On Mon, Nov 25, 2013 at 08:25:44AM +0100, jg wrote:
So:
A calls B
B answers
B puts A on hold
B calls C
B talks to C
B ends conversation with C
B talks to A again, regardless
I this correct? Looks like a simple Hold exercise.
Hello JG,
thanks for your reply..
not exactly, it's rather
I would start to combine audio and video sources inside a conference room.
HTH,
Ioan
On Sun, Nov 24, 2013 at 11:44 PM, Eric Cooper e...@cmu.edu wrote:
I'd like to cobble together a videophone from an analog phone,
connected to an Asterisk FXS channel, and a co-located video camera,
Ok, that's called blond transfer, though this is how some Asterisk people coined this. Should
work that way when using features.conf (DTMF).
With Hold and Transfer, I don't know. It probably depends on the phone. Which phones are
your using?
jg
--
Hi JG,
On Mon, Nov 25, 2013 at 12:06:36PM +0100, jg wrote:
Ok, that's called blond transfer, though this is how some Asterisk people
coined this. Should work that way when using features.conf (DTMF).
hmm, yes, I've seen this term before :-)
However, the problem is, I don't really want blond
in features.conf there is the option Disconnect Call that retrieves the call
from callee C.
Depending on your setup you may need to change it to something like *0. I don't like single
'*' or '#' keys as some of these combinations might trigger a reboot of some phone models.
jg
--
On Mon, Nov 25, 2013 at 08:25:44AM +0100, jg wrote:
So:
A calls B
B answers
B puts A on hold
B calls C
B talks to C
B ends conversation with C
B talks to A again, regardless
I this correct? Looks like a simple Hold exercise.
Hello JG,
thanks for your reply..
not exactly, it's rather
Hi Don,
Is B deciding C is unavailable and hanging up? In that case, rather than
hanging up, B wants to abandon the transfer and recover the original call.
Is that right?
well, yes actually.. I know this can be achieved by using hangup/cancel transfer
feature, but the request is, that when
I think I got it now:
While A is on hold,
B dials C
after a few seconds B wants to stop dialing C and hangs up,
the system calls back and B is connected again with A
Is that correct?
If (yes) {
I see a logical problem
} else {
please be more specific about the events
}
jg
--
On Mon, Nov 25, 2013 at 03:23:05PM +0100, jg wrote:
I think I got it now:
While A is on hold,
B dials C
after a few seconds B wants to stop dialing C and hangs up,
the system calls back and B is connected again with A
yes, I think this is what is required...
Is that correct?
If
I think I got it now:
While A is on hold,
B dials C
after a few seconds B wants to stop dialing C and hangs up,
the system calls back and B is connected again with A
yes, I think this is what is required...
You should know this, not think what it could be. Even if the customer doesn't know
Hello Friends:
I've just installed Asterisk 11 on my Linux (debian) server but it is not
starting up when trying with asterisk -vvc and service asterisk
start. Starting process just stop and shows: Illegal instruction as
final output.
Looking at logs I fouind at
On which kind of processor are you trying to run asterisk? Is it a real or
emulated CPU?
Leandro
2013/11/25 Daniel - Asterisk earohua...@gmail.com
Hello Friends:
I've just installed Asterisk 11 on my Linux (debian) server but it is not
starting up when trying with asterisk -vvc and
Hi Elder,
on Linode VPS, when you execute make menuselect, on Compiler Flags menu
you have to deselect BUILD NATIVE parameter. Then make, make install,
make samples, make config
Regards
El 25/11/2013 11:49, Leandro Dardini escribió:
On which kind of processor are you trying to run asterisk?
Hey all
I have been beating on this all weekend long.
Any feed back would be appreciated.
We stood up a 11.6 system. We tested everything we could think of.
We moved over to it and all seemed to be working good than a customer told
us that they were not hearing our vociemail greetings.
When
Use command line video capable SIP software (linphone / linphonec) with
source and modify it so an incoming call forwards to an outgoing
destination with video added. You might want to talk to developers of
linphone and ask them to make the change for you. Your analog phone calls
to linphonec
Hello Bakko:
Asterisk 11 is working now.
Would that selection on COMPILER FLAGS needed for al IaaS plattforms?
Thank you!
Elder D. Arohuanca
Lima - Peru
On Mon, Nov 25, 2013 at 12:07 PM, Bakko asannu...@gmail.com wrote:
Hi Elder,
on Linode VPS, when you execute make menuselect, on
Hello Leandro, I don't know exactly if it is real, but I think is emulated
since it is a cheap VPS.
Elder D. Arohuanca
On Mon, Nov 25, 2013 at 11:49 AM, Leandro Dardini ldard...@gmail.comwrote:
On which kind of processor are you trying to run asterisk? Is it a real or
emulated CPU?
Leandro
Hello Elder,
I don't remember where but I read in some place on virtualized servers
you have to deselect this parameter.
I'm using Linode too.
Regards
El 25/11/2013 13:16, Daniel - Asterisk escribió:
Hello Bakko:
Asterisk 11 is working now.
Would that selection on COMPILER FLAGS needed for
Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes
away.
I don't see this under 11.5.1
Doug
--
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On Mon, Nov 25, 2013 at 10:14:03AM -0800, Brandon B. wrote:
Use command line video capable SIP software (linphone / linphonec)
with source and modify it so an incoming call forwards to an
outgoing destination with video added. You might want to talk to
developers of linphone and ask them to
From: Doug Lytle supp...@drdos.info
Sent: Monday, November 25, 2013 2:01 PM
To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?
From: Bryant Zimmerman brya...@zktech.com
Sent: Monday, November 25, 2013 2:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?
From: Doug Lytle
On 11/24/2013 2:47 AM, Todd R. wrote:
Did you have the externalip setting in sip.conf set to the Elastic IP?
I believe I did. But I didn't really get a chance to plow into it too
much, I had a client holding me at gunpoint.
--
The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 12.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
All interested users of Asterisk are encouraged to participate in the
Asterisk 12
Bryant Zimmerman wrote:
Hey all
I believe I found the bug in Asterisk 11.xxx If someone can help me
verify it.
Actually,
I wouldn't consider it a bug. I've know for years that you need to
answer a channel before you play back audio or strange things can and
will happen.
Doug
--
Ben
On 11/26/2013 12:24 AM, Doug Lytle wrote:
Bryant Zimmerman wrote:
Hey all
I believe I found the bug in Asterisk 11.xxx If someone can help me
verify it.
Actually,
I wouldn't consider it a bug. I've know for years that you need to
answer a channel before you play back audio or strange
On Mon, Nov 25, 2013 at 7:17 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 11/26/2013 12:24 AM, Doug Lytle wrote:
Bryant Zimmerman wrote:
Hey all
I believe I found the bug in Asterisk 11.xxx If someone can help me
verify it.
Actually,
I wouldn't consider it a
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