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 _______________________________________________ --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