Dave Morrow
2005-Nov-02 05:04 UTC
[Asterisk-Users] Options for 3-way or Conference Calling
Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way calling or conference calling. My goal is to keep this as simple for my users as it was with our legacy PBX. On our old phone system, a user could simply, during a call, press a Conference button on their phone to bring in a third party to a call. Can this be accomplished with Asterisk? My phones are all SIP devices (Cisco and Sipura). David A. Morrow Technical Systems Lead Autodata Solutions Company David.Morrow@Autodata.Net http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 < Poor planning on your part does not necessarily constitute an emergency on my part! > This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at Administrator@autodata.net <mailto:Administrator@autodata.net> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051102/fef4d3a2/attachment.htm
Yes. I believe the Cisco phones do conferencing in the same fashion. I'm not 100% on whether or not the SPA-841 or the new SPA-941 does it. On 11/2/05, Dave Morrow <david.morrow@autodata.net> wrote:> > Hi all, I wonder if someone could lend a little insight into the best way > to configure either 3-way calling or conference calling. My goal is to keep > this as simple for my users as it was with our legacy PBX. On our old phone > system, a user could simply, during a call, press a Conference button on > their phone to bring in a third party to a call. Can this be accomplished > with Asterisk? My phones are all SIP devices (Cisco and Sipura). > > David A. Morrow > Technical Systems Lead > Autodata Solutions Company > *David.Morrow@Autodata.Net* <David.Morrow@Autodata.Net> > *http://www.autodata.net* <http://www.autodata.net/> > Tel: (519) 951-6079 > Fax: (519) 451-6615 > > < Poor planning on your part does not necessarily constitute an emergency > on my part! > > > This message has originated from Autodata Solutions. The attached material > is the Confidential and Proprietary Information of Autodata Solutions. This > email and any files transmitted with it are confidential and intended solely > for the use of the individual or entity to whom they are addressed. If you > have received this email in error please delete this message and notify the > Autodata system administrator at* **Administrator@autodata.net <**** > mailto:Administrator@autodata.net* <Administrator@autodata.net>*>* > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com<http://easynews.com/>-- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > 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/20051102/08a813f8/attachment.htm
Chris Shucksmith
2005-Nov-02 06:38 UTC
[Asterisk-Users] Asterisk as an internal pbs for a samall company
Thankyou, this was a great primer for me also. Chris trixter aka Bret McDanel wrote:>On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote: > > >>Well, >> >>U right, many missing informations. >> >>The case is quite simple(I guess), we have dids, and each call to these dids >>has to be routed to the right handset thru Asterisk, no Ivr at this time, at >>least an answering machine in case of busy or not available users. >>For the rest, we need to be able to have external calls to pstn, or even to >>other sip phones form other providers. >>Is that enough? >> >> > >Not for 100% setup, but enoughto at least get you started. From what I >understand this is what it appears you want (I may be wrong, if I am let >me know). > >You will want voicemail for each user. This is configured in >voicemail.conf >http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf > >You will need to edit sip.conf for the voip provider (register and >context) and if the office workers use sip to asterisk one for each of >them as well. >http://www.voip-info.org/wiki-Asterisk+config+sip.conf > >Lastly you will want to create a dialplan so that when a call comes in >from the DID it will then dial the appropriate user and if busy/no >answer goto voicemail. This is done from extensions.conf. >http://www.voip-info.org/wiki-Asterisk+config+extensions.conf > >You may want a macro like: >[macro-dialvmb] >exten => s,1,Dial(${ARG1},20,t) >exten => s,2,Voicemail(u${ARG2}) >exten => s,3,Hangup >exten => s,102,Voicemail(b${ARG2}) >exten => s,103,Hangup > >Then for each inbound DID something like: >exten => 18005551212,1,Macro(dialvmb,SIP/user1,1234) > >where user1 is the user defined in sip.conf, 1234 is the voicemail >extension defined in voicemail.conf and 18005551212 is the extension >that a given did goes to (ie last part of the register line). > >Hope this helps > > > >------------------------------------------------------------------------ > >_______________________________________________ >--Bandwidth and Colocation sponsored by Easynews.com -- > >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >