Archives : March-2015
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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..
Helloi have the following field (text string) in a MySQL database : ${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}I read this string form the database and want to have the dialplan variables to be replaced with the correct content…
all,I use asterisk with redirect 302 moved« Transfer() cmd » I want monitor asterisk with snmp and MIBs but I don’t know how you can monitor Transfer functionWith Dial function is easy and that run but when I use Transfer, I don’t seeDialplan ex..
Can anyone please guide us if theres any way of disabling alerting/ringing in asterisk when a call is placed to any subscriber. What we want is the channel establishment as it happens during a call progress but the subscriber should not ring. Is t..
I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBXDID, the call does come into my ..