[asterisk-users] conferencing help

[Nhadie]

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
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> 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
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