Nevermind,
I checked the code, and A* is not using the "F" option in MeetMe for
Page(), so it's not working by default.
Attached is a patch which fixes the problem for me, if anyone needs it.
Matteo
Il 11/02/2012 13:53, Matteo Fortini ha scritto:
Noone knows that? Where/w
Noone knows that? Where/whom could I ask?
Thanks
Il 10/02/2012 12:30, Matteo Fortini ha scritto:
Hi,
I'd like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a M
Hi,
I'd like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is connected
to the prev
offer some more options in
other fields.
Regarding the Playback issue, it seems that Playback into a
[ConfBridge|MeetMe] conference stutters and drops randomly. I think I'll
file a bug for that.
Thank you
Il 12/11/2010 10:23, Sebastian ha scritto:
> Hi
>
> On 11/11/2010 03:35
tream from A* becoming silent, then the new sound
from the phone comes up.
Do I have to file a bug?
Thank you,
Matteo
Il 11/11/2010 16:35, Matteo Fortini ha scritto:
> Hi,
> I dial on A* from a linphonec to a Playback() extension, then suddenly
> the sound stops after a while, without a
Hi,
I dial on A* from a linphonec to a Playback() extension, then suddenly
the sound stops after a while, without any notice.
I enabled debug both in linphone and A*, and the RTP packets are sent
from A* and received from linphone. It doesn't matter whether I choose
alaw, ulaw, gsm as codec (bes
ll is well, but may this be a bug?
Thanks,
M
Il 03/11/2010 12:48, Matteo Fortini ha scritto:
> hi all, please help... I am calling in the simplest way among two
> linphone clients connected to one asterisk server... the call ends on
> one side without any sign of problem, while on the
hi all, please help... I am calling in the simplest way among two
linphone clients connected to one asterisk server... the call ends on
one side without any sign of problem, while on the other side it stays
connected.
I checked the SIP dialogue and at some point the server sends a BYE
message t
Hi,
if I Page more than one extension, then the MeetMe conference stays up
even if all the called extensions aren't available or are hung up.
Is there a way of keeping track of how many extensions are attached to
the conference, and require a number or a particular extension to be
present?
Than
Hi,
I have the same extension registered with multiple softphones on
multiple servers, i.e.
100-lo...@hosta
100-lo...@hostb
and on both hostA and hostB I have the extension in extension.conf
exten => 100,1,Answer()
exten => 100,n,Dial(100-local)
When from softphone registered as 100-lo...@host
Hi!
thank you for your good answers.
Another related question:
I tried using Page() and it works perfectly, but I need to implement a
slightly different behavior, and I'm looking into ways of implementing it.
When a user picks up the phone and chooses to page the speakers, the
call should start
I'm building a paging system composed of roughly 10 switches in daisy
chain, with an embedded box with a speaker and a microphone for each
switch. The embedded box runs my software.
I need the system to be resilient to any network partition, so that
anyone can send announces from any mic to all
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