Danny Nicholas schrieb: > This is an HT-486 (HT-488, HT-496)? > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards > Sent: Friday, April 17, 2009 1:00 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] opening 2 and more channels on 1 SIP account > > On Fri, 17 Apr 2009, Tamer Higazi wrote: > > >> Sorry I write this message into the developer lailing list, I do this >> because nobody in the user list could answer me this question, due it's >> to technical. >> > > Date: Fri, 17 Apr 2009 18:51:17 +0200 > To: asterisk-users@lists.digium.com > > Date: Fri, 17 Apr 2009 19:34:46 +0200 > To: asterisk-...@lists.digium.com > > Impatient are we :) > > Yes you are entirely right! :)
> The dev list is for discussion that involves changing the C source code of > Asterisk. > > >> I have a Grandstream VoIP Device, at which a DECT base with 2 cordless >> phones are connected. If a call is placed and made through one cordless >> phone the other cordless phone appears as busy. >> >> What I want: 1. The Base station of the DECT cordless phones, is >> connected at 1 FXS Port of my Grandstream Telephone Adapter. 2. I want >> to place and receive as many calls at the same time through 1 SIP >> Account, and through this 1 FXS Port where the base station is connected >> through. >> > > Your assumption is incorrect. The question is not technical at all > (assuming I understand correctly). Connecting a cordless base that has 2 > handsets to an ATA does not magically give you 2 separate lines unless: > > ) The DECT base is a "2 line" base. If so, does it have 2 "rj-11's" or 1? > Since the other handset says "busy" this implies it is a single line DECT > base. > it has only 1 rj-11 port. Not true, with the DECT specification of ETSI I can send System messages where I can route the call from the base station to the handset direclty. DECT: http://www.etsi.org/WebSite/Technologies/DECT.aspx Standards EN-300 175 parts 1-8 (part 5) with system messages you can receive device through outgoing calling number and assign through ISDN a DDI or MSN number, like those commercial PBX. The messages can be send through a small 2 way AGI script. Now, if you guys tell me that SIP isn't able to make it, to receive more then one calls at the same time, as well as placing, even if the base station has only 1 RJ11 port, then cordless phone systems aren't suitable at all for the asterisk PBX, which is a very sad issue. But answering your question with a single DECT base system, that it is only capable to receive one call is NOT RIGHT. Commercial PBX vendors from example Auerswald, and clients (like the FritzBox 7130 from AVM) are doing making these tasks possible with the system messaging system of DECT defined at the European Telecomunication Standards Institute. I hope still, that SIP can manage more then one call through one account, if not, it is a sad thing and I have to drop the development for DECT for the PBX system at all. > ) Your ATA supports "2 lines." If so, does it have 2 "rj-11's" or 1? > > > You would need to provide a bit more information, but there is nothing in > the information you have provided to indicate that this has anything to do > with Asterisk. > > Maybe you would have more success calling the Grandstream or the > manufacturer of your DECT set. > > Thanks in advance, > ------------------------------------------------------------------------ > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
