[asterisk-users] Asterisk as a SIP client, Need to auto-answer
Hi all,I want to initiate a call from the asterisk to an extension, where I will forward the asterisk side to another extension later (to the conference extension). I can initiate a call uning originate call from an extension to the desired extension, but it would need someone from the originator extension to answer the phone. How can i register an extension to asterisk where it automatically answers the phone and creates a channel where I may be able to redirect that channel later to the conference room.This is what I have done and didnt work:SIP.confregister = 7:[EMAIL PROTECTED] [7]type=friendauth=md5username=7secret=7callerid=7host=191.21.21.21reinvite=nocanreinvite=noqualify=1500nat=yes and in Extension.conf I got: exten = 7,1,Answer and when I originate a call using Manager API with these parameters: Channel: SIP/[EMAIL PROTECTED]CallerID: 7Exten: Any number I got the following error in asterisk CLI: == Manager 'manager' logged on from 191.21.21.21 -- Got SIP response 482 "Loop Detected" back from 191.21.21.21 Channel SIP/0041435215309-3c5a was never answered. == Manager 'manager' logged off from 191.21.21.21 I want to create a dump connection between a dump extension to any extension then redirect the channel from the dump extension side to the conferece. but How can i make the dump extension to auto-answer and create a channel when I Originate Call using manager API? BestEhsan Sponsored Link$420,000 Mortgage for $1,399/month - Think You Pay Too Much For Your Mortgage? Find Out!___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager API - Originate Call - Need Help
Hi all,How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number?I can originate a call from my SIP-network using this parameters in Originate call command :Channel = SIP/0041435215301Context = defaultExten = 00982166501553Priority = 1CallerID = 0041435215301this works with out any problems I initiate a call from one of my network sip clients (0041435215301) and call someone at anyside of the world, but Can I initiate a call from (00982166501553) to one of my sip users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need Help in Meetme (Conferencing)
Hi all,Suppose I have a simple conference configuration as below ---meetme.conf[general][rooms]conf = 0041435215311-and I have a dial plan like this ---extensions.confexten = 0041435215311,1,Answerexten = 0041435215311,2,Wait(1)exten = 0041435215311,3,Agi(agi://localhost/agiconference.agi)exten = 0041435215311,4,MeetMe(0041435215311|p)exten = 0041435215311,5,Playback(vm-goodbye)exten = 0041435215311,6,Hangup-How can I Call someone using Originate Call in Manager API to the conference.If this are the parameters for Manager API call , what should i set in Channel to make a call from conference to the guy who I want to invite, will this work at all?CallerID = 0041435215311Channel = ?Context = defaultExten = "Extension of the one who i want to call him to join conference"Priority = 1Best Ehsan___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users