Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a "off hook" state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I "auto-answer" phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061014/d2b94535/attachment.pgp
The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada> Hi, > > I am looking to replace a quirk of our old PBX system functionality with > asterisk but after searching, archives, wiki, etc.. I cannot figure out > how. > > Here is what I would like to do: > > PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a > SIP ATA. When an incoming call comes in, I would like to ring both > phones, but if phoneA is answered first, I would like phoneB to be > answered as well and left in a "off hook" state so that when someone > picks up the receiver of phoneB, they can hear and participate in the > conversation between the calling party and phoneA. > > I believe I would have to put both phones in a MeetMe conference, but > how to I "auto-answer" phoneB when phoneA has answered the call? > > I suspect that this may not be possible with asterisk, but would like > confirmation of that. > > Thanks in advance. > > -m > > _______________________________________________ > --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 >
The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada> Hi, > > I am looking to replace a quirk of our old PBX system functionality with > asterisk but after searching, archives, wiki, etc.. I cannot figure out > how. > > Here is what I would like to do: > > PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a > SIP ATA. When an incoming call comes in, I would like to ring both > phones, but if phoneA is answered first, I would like phoneB to be > answered as well and left in a "off hook" state so that when someone > picks up the receiver of phoneB, they can hear and participate in the > conversation between the calling party and phoneA. > > I believe I would have to put both phones in a MeetMe conference, but > how to I "auto-answer" phoneB when phoneA has answered the call? > > I suspect that this may not be possible with asterisk, but would like > confirmation of that. > > Thanks in advance. > > -m > > _______________________________________________ > --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 >
On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote:> The quirk of your old PBX is in fact exactly what happens when you put any > two analog phones on the same line. The easiest way to duplicate this is > to connect another analog phone to your ATA. Some analog phones can > indicate when the other is on the line and can put a call on hold locally.In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital Nortel Meridian phone. The one phone has to be analogue because it interfaces with a radio broadcast phone patch. -m> > > Hi, > > > > I am looking to replace a quirk of our old PBX system functionality with > > asterisk but after searching, archives, wiki, etc.. I cannot figure out > > how. > > > > Here is what I would like to do: > > > > PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a > > SIP ATA. When an incoming call comes in, I would like to ring both > > phones, but if phoneA is answered first, I would like phoneB to be > > answered as well and left in a "off hook" state so that when someone > > picks up the receiver of phoneB, they can hear and participate in the > > conversation between the calling party and phoneA. > > > > I believe I would have to put both phones in a MeetMe conference, but > > how to I "auto-answer" phoneB when phoneA has answered the call? > > > > I suspect that this may not be possible with asterisk, but would like > > confirmation of that. > > > > Thanks in advance. > > > > -m > >-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061015/8fc12ae9/attachment.pgp
Are your sip phones capable of auto-answer? I can imagine you can terminate the incoming call into a meet-me conference (no pass code) and then trigger a script that creates a call file for each of the other participating phones. The auto-answer part seems like the sticky part. On 10/15/06, Marc Heckmann <mh@nadir.org> wrote:> > On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote: > > The quirk of your old PBX is in fact exactly what happens when you put > any > > two analog phones on the same line. The easiest way to duplicate this is > > to connect another analog phone to your ATA. Some analog phones can > > indicate when the other is on the line and can put a call on hold > locally. > > In fact no, I should have explained better, but in the old system one > phone was analogue and the other was a multi-line digital Nortel > Meridian phone. The one phone has to be analogue because it interfaces > with a radio broadcast phone patch. > > -m > > > > > > Hi, > > > > > > I am looking to replace a quirk of our old PBX system functionality > with > > > asterisk but after searching, archives, wiki, etc.. I cannot figure > out > > > how. > > > > > > Here is what I would like to do: > > > > > > PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a > > > SIP ATA. When an incoming call comes in, I would like to ring both > > > phones, but if phoneA is answered first, I would like phoneB to be > > > answered as well and left in a "off hook" state so that when someone > > > picks up the receiver of phoneB, they can hear and participate in the > > > conversation between the calling party and phoneA. > > > > > > I believe I would have to put both phones in a MeetMe conference, but > > > how to I "auto-answer" phoneB when phoneA has answered the call? > > > > > > I suspect that this may not be possible with asterisk, but would like > > > confirmation of that. > > > > > > Thanks in advance. > > > > > > -m > > > > > > _______________________________________________ > --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 > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061015/2f96a7fd/attachment-0001.htm
On 16/10/2006, at 2:32 PM, Paul Hales wrote:> We are currently writing a reception console for Asterisk - if > anyone is > interested in beta testing it, feel free to ask.If it can handle multiple Asterisk servers -- ME, ME! PICK ME! PICK ME! :) Thanks, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net ..... > > Open Source - Own It - Squiz.net ...... />
Count me in. Paul Hales wrote:>We are currently writing a reception console for Asterisk - if anyone is >interested in beta testing it, feel free to ask. > >Paul Hales > > >
Hi Yes I am interested. Regards Jon --- Paul Hales <pdhales@optusnet.com.au> wrote:> > We are currently writing a reception console for > Asterisk - if anyone is > interested in beta testing it, feel free to ask. > > Paul Hales > > -- > Paul Hales > Technical Manager > AsteriskIT > www.asteriskit.com.au > bus: 03 8320 8106 > mob: 0434 673 529 > > _______________________________________________ > --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>Jon Farmer Telford, Shropshire, UK ___________________________________________________________ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html
Sure thing, count me in Paul Hales wrote:> We are currently writing a reception console for Asterisk - if anyone is > interested in beta testing it, feel free to ask. > > Paul Hales > >
Hi, I am interested in test and work with your Reception appliation. Looking forward to your response. Thank you. Regards, Chaandra. Peter Lindquist <peter.lindquist.th@gmail.com> wrote: Sure thing, count me in Paul Hales wrote:> We are currently writing a reception console for Asterisk - if anyone is > interested in beta testing it, feel free to ask. > > Paul Hales > >_______________________________________________ --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 --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1?/min. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061016/6308db05/attachment.htm
Paul, I would love to test it out in a busy environment. I am sure I can provide quite alot of feedback from a "real receptionist" Thanks, Steve Totaro Crazy Boy wrote:> Hi, > > I am interested in test and work with your Reception appliation. > Looking forward to your response. Thank you. > > Regards, > Chaandra. > > */Peter Lindquist <peter.lindquist.th@gmail.com>/* wrote: > > Sure thing, count me in > > Paul Hales wrote: > > We are currently writing a reception console for Asterisk - if > anyone is > > interested in beta testing it, feel free to ask. > > > > Paul Hales > > > > >
Hello Paul Yes, I very interesting Viktor Tatianin vtatian@druzhba.lviv.ua -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Hales Sent: Monday, October 16, 2006 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Reception Console We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8106 mob: 0434 673 529 _______________________________________________ --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