I have a server that had been operating for a few years now with IAX2 trunks to several other servers.Since yesterday all IAX2 trunks now say UNREACHABLE.No configuration changes have been made and no upgrades have been done.The server is running 11.1..
Hi.I am using two servers in my configuration: one for phones registration and another one as gateway, where all the providers are connected. Both are connected through an IAX trunk. I am having some trouble on matching both CDR’s, since durati..
I have multiple Asterisk servers in various parts of the world all connected using ded..
I have 5 Asterisk servers all using mysql realtime to store queue log information.There is 1 out of 5 servers which stores the data in 4 columns : data1 –> data 5.All other servers store data in 1 column data with the data seperated by pipe.I see..
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 been working with distributed device states in Asterisk using XMPP attached to an OpenFire server. I have it working well across two servers and want to roll it out across every server in my company. All servers are Asterisk 11.6.0. I am runn..
everybody,what are the current options to get an Asterisk-system high available?Using two servers as active/passive with DRBD, Pacemaker/Corosync works very good, there are no quality issues of the voice quality, even not on high loaded servers and..
Anyone know if it is possible with Asterisk 11.7 using a Tigase 5.1.5 server to stop receiving notice about old and new messages waiting? Looking at xmpp.conf there does not seems to be a setting to disable voicemail notification. Only the Device Stat..
listi have create i trunk Sip between 2 servers in the same networkwhen i call a number (inbound calls) in the first server i can forward this number to my sip 222 in the second serverexten => 0522xxxxxx,1,Dial(SIP/222@trunk_created,30)my question..
all, Is there any way of originating calls in future without using call files?We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we l..