Hi,
On Mon, Feb 11, 2013 at 03:38:09PM +0100, Shitian Long wrote:
I am really a beginner of PRI ISDN board, I am wondering if there is a quick
start chan_dahdi.conf configuration I could use.
For starters, there's the example / reference chan_dahdi.conf:
On 02/12/2013 06:48 PM, sean darcy wrote:
On 02/12/2013 05:37 PM, Rusty Newton wrote:
Original Message -
From: sean darcy seandar...@gmail.com
Can I throw A and B into a confbridge and then add C? Create a new
channel that grabs the A - B channel? Or is there a more straight
I would like to configure Asterisk send back only a Trying or Progress message
to the SIP client and not any early audio for ringback. I've confirmed
Asterisk is sending RTP when the call is ringing by using rtp debug on Asterisk.
Does anyone have any ideas on how to accomplish this?
I've
On Tue, Feb 12, 2013 at 7:43 AM, Olivier oza_4...@yahoo.fr wrote:
Using the commands bellow, I could install in /usr/local/sbin
./configure --prefix=/usr/local
make
make install
Thanks for sharing this.
Just for reference, this is standard among tools compiled using ./configure
; make ;
On 02/13/2013 09:39 AM, Matthew Jordan wrote:
On 02/12/2013 06:48 PM, sean darcy wrote:
On 02/12/2013 05:37 PM, Rusty Newton wrote:
Original Message -
From: sean darcy seandar...@gmail.com
Can I throw A and B into a confbridge and then add C? Create a new
channel that grabs the
sean darcy wrote:
I had motif working two days ago but now:
Executing [1171@internal:1] Dial(DAHDI/1-1, Motif/1171) in new stack
[Feb 12 20:56:18] ERROR[7794][C-0001]: chan_motif.c:1762
jingle_request: Unable to determine endpoint name and target.
You haven't specified anything to call.
very polite *bump*
this is a real issue for us - anyone got _any_ clues or ideas ?
Thanks ;)
On 12 February 2013 14:29, Julian Lyndon-Smith aster...@dotr.com wrote:
Ever since we upgraded to asterisk 11 we have had audio problems with
our cisco 7940 phones.
The problems manifest
