By placing OPENSIP in front of Asterisk, we can register multiple accounts, and we can successfully make call for Outgoing only. But in case of incoming it fails.
If two users are registered with asterisk or OpenSIP then the user that is registered latest is considered to be valid, and he is able to make calls, other user with earlier registration can not make call. My point here is in chain_sip.c what are variables or structure that need to maintain so that we can consider all registered users as active users. Thanks! Faheem --- On Wed, 8/5/09, D Tucny <[email protected]> wrote: From: D Tucny <[email protected]> Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Date: Wednesday, August 5, 2009, 11:06 AM 2009/8/4 Faheem <[email protected]> how to implement CLONED LINE Feature in asterisk Hey, I want to implement Clone Line feature in asterisk. I am using SPA-2100. The feature should work in this way. There are two ports in the SPA-2100 both are registered with asterisk with same username/password, and have the same (phone number) No one on the phone One phone in use Both phones in use Incoming Calls Both phones ring Phone in use receives call waiting notification, unused phone rings Both phones receive call waiting notification Outgoing Calls Both phones can call out The unused phone can call out Neither phone can call out * Inbound: - Both ports will ring. Whichever port is picked up first, will field the call. - Any additional calls that come in would give call waiting notification to the first line, and ring the second line. - Once the second line is being utilized, all incoming calls will be notifications in the form of call waiting beeps. * Outbound: - You will have the ability to dial out from port one. - You will be able to dial a different party on port two. *** Note *** - If you have an active call on port one, and pick up port two, you will NOT have the same call that is currently active on port one. The Cloned Line will share the same voice mail and will have the same telephone number as the original telephone line. - The Cloned Line is NOT a second telephone number. The telephone number that is assigned to the second phone port on the device is the same telephone number as the number assigned to phone port one. In sip.conf [line1] username=line1 secret=line1password type=friend host=dynamic context=outboundcalls mailbox=1...@default [line2] username=line2 secret=line2password type=friend host=dynamic context=outboundcalls mailbox=1...@default In extensions.conf [default] exten => 1234,1,NoOp(About to dial both phones) exten => 1234,n,Macro(stdexten,${EXTEN},SIP/line1&SIP/line2) exten => 1234,n,Hangup() or for trunk [default] exten => 1234,1,NoOp(About to dial both phones) exten => 1234,n,Gosub(stdexten(${EXTEN},SIP/line1&SIP/line2)) exten => 1234,n,Hangup() then stdexten would be default as comes in the sample configs... That should be everything you want if you configure the SPA-2100 to register line 1 with username line1 and line 2 with username line2... d -----Inline Attachment Follows----- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
