Archives : October-2014
Does anyone know how to set the music on hold class with the Manager Interface in 1.8?Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere j..
I have a question please…i want to use Kamailio to do load balancing between multiple asterisk servers – say two servers – . I suppose that the configuration of the SIP users should be identical on the two servers but i do not know what is the b..
I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROMURI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIPservers (it doesnt recognize ..
all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF:1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMFsettings according to the incoming INVITE – RFC2833 or inba..
The Asterisk Development Team is pleased to announce the release of Asterisk 13.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releasesAsterisk 13 is the next major release series of Asteri..
how can I make asterisk do something when an outgoing call is picked up?The background is that I would like to record incoming and outgoing phone calls.In order to do this, I need to play an announcement telling the person calling or being called t..
Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration?Kind re..
I need to play some musiconhold content starting at a random duration from the start.Thanks to mode=custom option and either madplay or mpg123 programs, Icould successfully get what I was after on a Debian Wheezy system.Now I realized sox version..
HiIm using asterisk 1.8 but Im sure this applies to other versions.If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call:Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:xxxxxWh..
there,I am running Asterisk 11.9.0WANPIPE Release: 7.0.10DAHDI Version: 2.9.0 Echo Canceller: HWEClibpri version: 1.4.12When I start the ConfBridge application I get the following warning:[2014-10-24 14:36:21] WARNING[29177][C-00006934]: config_options.c:790uint_handler_..