Archives : May-2017
Home » 2017 (paged 27)
Hello!I just implemented a jitterbuffer for pjsip in the dialplan in a SBC:[fromtrunk]exten => _[+0-9]!,1,Set(JITTERBUF..
HelloI have the following scenario:[mynicecontext]exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC)As expected, by dialing 2000, all three devices will ring. And thats fine. However, there are situations where I only want deviceA and device..
Call is not forwarded to voicemail in below dial plan, why?exten => 4,1,Dial(${FD_L1},25,trw)exten => 4,n,GotoIf($[${DIALSTATUS}=BUSY]?line2)exten => 4,n(line2),Dial(${FD_L2},20,trw)exten => 4,n,Voicemail(4)exten => 4,n,Hangup()– Called SIP/4– SIP/4-00000..
!Yesterday Deutsche Telekom had a really big problem and Asterisk couldnt connect to the remote Server (by Telekom) until today about 7:30.Well, it could happen… What I find really annoying was that I needed to restart Asterisk as Ichecked with sip..
Id like to be able to save the choices made in menuselect in a way that they can be tracked in a CM system and applied to a later release of Asterisk using an automated tool like Ansible. Whats the best way t..
I think many people here connect their mobile phone with Asterisk. Can someone suggest me an App that allow me to add a VoIP-number in the contact?With my old Samsung Galaxy S2 it was possible direct in system without additional Apps, but with the Gal..
Is it possible to set up a feature code to move both a caller and callee to a meeting room?If yes, what should I be looking at?Bonus question, is it possible to then automatically dial a 3rd person and invite them to the meeting room?The client wa..
We run Asterisk 13 using the FreePBX 13.0.190.19 distro based on CentOS-6.4.We also run HylFAX+ 5.5.3 with iaxModem 1.2.0 on the same system with AdvantFAX as the web front-end.Our two fax lines are configured as iax2 DEVICES.These components have b..
Good day, guys As the subject, Im sending this command for executing CELGenUserEvent via AGI*EXEC CELGenUserEvent DIAL_DATA,abcxyz*in asterisk cli with agi debug mode, I see that I successfully sent my param *abcxyz *to asterisk, but it removes my dou..
Ive MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension.Anything longer than 3-digits is cut off, example I dial extension 1000:[..