I have a somewhat confusing use case.We use a mobile voip app and our users connect to our PBX via a public IP of our firewall which port forwards to asterisk (TLS and SRTP ports). Works fine. Sometimes however, our users are also connected to our ..
Im asking about this scenario:Asterisk(public IP)InternetRouter (public IP)SIP client (private IP and NAT)What settings in sip.conf will give this the best fighting chance of working?We already have nat=force_rpo..
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..
On Monday 06 Feb 2012, John Cahill wrote: > logger -s checksetexternip.sh: External IP address > has changed, changing /etc/asterisk/sip_general_custom.conf grep -v > externip /etc/asterisk/sip_general_custom.conf > > /etc/asterisk/sip_general_custom.conf…
One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. Ive set QOS, bandwidth management and turned off the SIP ALG via telnet but Im still having some probl..
All; I am using asterisk version 1.8 and I selected CDR mysql from the menuselect when I was doing the compilation and installation. How can I know if the unique id will be added to the cdr, and how I can know which information will be logged, also f..
my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct7 14:48:04] ERROR: netsock2.c:94 ast_sockaddr_stringify_f..
Hello! First of all, you should disable unused VoIP protocols. Than remove all guest accounts from used protocols, disable guest unauth access. Always use strong passwords for accounts, for users on your system. Passwords shouldnt be eq username. M..
we have 4 asterisk, versions are 1.4.35 1.4.36 18.104.22.168 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 22.214.171.124 asa well as 1.4.35. Calling echo test is OK on both servers, get audio and video. Calling echo t..
Hey, I have installed asterisk 1.8 on Slackware 13.1 from source and it is working well. I have 300 ip phones in a natted environment and my asterisk server has a public IP I would love to monitor my SIP activity on my VOIP Server, statistics like amo..