My sip provider gave me 2 numbers for the incoming call via pstn.nro1 = 12341234nro2 = 45674567I have a dialplan for each. if i put this on my dialplan:exten => s,1,Dial(SIP/1001)exten => Hangup()Works!But if i put one of them:exten => 12341234,1,Dial(SIP/1001)ex..
I am making SIP calls using SIP.js and configuring Asterisk 11.x for websockets calls under CentOS 7. On 11.23.1 and earlier, I had to patch the code to disable auto negociation due to ASTERISK-25659. Now that the bug is supposedly fixed in commit 8653da4fa228e1e289e09e5d024e11d24da87d..
HelloI keep getting the following error when trying to connect to the Asterisk server using AMI :$socket = fsockopen(tls://188.8.131.52,5039, $errno, $errstr, 5);Erorr on CLI :[Oct 26 14:38:19] ERROR: tcptls.c:609 handle_tcptls_connection: Prob..