Archives : June-2017
Ive run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each c..
I am in the process of configuring my systems to store voicemail in a mysql databse as opposed to on the filesystem, as it is now.My backup server is currently configured for db storage, while my production server is still using the filesystem dur..
list,I want to connect 2 sites both having asterisk installed (1.4 and 13.16from Ubuntu 14.04). When calling from 13.16 to 1.4 (call to echo test which should show video) I get in logs[2017-06-13 14:45:26] WARNING[17176][C-000003b0] channel.c: Una..
Yes, its been some time since I reached out to you with something good bu..
is somebody attending, that wants to share his outgoing dial rules of extension.conf, like used in typical(?) german pbx setups? * zero prefix for outside calls * zero zero or plus prefix for international calls * handle emergency callsWith ISDN, ..
I noticed that when a channel is destroyed, two different events can be raised : ChannelDestroyed and ChannelHangupRequest. These two events seem to be mutually exclusive : if I receive a ChannelHangupRequest, I will never receive a ChannelDestroy..
I was tasked to install Asterisk 13.16.0. from source on a CentOS7 platform.For that purpose, I used an unmaintened script of mine, written 10 monthes ago, and I was surprised to get segmentation violations whenever I ranasterisk -cvvvvvvv -U asterisk.Usual..
Im a faithful reader of this mailing list, for several years now.Lately, Im receiving emails asking me to re-enable my list subscription due to excessive bouncing.What does this exactly mean and why am I receiving this ?Beside re-enabling my subscripti..
With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified:On the incoming leg:From: anonymous ;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From: Anonym..
Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest 13.16.0 release), we have a problem with attended transfers to an alcatel pbx in which the call being transferred still has music on hold even after the transfer has complet..