Hello!
I use asterisk with TE420 as PRI switch for two channels :
;panasonic uplink
group=3
context=panasuplink
;relaxdtmf=yes
;immediate=yes
rxgain=0.0
txgain=0.0
mohsuggest=default
jbenable = no
;jbenable = yes
;
On Mon, May 26, 2014 at 3:12 PM, Bart Remmerie remme...@gmail.com wrote:
Hi,
I guess something's wrong with my chan_dahdi configuration, ... but I
can't seem to get it.
When I test incoming calls on a DAHDI-channel (incoming from pstn),
asterisk seems to interpret it as a caller hangup
On Mon, May 26, 2014 at 10:12:17PM +0200, Bart Remmerie wrote:
Hi,
I guess something's wrong with my chan_dahdi configuration, ... but I can't
seem to get it.
When I test incoming calls on a DAHDI-channel (incoming from pstn),
asterisk seems to interpret it as a caller hangup after each
On Sat, May 24, 2014 at 3:03 PM, Maximilian Grobecker
m.grobec...@portunity.de wrote:
Hello,
I've got a problem at the moment, that setting transmit_silence = yes
seems to have no effect on Asterisk 1.8-Certified.
Although it's enabled and core show settings confirms, that it is
really
On Fri, May 23, 2014 at 4:51 PM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
I am trying to get something working that is just not doing quite what I
want. It may not be possible, but I figured it was worth asking about.
The details:
Asterisk 11.6.0
Polycom SoundPoint IP650 phones
Unfortunately, notifyringing is only set in the [general] section in
sip.conf. It does not have a peer level override.
It would be nice if it was set on a peer by peer basis - that would be
a useful improvement.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive
Hi,
How do you figure out if one of gateways in your network leads to voice
quality loss f.e. due to transcoding? The point is that all VoIP metrics in
this case remain the same
Thanks!
Sevana
http://www.sevana.fi
--
_
--
On Tue, May 27, 2014 at 12:31 PM, Sevana Oy sa...@sevana.fi wrote:
Hi,
How do you figure out if one of gateways in your network leads to voice
quality loss f.e. due to transcoding? The point is that all VoIP metrics in
this case remain the same
Thanks!
Sevana
http://www.sevana.fi
Recently, I made a change to our dialplan and reloaded Asterisk. To my
surprise, the dialplan was reloaded for calls in progress. This caused a
problem because some of the dialplan changes affected some loops and this
caused an infinite loop.
Is there a way to change this so that reloading
Henry Fernandes wrote:
Kia ora,
Recently, I made a change to our dialplan and reloaded Asterisk. To my
surprise, the dialplan was reloaded for calls in progress. This caused a
problem because some of the dialplan changes affected some loops and
this caused an infinite loop.
Is there a way to
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