I will not be in the office till Feb 24, but I will be checking my email
periodically.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
I will not be in the office from April 10 to 14. I will get back to you when
I'm back.
Joseph
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
There seems to be a problem with call involving
parties with different language. For example, if SIP user 101 (with
default language en) calls SIP user 102 (with default language fr), SIP user 102
will hear English prompt when he presses the # key for transfer even though his
default
It seems that there areoccasional problems
with files generated by soxmix utility.
The Asterisk console would show the following
message:
soxmix: Overriding output size to bytes for
compressed data
soxmix: help! internal inconsistency - data_written
12156 gsmbytecount 12155.
When trying
FYI, I just download the latest stable version from CVS and the problem is
gone.
- Original Message -
From: Joseph Shi
To: asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 9:22 PM
Subject: Problem with call hold
I got a very strange problem with call-hold function
the
answer. If somebody can shred some light on the problem, it will be very
much appreciated.
I'm running the Asterisk stable version at Dec 21,
2004.
Thanks ahead.
Joseph Shi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
Does anybody know if the voice actually gets routed
through Asterisk for callsbetween SIP devices? I just wonder if
calls between SIP devices would take up any bandwidth or CPU at the Asterisk
server. Please advise.
Thanks, Joseph
___
[EMAIL PROTECTED]
Sent: Monday, October 25, 2004 5:30 AM
Subject: RE: [Asterisk-Users] Bandwdith usage
Joseph Shi [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
Does anybody know if the voice actually gets routed through Asterisk
for
calls
Does anybody know whether Asterisk is capable of detecting fax tone (i.e.
CED tone) or answering machine (i.e. long voice) on an outbound call - using
Zaptel and/or SIP? It seems that it is only capable of detecting fax tone
on an incoming call using Zaptel only. I'm trying to determine if
Steve Underwood Wrote:
Joseph Shi wrote:
Does anyone know if there are any reseller for the book VoIP
Telephony with Asterisk in Hong Kong/Asia region? I'm interested in
purchasing the book but the shipping charge to Hong Kong is expensive.
Thanks.
Joseph
Just wait for the simplified
Does anyone know if there are any reseller for the
book "VoIP Telephony with Asterisk" in Hong Kong/Asia region? I'm
interested in purchasing the book but the shipping charge to Hong Kong is
expensive.
Thanks.
Joseph
___
Asterisk-Users mailing
Does anybody know where I can get Grandstream
BudgeTone IP phone in Hong Kong or South East Asia region? Please
advise.
Thanks.
Joseph
12 matches
Mail list logo