I have two asterisk boxes connected via SIP protocol. I want to deploy SIP peer authentication to this connection. What is the needed configuration? I have the following configuration but changing username and secret does not affect the connection..
Kia ora,As many of you are aware (and for those who arent) as part of Asterisk13.8 changes went into the res_odbc module to more heavily leverage UnixODBC connection management and pooling capabilities. Previously we would use only a single connect..
allcan anybody help me there to search a problem from time to time Connection closed before receiving a handshake responseWebSocket connection to wss://XXXXXXXXXXX:8089/ws failed: Connection closed before receiving a handshake response sipml.js?14636613801642821:16..
everyone,I upgraded from Asterisk 13.5.0 to 13.7.0 and I am having database connection problems. I am doing Asterisk realtime with PJSIP 2.4.5 and it works perfectly in 13.5.0. But now I am losing my database connection (running on a virtual box) ..
everybodyi want to have sip connection between two asterisk systems (145 and146). connection from 145 to 146 is ok but i can not call from 146 to145. this is h323.conf file in 145:[peer146]host184.108.40.206type=friend context=from-trunk[to-146]type=p..
I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210with 2 port E1.now i bought another card Diguim TE410 and I want to add itthe current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from ..
What are best practices for allowing connection by remote SIPextensions over the internet? Im thinking of putting the SIP inside a VPN connection.Kind Rega..
I am currently trying to set up CDR logging. I got all the ODBC stuff for my mysql server set up, also checked theres a connection using odbc showin the Asterisk console:Name: asteriskI also created a table for the CDR logs:This table is also referen..
I have a process that runs on a server and does a simple asterisk -rx sup show peers > /tmp/peers and then looks for any (Unspecified) items and reports them as having lost connection. My server is running 1.4.43 and the two boxes I am monitoring ..
I am attempting to get an asterisk server to step out of the media path, but am running into a brick wall. Can someone assist? Heres my setup.. Ultimate SIP Provider —> LCR Trunk(Asterisk 1.6) —-> PBX (Asterisk 1.8). I am attempting to get the tr..