Unfortunately I don't have a Zaptel Card, I'm using Asterisk like SIP
server then I would like to implement meetme function. What can I do?
Is
