search for: callcomplet

Displaying 4 results from an estimated 4 matches for "callcomplet".

2011 Feb 08
0
Asterisk CallCompletion dialplan
...wing this document https://wiki.asterisk.org/wiki/display/AST/Generic+Call+Completion+Example I am getting error non-zero error on console. I am using softphone x-lite root at tux:/etc/asterisk# asterisk -r Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [30 at from-sip:1] CallCompletionRequest("SIP/7623-00000013", "") in new stack == Spawn extension (from-sip, 30, 1) exited non-zero on 'SIP/7623-00000013' sip.conf [Mark]context=phone_callscc_agent_policy=genericcc_monitor_policy=generic ;We will accept defaults for the rest of the cc parameters...
2014 Jan 07
0
Asterisk CallCompletion issue (NO_CORE_INSTANCE)
...got the following error: "NO_CORE_INSTANCE" So, I was in shocked sometimes without making any changes on the dialplan coed problem solved. But in most of the time the problem kept persisting. Now I need some helps to solved the problem permananely? Thanks for your help. ?exten =>? 6,1,CallCompletionRequest() exten =>? 6,n,playback(beep) exten =>? 6,n,verbose(${CC_REQUEST_RESULT}) exten =>? 6,n,goto(${CC_REQUEST_RESULT},1) exten =>? SUCCESS,1,verbose(${CC_REQUEST_RESULT}) exten =>? FAIL,1,verbose(${CC_REQUEST_REASON})same =>?? n,hangup() exten =>? 7,1,CallCompletionCan...
2020 May 28
0
Notification when on the phone
...ou've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... same => n,Set(CONNECTEDLINE(name)=User is Busy) ...sort of 'Reverse Caller ID Name' to immediately change what the caller sees on their phone display. More involved is the CALLCOMPLETION function eg. for automating redials to the busy user when they hang up their call (see the ccss.conf.sample file.) Kind Regards, -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit o...
2020 May 28
2
Notification when on the phone
Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying