" /> Release Call on SIP Server - Genesys CTI User Forum

Author Topic: Release Call on SIP Server  (Read 3897 times)

Offline cavagnaro

  • Administrator
  • Hero Member
  • *****
  • Posts: 7641
  • Karma: 56330
Release Call on SIP Server
« on: December 27, 2012, 06:38:15 PM »
Advertisement
Hi guys,
Any idea on how I can finish a call on IRD with End call treatment? It does work fine on Alcatel for example, but SIP Server sends it to the overflow...and as there is no such the call ends anywhere...my idea is to provide a small prompt to some customers and then release the call. I did a workaround by sending a Busy treatment but would like to handle also the call release.

Thanks!

Offline bublepaw

  • Sr. Member
  • ****
  • Posts: 283
  • Karma: 10
Re: Release Call on SIP Server
« Reply #1 on: December 27, 2012, 08:36:15 PM »
TRoute['','',RouteTypeReject,''] should do the trick. In addition this can be used to reject call (before treatment is played) with specific SIP error code by adding ExtensionAttached[‘{d}sip-status-code’,’error code’]

Offline Timur Karimov

  • Sr. Member
  • ****
  • Posts: 415
  • Karma: 2
Re: Release Call on SIP Server
« Reply #2 on: December 27, 2012, 09:34:41 PM »
Hi cav
I'm just check right now - every time then i need finish call from strategy side - i play the small announcment to caller about the end call and send call to  standard  Exit block. And as i can see in URS log - it's always rise the EventReleased.
It's so easy so i think you ask about another point. You need to continue call procesing after the caller or agent end call to futher operations with call data, right ?
Latest SIP server have such function  - check the manual for after-call-divert-destination options
HIH
WBR Timur

Offline cavagnaro

  • Administrator
  • Hero Member
  • *****
  • Posts: 7641
  • Karma: 56330
Re: Release Call on SIP Server
« Reply #3 on: December 28, 2012, 07:07:15 AM »
Nice! Thank you guys! Will check documentation

Offline mbouvet

  • Newbie
  • *
  • Posts: 5
  • Karma: 0
Re: Release Call on SIP Server
« Reply #4 on: January 04, 2013, 10:54:52 AM »
Hi Guys,

In URS, we use the bloc "Multifonction Function properties" and we first use sip option : "[b]ExtensionAttach['{d}sip-status-code','486'][/b]" and then the function [b]TRoute['','',RouteTypeReject,''].[/b].

Hope it will help you !

Matthieu