Re: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help
nhadie, meetme requires a zaptel timing device... ztdummy is unreliable when using meetme conferencing... I suggest you spend time elsewhere in * until you get a digium tdm400 w/ or w/o any daughter modules... you just need the board for the timing device you don't actually need any modules. $195 for tdm400p + one mondule.. developers kit... daveC Nhadie wrote: hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. regards, nhadie Shane D wrote: Wouldn't you need someone besides yourself in the conference? On 1/7/08, Nhadie [EMAIL PROTECTED] wrote: Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing Macro(SIP/104-519e, user-callerid|) in new stack -- Executing NoOp(SIP/104-519e, user-callerid: device 104) in new stack -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack -- Executing GotoIf(SIP/104-519e, 0?report) in new stack -- Executing GotoIf(SIP/104-519e, 0?start) in new stack -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack -- Executing NoOp(SIP/104-519e, REALCALLERIDNUM is 104) in new stack -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack -- Executing Set(SIP/104-519e, AMPUSERCIDNAME=104) in new stack -- Executing GotoIf(SIP/104-519e, 0?report) in new stack -- Executing Set(SIP/104-519e, AMPUSERCID=104) in new stack -- Executing Set(SIP/104-519e, CALLERID(all)=104 104) in new stack -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack -- Executing NoOp(SIP/104-519e, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/104-519e, 0?continue) in new stack -- Executing Set(SIP/104-519e, __TTL=64) in new stack -- Executing GotoIf(SIP/104-519e, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing NoOp(SIP/104-519e, Using CallerID 104 104) in new stack -- Executing Set(SIP/104-519e, MEETME_ROOMNUM=6000) in new stack -- Executing GotoIf(SIP/104-519e, 0?USER) in new stack -- Executing Answer(SIP/104-519e, ) in new stack -- Executing Wait(SIP/104-519e, 1) in new stack -- Executing Set(SIP/104-519e, MEETME_OPTS=) in new stack -- Executing Goto(SIP/104-519e, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/104-519e, 6000||) in new stack ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ -- 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
Re: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help
Is this also the case with FC7? I have heard multiple times that FC7 has a different/better timing method. I wonder if this will help with ztdummy. Thanks, Steve Totaro On 1/8/08, dave cantera [EMAIL PROTECTED] wrote: nhadie, meetme requires a zaptel timing device... ztdummy is unreliable when using meetme conferencing... I suggest you spend time elsewhere in * until you get a digium tdm400 w/ or w/o any daughter modules... you just need the board for the timing device you don't actually need any modules. $195 for tdm400p + one mondule.. developers kit... daveC Nhadie wrote: hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. regards, nhadie Shane D wrote: Wouldn't you need someone besides yourself in the conference? On 1/7/08, Nhadie [EMAIL PROTECTED] wrote: Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing Macro(SIP/104-519e, user-callerid|) in new stack -- Executing NoOp(SIP/104-519e, user-callerid: device 104) in new stack -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack -- Executing GotoIf(SIP/104-519e, 0?report) in new stack -- Executing GotoIf(SIP/104-519e, 0?start) in new stack -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack -- Executing NoOp(SIP/104-519e, REALCALLERIDNUM is 104) in new stack -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack -- Executing Set(SIP/104-519e, AMPUSERCIDNAME=104) in new stack -- Executing GotoIf(SIP/104-519e, 0?report) in new stack -- Executing Set(SIP/104-519e, AMPUSERCID=104) in new stack -- Executing Set(SIP/104-519e, CALLERID(all)=104 104) in new stack -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack -- Executing NoOp(SIP/104-519e, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/104-519e, 0?continue) in new stack -- Executing Set(SIP/104-519e, __TTL=64) in new stack -- Executing GotoIf(SIP/104-519e, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing NoOp(SIP/104-519e, Using CallerID 104 104) in new stack -- Executing Set(SIP/104-519e, MEETME_ROOMNUM=6000) in new stack -- Executing GotoIf(SIP/104-519e, 0?USER) in new stack -- Executing Answer(SIP/104-519e, ) in new stack -- Executing Wait(SIP/104-519e, 1) in new stack -- Executing Set(SIP/104-519e, MEETME_OPTS=) in new stack -- Executing Goto(SIP/104-519e, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/104-519e, 6000||) in new stack ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ -- 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
[asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help
hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. regards, nhadie Shane D wrote: Wouldn't you need someone besides yourself in the conference? On 1/7/08, Nhadie [EMAIL PROTECTED] wrote: Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing Macro(SIP/104-519e, user-callerid|) in new stack -- Executing NoOp(SIP/104-519e, user-callerid: device 104) in new stack -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack -- Executing GotoIf(SIP/104-519e, 0?report) in new stack -- Executing GotoIf(SIP/104-519e, 0?start) in new stack -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack -- Executing NoOp(SIP/104-519e, REALCALLERIDNUM is 104) in new stack -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack -- Executing Set(SIP/104-519e, AMPUSERCIDNAME=104) in new stack -- Executing GotoIf(SIP/104-519e, 0?report) in new stack -- Executing Set(SIP/104-519e, AMPUSERCID=104) in new stack -- Executing Set(SIP/104-519e, CALLERID(all)=104 104) in new stack -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack -- Executing NoOp(SIP/104-519e, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/104-519e, 0?continue) in new stack -- Executing Set(SIP/104-519e, __TTL=64) in new stack -- Executing GotoIf(SIP/104-519e, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing NoOp(SIP/104-519e, Using CallerID 104 104) in new stack -- Executing Set(SIP/104-519e, MEETME_ROOMNUM=6000) in new stack -- Executing GotoIf(SIP/104-519e, 0?USER) in new stack -- Executing Answer(SIP/104-519e, ) in new stack -- Executing Wait(SIP/104-519e, 1) in new stack -- Executing Set(SIP/104-519e, MEETME_OPTS=) in new stack -- Executing Goto(SIP/104-519e, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/104-519e, 6000||) in new stack ___ --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
Re: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. Can you do a zap show channels in the Asterisk console (without the ) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh JEjcAt3QDqV3aN0rAZGNq9g= =Zqs+ -END PGP SIGNATURE- ___ --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
