[asterisk-users] conferencing help

[Paul Hales]

Then it's time to build zaptel, then rebuild asterisk....

later,

PaulH


On Tue, 2008-01-08 at 13:16 +0800, Nhadie wrote:
> Hi Matt,
> 
> it seems i don't have that command.
> 
> *CLI> zap show channels
> No such command 'zap' (type 'help' for help)
> *CLI>
> !           abort       add         ael         agent       agi 
> cdr         database    debug       dnsmgr      dont        dump 
> dundi
> extensions  feature     group       help        iax2        include 
> indication  init        load        local       logger      meetme 
> mgcp
> mixmonitor  moh         no          realtime    reload      remove 
> restart     rtp         set         show        sip         skinny 
> soft
> stop        unload
> 
> *CLI> show channeltypes
> Type        Description                    Devicestate  Indications 
> Transfer
> ----------  -----------                    -----------  ----------- 
> --------
> Feature     Feature Proxy Channel Driver   no           yes          no 
> 
> Agent       Call Agent Proxy Channel       yes          yes          no 
> 
> Local       Local Proxy Channel Driver     no           yes          no 
> 
> Skinny      Skinny Client Control Protocol no           yes          no 
> 
> Phone       Standard Linux Telephony API D no           no           no 
> 
> SIP         Session Initiation Protocol (S yes          yes          yes 
> 
> IAX2        Inter Asterisk eXchange Driver yes          yes          yes 
> 
> MGCP        Media Gateway Control Protocol no           yes          no 
> 
> 
> *CLI> show channeltypes
> Type        Description                    Devicestate  Indications 
> Transfer
> ----------  -----------                    -----------  ----------- 
> --------
> Feature     Feature Proxy Channel Driver   no           yes          no 
> 
> Agent       Call Agent Proxy Channel       yes          yes          no 
> 
> Local       Local Proxy Channel Driver     no           yes          no 
> 
> Skinny      Skinny Client Control Protocol no           yes          no 
> 
> Phone       Standard Linux Telephony API D no           no           no 
> 
> SIP         Session Initiation Protocol (S yes          yes          yes 
> 
> IAX2        Inter Asterisk eXchange Driver yes          yes          yes 
> 
> MGCP        Media Gateway Control Protocol no           yes          no 
> 
> 
>      -- Executing NoOp("SIP/104-58ae", "Using CallerID "104" <104>") in 
> new stack
>      -- Executing Set("SIP/104-58ae", "MEETME_ROOMNUM=6000") in new stack
>      -- Executing GotoIf("SIP/104-58ae", "0?USER") in new stack
>      -- Executing Answer("SIP/104-58ae", "") in new stack
>      -- Executing Wait("SIP/104-58ae", "1") in new stack
>      -- Executing Set("SIP/104-58ae", "MEETME_OPTS=") in new stack
>      -- Executing Goto("SIP/104-58ae", "STARTMEETME|1") in new stack
>      -- Goto (from-internal,STARTMEETME,1)
>      -- Executing MeetMe("SIP/104-58ae", "6000||") in new stack
> 
> 
> 
> Matt Riddell wrote:
> > -----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
> > 
> > 
> 
> _______________________________________________
> --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