If I issue a dialplan reload and some AGI starts as its reloading and directs something into the diaplan that is still reloadingwhat happens????I presume my context is not there?What I see is the diaplan is messed up somehow and I goto the default cont..
I am using two polycom phones to call into an asterisk box and the console/dsp. First phone calls in and I get connected just fine.second phone calls in and I detect the Console/dsp is busy, and i try to use playtones(busy) and I hear nothing. (see below)..
Lets say I have a SIP client that supports both G711 and G729 codecs and Ihave them both enabled in sip.conf and G729 has higher priority.Can I force the call to choose a different codec based on the dialed number or other conditions?For instance I wo..
Sounds like you need disconnect supervision enabled somewhere.—–Original ..
I installed Asterisk 11 via the following command*> svn co http://svn.asterisk.org/svn/asterisk/branches/11*(as written in asteriskdocs.org http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Installing_id240883.html)But it seems that I h..
I am configuring my asterisk server as below scenarioJisi—-> Astrisk—–>Analog PBX—–> PhonesFor that I have Asterisk Server 1.8.1 in my PC Digium card in my PCTDM2400P with 5 Red (FXO) moduleI install dahdi and modprobe in my system. After t..
The service provider wants me to setup dtmfmode to rfc2833 and dtmf payload to 101.I can configure SIP trunk as dtmfmode=rfc2833 but how to configur..