Re: [asterisk-users] 3-way calling for IAX channels
Hi Daniel - > How can I made a 3-way conference betwwen IAX channels? > My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The "default" conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] 3-way calling for IAX channels
Thanks for answering Noah, There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller <[EMAIL PROTECTED]> wrote: > Hi Daniel - > > > How can I made a 3-way conference betwwen IAX channels? > > My current version is: 1.4.21.1 > > Anytime you need a call with more than 2 parties, you need to use some > kind of conferencing application. The "default" conference > application for asterisk is meetme. You can use meetme with any kind > of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() > application in extensions.conf, and create your conference rooms in > meetme.conf > > > - Noah > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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
Re: [asterisk-users] 3-way calling for IAX channels
Hi Daniel - > There is no way to enable it at the softphone itself? As is the case for > hardphones like my Polycom. A phone can definitely do conference mixing. As you asked about IAX channels on the asterisk-users list, I assumed you were asking about how to do this in asterisk. My experience with IAX softphones is somewhat limited, but maybe if you indicate which phone you're using, somebody could provide you with assistance. - Noah > Daniel > On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller <[EMAIL PROTECTED]> > wrote: >> >> Hi Daniel - >> >> > How can I made a 3-way conference betwwen IAX channels? >> > My current version is: 1.4.21.1 >> >> Anytime you need a call with more than 2 parties, you need to use some >> kind of conferencing application. The "default" conference >> application for asterisk is meetme. You can use meetme with any kind >> of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() >> application in extensions.conf, and create your conference rooms in >> meetme.conf >> >> >> - Noah >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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
Re: [asterisk-users] 3-way calling for IAX channels
We are developing an softphone based on IAX client version 1.2 (my current SIP softphone has many eoors), but it doesn´t have a specific function for Conferencing (3-way calling) or to place the other party on HOLD. I´m trying to do it through the PBX because our softphone´s lack of functions. I´ll be gratefull for further comments. Thanks again, Daniel On Tue, Jul 22, 2008 at 11:49 PM, Noah Miller <[EMAIL PROTECTED]> wrote: > Hi Daniel - > > > There is no way to enable it at the softphone itself? As is the case for > > hardphones like my Polycom. > > A phone can definitely do conference mixing. As you asked about IAX > channels on the asterisk-users list, I assumed you were asking about > how to do this in asterisk. > > My experience with IAX softphones is somewhat limited, but maybe if > you indicate which phone you're using, somebody could provide you with > assistance. > > > - Noah > > > > > Daniel > > On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller <[EMAIL PROTECTED]> > > wrote: > >> > >> Hi Daniel - > >> > >> > How can I made a 3-way conference betwwen IAX channels? > >> > My current version is: 1.4.21.1 > >> > >> Anytime you need a call with more than 2 parties, you need to use some > >> kind of conferencing application. The "default" conference > >> application for asterisk is meetme. You can use meetme with any kind > >> of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() > >> application in extensions.conf, and create your conference rooms in > >> meetme.conf > >> > >> > >> - Noah > >> > >> ___ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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
Re: [asterisk-users] 3-way calling for IAX channels
Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. Noah Miller wrote: > Hi Daniel - > > >> How can I made a 3-way conference betwwen IAX channels? >> My current version is: 1.4.21.1 >> > > Anytime you need a call with more than 2 parties, you need to use some > kind of conferencing application. The "default" conference > application for asterisk is meetme. You can use meetme with any kind > of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() > application in extensions.conf, and create your conference rooms in > meetme.conf > > > - Noah > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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
Re: [asterisk-users] 3-way calling for IAX channels
2008/7/23 MFH <[EMAIL PROTECTED]>: > Noah Miller wrote: >> Hi Daniel - >> >> >>> How can I made a 3-way conference betwwen IAX channels? >>> My current version is: 1.4.21.1 >>> >> >> Anytime you need a call with more than 2 parties, you need to use some >> kind of conferencing application. The "default" conference >> application for asterisk is meetme. You can use meetme with any kind >> of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() >> application in extensions.conf, and create your conference rooms in >> meetme.conf >> > Asterisk supports conferencing without using meetme. In this case you > don't have a central dial in number but a single extension can initiate > the conference call. Generally this is done the same way as with > traditional PSTN service which is that while on a call between two > parties, flash the line, dial out to the third party then flash again > and all the parties should be connected. > I believe that response is slightly misleading - "Asterisk" does not support conferencing without using meetme, but Zaptel/DAHDI will emulate the PSTN flash/recall facility which looks a bit like a conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI channel types, the endpoint must manage the equivalent of a PSTN flash/recall conference. Anything cross-channel or otherwise more complex does indeed require app_meetme. Given that the OP was referring to IAX, I believe they will need app_meetme. Of course I could be wrong :) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] 3-way calling for IAX channels
On Wednesday 23 July 2008 12:17:26 Steve Davies wrote: > 2008/7/23 MFH <[EMAIL PROTECTED]>: > > Noah Miller wrote: > >> Hi Daniel - > >> > >>> How can I made a 3-way conference betwwen IAX channels? > >>> My current version is: 1.4.21.1 > >> > >> Anytime you need a call with more than 2 parties, you need to use some > >> kind of conferencing application. The "default" conference > >> application for asterisk is meetme. You can use meetme with any kind > >> of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() > >> application in extensions.conf, and create your conference rooms in > >> meetme.conf > > > > Asterisk supports conferencing without using meetme. In this case you > > don't have a central dial in number but a single extension can initiate > > the conference call. Generally this is done the same way as with > > traditional PSTN service which is that while on a call between two > > parties, flash the line, dial out to the third party then flash again > > and all the parties should be connected. > > I believe that response is slightly misleading - "Asterisk" does not > support conferencing without using meetme, but Zaptel/DAHDI will > emulate the PSTN flash/recall facility which looks a bit like a > conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI > channel types, the endpoint must manage the equivalent of a PSTN > flash/recall conference. > > Anything cross-channel or otherwise more complex does indeed require > app_meetme. Given that the OP was referring to IAX, I believe they > will need app_meetme. The interesting thing is that Zaptel/DAHDI is using exactly the same conferencing/audio mixing engine as app_meetme. Or more correctly, app_meetme is using the Zaptel/DAHDI engine for audio mixing. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
