[asterisk-users] dahdi-dahdi native bridging and audio level

2014-05-27 Thread Dmitry Melekhov
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 ;

Re: [asterisk-users] dahdi hungup after each ring

2014-05-27 Thread Richard Mudgett
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

Re: [asterisk-users] dahdi hungup after each ring

2014-05-27 Thread Shaun Ruffell
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

Re: [asterisk-users] transmit_silence not properly recognized on 1.8 ?

2014-05-27 Thread Matthew Jordan
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

Re: [asterisk-users] BLF and notifyringing in Asterisk 11

2014-05-27 Thread Matthew Jordan
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

Re: [asterisk-users] BLF and notifyringing in Asterisk 11

2014-05-27 Thread Kevin Larsen
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

[asterisk-users] Figuring out gateway that degrades call quality

2014-05-27 Thread Sevana Oy
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 -- _ --

Re: [asterisk-users] Figuring out gateway that degrades call quality

2014-05-27 Thread Josh Metzger
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

[asterisk-users] dialplan changes in middle of call

2014-05-27 Thread Henry Fernandes
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

Re: [asterisk-users] dialplan changes in middle of call

2014-05-27 Thread Joshua Colp
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