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nhadie at tbgi.net.ph
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PostPosted: Mon Jan 07, 2008 8:13 pm    Post subject: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: confere Reply with quote

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:
Quote:
Wouldn't you need someone besides yourself in the conference?

On 1/7/08, Nhadie <nhadie at tbgi.net.ph> wrote:
Quote:


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


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matt at venturevoip.com
Guest





PostPosted: Mon Jan 07, 2008 8:51 pm    Post subject: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: confere Reply with quote

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Nhadie wrote:
Quote:
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)
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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david.cantera at iacne...
Guest





PostPosted: Tue Jan 08, 2008 10:42 am    Post subject: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: confere Reply with quote

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:
Quote:
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:

Quote:
Wouldn't you need someone besides yourself in the conference?

On 1/7/08, Nhadie <nhadie at tbgi.net.ph> wrote:

Quote:
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


_______________________________________________
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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





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