Matt Riddell lists at venturevoip.com writes:
On 1/09/09 5:19 PM, Glen Ganderton wrote:
app_cbmysql.c:37:1: warning: AST_MODULE redefined
command-line: warning: this is the location of the previous definition
app_cbmysql.c: In function âcheckMaxâ:
app_cbmysql.c:116: warning:
Hi,
is possible that two sip extensions: user-1 and user-2 are connected and I
want to transfer the call from user-1 to a third user user-3.
I know it is possible through feature keys mapping in features.conf, but I
want to do this through AMI or Asterisk CLI Commands?
Please suggest if possible?
Hi faheem,
You can do this:
ACTION: Redirect
Channel: Channel ID
Context: Context
Exten: Exten
Priority: Priority
Regards,
Qasim
On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem faheem2...@gmail.comwrote:
Hi,
is possible that two sip extensions: user-1 and user-2 are connected and I
want
As we plan for our 10th AstriCon, which will be in Atlanta, GA the week
commencing October 7th, we want to make sure that our conference sessions are
the best they've ever been!
That's why we need YOU to submit a speaking proposal - to share you experiences
and ideas around Asterisk!
The
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
sean
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sean darcy wrote:
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
Motif itself has no imposed limitations, but that's not to say Google
Voice doesn't.
--
Joshua
On 05/16/2013 09:41 AM, sean darcy wrote:
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
sean
More:
Two different motif sections. Two different xmpp sections.
xmpp
Thanks all for your help, in the end I was able to do something like:
Action: Originate
Channel: Local/300@from-internal/n
Application: MusicOnHold
Async: 1
As soon as this connects, the callee hears MOH. I get the channel out via
AMI events and start another call:
Action: Originate
Channel:
Hello all.
As part of a project I'm working on to migrate from Asterisk 1.0.11.1 to
the latest LTS version, I'm looking into providing a HA/failover
solution for the new Asterisk installation I'll be deploying.
It would appear my best bet would be to use the R850 Digium appliance.
Does
Rohit Mahajan wrote:
Matt Riddell lists at venturevoip.com writes:
Are you using the latest version of the app_cbmysql?
It looks like it needs to be updated for the latest version.
Alternatively it may say somewhere on their website which version of
Asterisk this works with?
I have
Hi,
I am looking for the easiest and fastest way to send or pull callerID and
extension# from asterisk to a web server for sales data lookup and
display. User would be logged in with known extension. I have been
looking at several options but was hoping someone here would have the best
one. I
On Thursday, May 16, 2013 , the Asterisk community services listed
below will be unavailable due to maintenance being performed. This
maintenance will begin at approximately 9:00 PM CDT (02:00 May 17
UTC) and should last no longer than 1 hour.
The affected services are:
*
Brian,
KDDI does provide a list of supported equipment and vendors. Specific
hardware or license based software products that quickly become cost
prohibitive.
I doubt that Asterisk will find it's way on the list any time soon. Because
KDDI follows the traditional big telco method of
Hello All;
Wanpipe is working only with sangoma cards so it does not work with digium
cards?
Also, who is better: to have echo canceler built in with the hardware or using
olsec?
Regards
Bilal
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