tonyclewis
September 1, 2016, 4:04pm
21
pjsip is part of asterisk. You would need to modify the c code in asterisk. As I already stated please open a ticket with digium on this as its part of asterisk and the code in asterisk
There is a patch for Asterisk’s dial Plan available now for testing.
https://issues.asterisk.org/jira/secure/attachment/55434/patch-for-awesome-transfers.diff
Where can i put this into the FreePBX dial Plan to test if this works? Where are the lines in FreePBX? I did not find them in extensions.conf@tonyclewis @lgaetz
lgaetz
May 2, 2017, 12:03pm
24
This dialplan was not written for FreePBX, it bears some resemblance to the sample config distributed with Asterisk:
[globals]
; General internal dialing options used in context Dial-Users.
; Only the timeout is defined here. See the Dial app documentation for
; additional options.
INTERNAL_DIAL_OPT=,30
[Hints]
; Allow dynamic hint creation for every extension.
exten = _11XX,hint,PJSIP/${EXTEN}
[Features]
; Extension to check user voicemail. We don't require the user to enter
; their pincode.
exten = 8000,1,Verbose(1, "User ${CALLERID(num)} dialed the voicemail feature.")
same = n,VoiceMailMain(${CALLERID(num)}@example,s)
same = n,Hangup()
; Exten to dial the main IVR internally.
exten = 1100,1,Verbose(1, "User ${CALLERID(num)} dialed the IVR.")
same = n,Goto(Main-IVR,2565551100,1)
show original
Questions about what it does or how it’s used should go to whomever created it.