Archives : March-2015
Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtpin endpoints table “direct_media..
Guys I have a 4 port PRI card that I need to setup each port in their own group.In chan_dahdi.conf I have the following which works for one portHow do I add the rest of the ports in their own groups so that I can have different signaling on each?[channels]language=enswitchtype=euroisdnpridialplan=unknownresetinterval`0echocancel=yesechotraining=yes;echocancelwhenbridged=no;rxgain=0;txgain=0callerid=asreceivedm..
All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provi..
Hello.Is there an analog option outofcall_message_context for pjsip?or: how to determine that the call is an outbound text message?Dmit..
I am in need of information about how to configure the sip.conf and extensions.conf for subscribers to support the dialog-info event package rfc 4235. I am using Asterisk 11.7.0.4 version. Also please inform if the phone must have the support for t..
Hey,I am running default Asterisk 11.16.0 on a FreeBSD-Machine. I need to register to several other SIP-Services (actually 3):short sip.confregister => XX@a register => XX@b register => XX@cIf I remember correctly this worked quite well, but I now chec..
everybody,i want to configure a sip trunk between my system which has asterisk 11.5.1and a cisco router. this is my scenario:Freepbx—–my system—–cisco-router—-Freepbxmy system acts like a router. if i set just one codec in dial-peers on ci..
Sorry for posting slightly off topic,I use an Avaya 1152a1x midspan device to deliver power some Snom phones. The Power distrubution unit seems to be identical toPowerDsine PD-6024G/AC/M.I can get some basic information via SNTP (networking setup, upti..
Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced w..
Ciao,ho recentemente acquistato un metodo molto costoso per guadagnare su Internet. Il metodo funziona davvero!!!Ora, in un giorno g..