Bartosz Wegrzyn - asterisk
2006-May-15 19:17 UTC
[Asterisk-Users] Asterisk as a bridge between voip clients and POTS confrence bridge
Hello, I am thinking about this, ----POTS--CONFERENCE-BRIDGE | | | PSTN | | ASTERISK | INTERNET | | VOIP USERS Users registers with asterisk, they join the confrence and later (or maybe at the begining) asterisk automatically (or maybe manually) calls the POTS conference bridge using the PSTN network. This would allow all VOIP users to interact with users on the pstn conference side. Any ideas how this could be done if possible. Thanks
Bartosz Wegrzyn - asterisk
2006-May-15 20:35 UTC
[Asterisk-Users] Asterisk as a bridge between voip clients and POTS confrence bridge
I was able to create simple solution VOIP users call exten 500 which is [meetme] exten => 500,1,Playback,thereare exten => 500,2,MeetmeCount,500 exten => 500,3,Playback,callersin exten => 500,4,Meetme,500|pMs|1234 exten => 500,5,Playback,goodbye exten => 500,6,Hangup later somebody calls extension 501 which moves 1-test to /var/spool/asterisk/outgoing/ 1-test looks like this Channel: Sip/number@context (you put whatever you want) Callerid: 1 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: common Extension: 500 Priority: 1 exten => 501,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/ ) exten => 501,2,Hangup The problem with this solution is that the 501 needs to be dialed separately. Any ideas how to enable 501 in conference call. Thx> Hello, > > I am thinking about this, > > ----POTS--CONFERENCE-BRIDGE > | > | > | > PSTN > | > | > ASTERISK > | > INTERNET > | > | > VOIP USERS > > Users registers with asterisk, they join the confrence and later (or maybe > at the begining) asterisk automatically (or maybe manually) calls the POTS > conference bridge using the PSTN network. > This would allow all VOIP users to interact with users on the pstn > conference side. Any ideas how this could be done if possible. > > Thanks > > > > > _______________________________________________ > --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 >
Mojo with Horan & Company, LLC
2006-May-17 15:32 UTC
[Asterisk-Users] Asterisk as a bridge between voip clients and POTS confrence bridge
The other problem with this: Bartosz Wegrzyn - asterisk wrote:> exten => 501,1,System(/bin/cp /etc/asterisk/1-test > /var/spool/asterisk/outgoing/is that the call file must be moved to the outgoing directory with the mv command instead of copied with the cp command, or you run the risk of asterisk grabbing the call file and trying to act on it before it contains all the data you intend to be there.
Bartosz Wegrzyn - asterisk
2006-May-18 16:49 UTC
[Asterisk-Users] Asterisk as a bridge between voip clients and POTS confrence bridge
anyone whos the answer? thx> I was able to create simple solution > VOIP users call exten 500 which is > [meetme] > exten => 500,1,Playback,thereare > exten => 500,2,MeetmeCount,500 > exten => 500,3,Playback,callersin > exten => 500,4,Meetme,500|pMs|1234 > exten => 500,5,Playback,goodbye > exten => 500,6,Hangup > > later somebody calls extension 501 which moves 1-test > to /var/spool/asterisk/outgoing/ > > 1-test looks like this > Channel: Sip/number@context (you put whatever you want) > Callerid: 1 > MaxRetries: 1 > RetryTime: 60 > WaitTime: 30 > Context: common > Extension: 500 > Priority: 1 > > exten => 501,1,System(/bin/cp /etc/asterisk/1-test > /var/spool/asterisk/outgoing/ > ) > exten => 501,2,Hangup > > The problem with this solution is that the 501 needs to be dialed > separately. Any ideas how to enable 501 in conference call. > > Thx > > > >> Hello, >> >> I am thinking about this, >> >> ----POTS--CONFERENCE-BRIDGE >> | >> | >> | >> PSTN >> | >> | >> ASTERISK >> | >> INTERNET >> | >> | >> VOIP USERS >> >> Users registers with asterisk, they join the confrence and later (or >> maybe >> at the begining) asterisk automatically (or maybe manually) calls the >> POTS >> conference bridge using the PSTN network. >> This would allow all VOIP users to interact with users on the pstn >> conference side. Any ideas how this could be done if possible. >> >> Thanks >> >> >> >> >> _______________________________________________ >> --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 >> > > _______________________________________________ > --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 >
HaoXu
2006-May-18 18:07 UTC
[Asterisk-Users] Asterisk as a bridge between voip clients and POTS confrence bridge
HI Bartosz, Such solution is very strange. If you want a voip conference, Asterisk can do. If you want some pstn user to join the conference , just use some voip termination to make some invite calls. If you want to use the pots conference bridge with voip , only fxo gateway need . it is the same to a normal voip system while the pstn gateway is the conference bridge. Hawk -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bartosz Wegrzyn - asterisk Sent: Friday, May 19, 2006 7:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk as a bridge between voip clients and POTS confrence bridge anyone whos the answer? thx> I was able to create simple solution > VOIP users call exten 500 which is > [meetme] > exten => 500,1,Playback,thereare > exten => 500,2,MeetmeCount,500 > exten => 500,3,Playback,callersin > exten => 500,4,Meetme,500|pMs|1234 > exten => 500,5,Playback,goodbye > exten => 500,6,Hangup > > later somebody calls extension 501 which moves 1-test to > /var/spool/asterisk/outgoing/ > > 1-test looks like this > Channel: Sip/number@context (you put whatever you want) > Callerid: 1 > MaxRetries: 1 > RetryTime: 60 > WaitTime: 30 > Context: common > Extension: 500 > Priority: 1 > > exten => 501,1,System(/bin/cp /etc/asterisk/1-test > /var/spool/asterisk/outgoing/ > ) > exten => 501,2,Hangup > > The problem with this solution is that the 501 needs to be dialed > separately. Any ideas how to enable 501 in conference call. > > Thx > > > >> Hello, >> >> I am thinking about this, >> >> ----POTS--CONFERENCE-BRIDGE >> | >> | >> | >> PSTN >> | >> | >> ASTERISK >> | >> INTERNET >> | >> | >> VOIP USERS >> >> Users registers with asterisk, they join the confrence and later (or >> maybe at the begining) asterisk automatically (or maybe manually) >> calls the POTS conference bridge using the PSTN network. >> This would allow all VOIP users to interact with users on the pstn >> conference side. Any ideas how this could be done if possible. >> >> Thanks >> >> >> >> >> _______________________________________________ >> --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 >> > > _______________________________________________ > --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 >_______________________________________________ --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