vdadtransfer not passing calls to agent made over sip trunk

All installation and configuration problems and questions

Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N

vdadtransfer not passing calls to agent made over sip trunk

Postby electriccowboy79 » Wed Apr 09, 2008 2:13 pm

I have been trying to debug this for hours. I have setup trunks to dail out over Sangoma cards but this has me stumped. :shock:

Here is what occurs: I can log in, unpause, dialing occurs, the phone dialed rings but then the call is not transferred to agent. The call is placed over a sip trunk.

I have verified that my sip connection gets registered and verified this in the CLI.

I have read a great deal of the existing posts surrounding and about this type of bug and believe that somehow with the sip configuration, it causes the callerid to not match up the conference correctly with the call. This is what it seems like. Please help me.

-Martin
electriccowboy79
 
Posts: 2
Joined: Wed Apr 09, 2008 11:15 am

Postby mflorell » Thu Apr 10, 2008 3:58 pm

make sure that the sip-silence Playback is before your ag-VDADtransfer extensions in extensions.conf.
mflorell
Site Admin
 
Posts: 18387
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby nobesnickr » Fri Apr 11, 2008 4:52 pm

i seem to be very much in the same boat. Our system has been working flawlessly with Zap channels but we want to test out SIP VOIP to keep cost down and such.

I configured the SIP settings and dial plan changes (including the Playback(sip-silence) and the outbound route) but calls will not be passed to the agents and it is driving me crazy. My voip provider (Gafachi) see's us placing the calls and even my AMD works fine to recognize the difference between people and machines but for some reason NONE of the calls are getting passed to the agents. They are either being dispositioned as 'busy' or 'No Answer autodial'

Any suggestions would be GREATLY appreciated
nobesnickr
 
Posts: 56
Joined: Thu Nov 01, 2007 6:44 pm

Postby nobesnickr » Fri Apr 11, 2008 5:35 pm

i figure it out on mine, i didnt have the tTo and the end of my outbound call extensions
nobesnickr
 
Posts: 56
Joined: Thu Nov 01, 2007 6:44 pm

Postby abracsas » Tue May 13, 2008 5:32 am

This issue was driving me crazy too. Is there any solution ? sip-silence and so on solved for only one user, usually the first or the second one called.
Please help ! :shock:
abracsas
 
Posts: 1
Joined: Tue May 13, 2008 5:24 am


Return to Support

Who is online

Users browsing this forum: Google [Bot] and 57 guests