Howdy,Im looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing.The problem I see is that when the queue pauses an Member it doesnt jump into the dialplan to do so which means my ha..
Author : John Kiniston
Home » Articles posted by John Kiniston (paged 2)
Im attempting to find where my extra long DTMFTones are coming from.Im dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway.Im pressing 4 to select a menu and everything is fine.[Feb 12 16:58:18] DTMF[29762] channel..
Is there an equivalent to ${SIPPEER(${peer},status)} for PJSIP?The closest Ive been able to get is to use AST_SOURCERY to see if they have a contact${AST_SORCERY(res_pjsip,aor,${peer},contact) but Im not certain if Ill still have a contact entry af..
Howdy,Is there a way to use realtime with phoneprov.com and pjsip?Ive got a working pjsip realtime config currently but I have to add a phoneprov section to my pjsip.conf for each phone I want to provision.I was hoping the Sorcery page in the wiki wo..
Howdy,Im trying to re-write my voicemail check extension.I formerly used the SIPPEER function to get the mailbox for a peer with${SIPPEER(${peer},mailbox)}Is there a way to do this with PJSIP now that Ive converted over?I see a function PJSIP_ENDPO..
Howdy,Im trying to get my feet wet with pjsip using the conversion script mentioned on the Wiki on this page:https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsipIm using the copy of the script thats included with Asterisk 13/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjs..
Im trying to address a problem with users transferring to invalid destinations.In my sip peer Im setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to a context with a extension defined below to set some CDR variables before the call is transferred.[customer-forward]ex..