Hi Paul,
No cigar unfortunately. I also tried encoding the message as gsm, ulaw and alaw
with no success.
The ISDN interface is alaw and the SIP phones I was testing with are definately
alaw.
Not sure what to do from here. I might just need to bypass the issue using some
alternate way to put
Hi everyone,
This is my first post to the list, although I am a long term user of Asterisk.
I have recently found a problem that I just can't seem to solve.
I have a client that has an Ubuntu x64 based Asterisk server with and ISDN
Dahdi interface and about 25 SIP handsets. Everything was
Hi!
After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
finishes. On the Asterisk console, I can see that the sound file is indeed
playing, but we can't hear it. [...]
I have tried so many things that I have lost count, and I humbly ask the
collective intelligence of
Hi Alex,
I'm new to this list, but I had this problem too, and I solved it looking at
the codecs the sip handsets use, and then I converted the voice prompts to
that codec just like Philipp said..
Ondrej
On Tue, Aug 31, 2010 at 10:04 AM, Alex Ferrara a...@receptiveit.com.auwrote:
Hi
On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara a...@receptiveit.com.au wrote:
exten = 849,1,Playback(custom/ceh-meetingmsg)
exten = 849,n,Hangup
exten = 849,1,Progress()
exten = 849,n,Playback(custom/ceh-meetingmsg)
exten = 849,n,Hangup
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
Hi Paul,
I tried adding Progress() to no avail. I still get no audio and below is what
comes up in the console.
-- Accepting call from '403xx' to '0812' on channel 0/10, span 1
-- Executing [0...@isdn-incoming:1] Dial(DAHDI/10-1, SIP/812,60) in
new stack
== Using SIP RTP CoS
On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara a...@receptiveit.com.au wrote:
Hi Paul,
I tried adding Progress() to no avail. I still get no audio and below is what
comes up in the console.
Try moving Progress() before the Dial(). If you Answer() the channel,
do you have the same problem?
