Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Thursday, October 21, 2010 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to her 10-digit POTS number. If she is on the phone and her line is busy, I want Asterisk to place the caller into her Asterisk voicemail rather than hearing a busy signal. The way I have this working currently is by using Followme without a preceding Dial command. Seems that the Followme app handles the busy properly. The problem is that every call she receives is announced and requires her to press 1 to accept or 2 to reject. I suppose I could modify the Followme code, but I'd rather not. Any ideas are appreciated. Thanks. I know how this works with DAHDI/POTS; don't know what it will do dialing over SIP Exten = 1234,1,Dial(DAHDI/1/w5551212,20,KkTt) Exten = 1234,n,voicemail(1...@default) Exten = 1234,n,hangup Exten = 1234-BUSY,1,voicemail(1...@default) Exten = 1234-CONGESTION,1,voicemail(1...@default) When I dial 1234, the other side has 20 seconds (about 4 rings) to pick up. If no pickup, voicemail is called. Lines 4 and 5 might (or might not) be redundant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
Didn't work. It correctly times out after 20 seconds and continues to voicemail, but the caller still hears the remote busy signal during those 20 seconds. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 21, 2010 9:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Thursday, October 21, 2010 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to her 10-digit POTS number. If she is on the phone and her line is busy, I want Asterisk to place the caller into her Asterisk voicemail rather than hearing a busy signal. The way I have this working currently is by using Followme without a preceding Dial command. Seems that the Followme app handles the busy properly. The problem is that every call she receives is announced and requires her to press 1 to accept or 2 to reject. I suppose I could modify the Followme code, but I'd rather not. Any ideas are appreciated. Thanks. I know how this works with DAHDI/POTS; don't know what it will do dialing over SIP Exten = 1234,1,Dial(DAHDI/1/w5551212,20,KkTt) Exten = 1234,n,voicemail(1...@default) Exten = 1234,n,hangup Exten = 1234-BUSY,1,voicemail(1...@default) Exten = 1234-CONGESTION,1,voicemail(1...@default) When I dial 1234, the other side has 20 seconds (about 4 rings) to pick up. If no pickup, voicemail is called. Lines 4 and 5 might (or might not) be redundant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
This did the trick! Masks the busy signal. Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 21, 2010 1:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 Try changing KkTt to rKkTt. This should generate a phony ring until the call is picked up or stops. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
