2012/6/26, Richard Mudgett rmudg...@digium.com:
This is the option I will try.
I'll report my findings here.
My findings, after setting layer1_presence=ignore in chan_dahdi.conf
are :
== Starting D-Channel on span 1
== Starting D-Channel on span 2
== Primary D-Channel on span 1 up
Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?
Thanks, Greetings and nice Day/Evening
Michelle Konzack
--
# Debian GNU/Linux Consultant ##
Development of Intranet and Embedded Systems with Debian GNU/Linux
Internet
This is the option I will try.
I'll report my findings here.
My findings, after setting layer1_presence=ignore in
chan_dahdi.conf
are :
== Starting D-Channel on span 1
== Starting D-Channel on span 2
== Primary D-Channel on span 1 up
== Primary D-Channel on span 2
Hi All,
We have a 1.6.2.6 Asterisk box connected to a 1.2 asterisk box, when people
dial to the conference room in 1.2 (from 1.6), of course they are prompted
for a room number and flush it by dialing # sign, the problem when they hit
#, this happened:
-- Started music on hold, class 'default', on
That's fine guys I figured it out:
under features.conf:
[featuremap]
;blindxfer = #1; Blind transfer (default is #) -- Make
sure to set the T and/or t option in the Dial() or Queue() app call!
blindxfer = *
I changed it to * and got rid of the pb
--
On 6/27/2012 3:44 PM, khalid touati wrote:
#, this happened:
-- Started music on hold, class 'default', on SIP/USPBX2-07d5
-- SIP/8425-07d4 Playing 'pbx-transfer.gsm' (language 'en')
and it gets disconnected. Anyone has a clue?
do you have # assigned in
Thank you Jeremy! I just posted the solution :)
On Wed, Jun 27, 2012 at 4:17 PM, Jeremy Kister asterisk...@jeremykister.com
wrote:
On 6/27/2012 3:44 PM, khalid touati wrote:
#, this happened:
-- Started music on hold, class 'default', on SIP/USPBX2-07d5
-- SIP/8425-07d4 Playing
On 6/23/2012 1:45 AM, Douglas Mortensen wrote:
Hi,
I currently have some systems on AsteriskNOW 1.7 have been happy with its clean simplicity
reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick
with their previous philosophy of keeping
Steve,
Thanks for the reply.
Would anyone else know if Asterisk allows use of SpanDSP's time zone
conversion?
On 27 June 2012 00:24, Steve Underwood ste...@coppice.org wrote:
On 06/26/2012 10:24 AM, David Cunningham wrote:
Hello,
Does SendFAX have the ability to put the caller ID and
