Got a strange situation[ext-queues]… exten => h,2,ExecIf($[${CALLERID(num)} = ]?Set(var29=${SHELL(curl -XPOST –header Content-Type: application/json –header Accept:application/json -d {\Phone\: ${FROMEXTEN}, \Source\: \asterisk\} http://sIte.com:80/api/v1/calls?apiKey=UABVAEI&clientId=3)}))ex..
Author : Антон Сацкий
Home » Articles posted by Антон Сацкий
ALLgotsmall questioni use call-limit=1 on peersbutcall limit is not workingifuseris notregistered on PBX and making callsso the main question is — how to Disallow CALLS without register..
my CLI full of WebSocket connection from XXXXXXXXXforcefully closed due to fatal write error What sh..
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..
need your advice i dont understand why reply ACK goes to diferrent ip (192.168.88.32)SCREEN belowhttp://tinypic.com/view.php?pic=s6m7me&s=9#.VzsVhvl96IkT..
need your help i have call in queue it shows that it was answered by 4003============================[root@asterisk ~]# grep –color 1456128646.157422/var/log/asterisk/queue_log-201602281456128688|1456128646.157422|800|NONE|ENTERQUEUE||0967145750|21456128717|1456128646.157422|800|SIP/4003|CONNECT|29|1456128688.157426|281456128817|1456128646.157422|800|SIP/4003|COMPLETECALLER|29|100|2============================..
can U help mecaller id in USTM if now working– Starting switch on 4211@4211-1 to 4203– Executing [4203@office:1] DumpChan(USTM/4211@4211-0x7f7ba4228fd0,) in new stackDumping Info For Channel: USTM/4211@4211-0x7f7ba4228fd0:================================================================================Info:Na..
CAN U HELP MEIf there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidi..
facing problem withoriginatingwebrtc calls1-when iamdoing call from webrtc iget ice workingINVITE sip:0669197533@77.91.132.9 SIP/2.0Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315Max-Forwards: 69To: From: Anton ;tag=5i21qaop43Call-ID: ocq4hu8eol3kijsgv..
*friends help me **cant get incoming calls in asterisk**(when i connect **80081 in softphone —every thing is ok**)****INVITE sip:80081@10.47.10.10:5060 SIP/2.0**Record-Route: **Via: SIP/2.0/UDP 200.152.125.221;branch=z9hG4bKd4fd.b3489837.0**Via: SIP/2.0/..