On Thursday 01 December 2005 10:04, Greg Lim wrote:
I got a couple of answers to my post, but no-one actually answered my
question, namely:
What does AST_FLAG_DEFER_DTMF do?
Should be fairly simple. Don't deliver DTMF received from a particular
channel, when you're in a certain section (like
There was some talk a while back about making changes to the meetme application.
I would like to see something similar to app_conference where the
transcoding occurs only once per codec. This will allow meetme to
scale much better.
Is there any interest in this?
Geoff
Geoff Karl wrote:
I would like to see something similar to app_conference where the
transcoding occurs only once per codec. This will allow meetme to
scale much better.
Is there any interest in this?
Absolutely.
___
--Bandwidth and Colocation
On 12/1/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Geoff Karl wrote:
I would like to see something similar to app_conference where the
transcoding occurs only once per codec. This will allow meetme to
scale much better.
Is there any interest in this?
Absolutely.
Kevin, you know
On Dec 1, 2005, at 4:22 PM, Geoff Karl wrote:
On 12/1/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Geoff Karl wrote:
I would like to see something similar to app_conference where the
transcoding occurs only once per codec. This will allow meetme to
scale much better.
Is there any
On 12/1/05, SteveK [EMAIL PROTECTED] wrote:
On Dec 1, 2005, at 4:22 PM, Geoff Karl wrote:
On 12/1/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Geoff Karl wrote:
I would like to see something similar to app_conference where the
transcoding occurs only once per codec. This will allow
On Dec 1, 2005, at 5:08 PM, Geoff Karl wrote:
On 12/1/05, SteveK [EMAIL PROTECTED] wrote:
On Dec 1, 2005, at 4:22 PM, Geoff Karl wrote:
On 12/1/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Geoff Karl wrote:
I would like to see something similar to app_conference where the
transcoding
I also agree that the conferense handeling in meetme is god but we can se
the preformace problems.
But as its a standard part of asterisk its very nice to have it in there
and i would vote for meetme to be more optimised.
Best regards
jan
--On Thursday, December 01, 2005 02:08:16 PM -0800
Hi All,
I have a really funky problem, which I can't seem to pin point.I have a
setup that looks something like this:
SS7 Networks --SS7-- Veraz IGate4000 --SIP-- Asterisk
Now, Asterisk has a second connection, that looks like this:
Asterisk --PRI-- Avaya CTI
Now, I'll describe several
This is observed with 1.2.0, not CVS.
When a sip peer that is behind NAT is invited, it returns a contact header
with a port other than 5060 in it's 180 and 200 messages. For instance:
-- Called otherguy/04082098516
ev02a*CLI
-- SIP read from 1xx.yy.zz.aa:5060:
SIP/2.0 100 Trying
Call-ID:
Geoff Karl wrote:
I have looked at app_conference many, many times and it looks great,
but I really want to see something supported within Asterisk.
Conferencing is a killer app and Asterisk itself should have a very
good implementation.
I too have looked at app_conference but will not run
Jeremy McNamara wrote:
Geoff Karl wrote:
I have looked at app_conference many, many times and it looks great,
but I really want to see something supported within Asterisk.
Conferencing is a killer app and Asterisk itself should have a very
good implementation.
I too have looked at
Hi, Alex:
After taking your suggestion change from em to fxoks, it still did
not work, and this time even calling to normal PSTN number also failed?
Any more suggestion?
Charles
On Wed, 30 Nov 2005, Charles Huang wrote:
span=1,0,0,esf,b8zs em=1-24
and in my /etc/asterisk/zapata.conf file,
I guess, you could also take the guts of app_conference, and
transplant them into meetme (you'd still need the disclaimer), but
then you'd still have all the muxing functionality mixed into the
features..
Any particular reason why the author does not care to disclaim the code?
--
On 12/1/05, Charles Huang [EMAIL PROTECTED] wrote:
Hi, Alex:
After taking your suggestion change from em to fxoks, it still did not
work, and this time even calling to normal PSTN number also failed?
Any more suggestion?
Charles
Is this a dedicated LD trunk or a DID PRI trunk from
There's no technical reason why app_conference couldn't go inside
asterisk. The code could be disclaimed, if people were interested in
maintaining it in the core.
It _is_ maintained, as I use it all the time, but there's actually a
couple of issues which make it less than ideal at the
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