#asterisk - Mon 9 Apr 2007 between 22:23 and 22:32



Strom_Mnot solid green
HeyItsMeYes, after I plugged in the crossover cable they began to turn flashing red
Strom_Msee, i ask the question one way, and the circuit is in red alarm. I ask it another way, and the circuit is green.
fab5freddyi registered a did and i am now looking to place outbound calls, i put the configurations as per the provider in the proper configuration files. what is my next step?
jncfab5freddy: make some calls?
:)
HeyItsMeOoops sorry I see it. I am getting tired I guess. Thanks for your patience Strom
fender211anyone familiar with UnixODBC?
Strom_MHeyItsMe: now, please, answer my question definitively
HeyItsMecrossover=red, straight=green
fab5freddyjnc: yes, but all i get is user not found when i try to make calls
Strom_Mok, then your problem is not the cable. do not use the crossover cable.
HeyItsMeyes
Strom_Mwhat exactly is your problem, anyway
jncfab5freddy: okay, that's not related to the outgoing call part, are you using a pstn or sip phone to make the call?
fab5freddyjnc: i have sip client
HeyItsMeI have gotten the zaptel to populate the channels, but can't get out.
Strom_M"can't get out"?
fender211so I'm working on setting up voicemail ODBC which I've done before.. this time it's on a Centos 4.4 64 bit O/S and while it compiles fine I get a message about a shared object when trying to use isql to my dsn? Anyone familiar with this setup?
jncfab5freddy: you'll need to make sure that there is an extension handy (users.conf?) and that your sip client is authenticating to it properly.
fab5freddy: you doing this by hand or with a GUI ?
fab5freddyjnc: by hand
jncfabulous
sip set debug
core set verbose 3
that should give you plenty of output going
heh
HeyItsMeI can dial extensions and they work fine. When I try to dial out, I get a dead sound connection
Strom_Mand where is "out"?
fab5freddyjnc: says invalid command in the asterisk cli prompt
jnchm. maybe asterisk 1.4 (which I'm messing with) is different
HeyItsMean outside telephone number
Strom_Mbut are you dialing out over a PRI? SIP trunk? thin air? cheesecake?
jncfab5freddy: set debug, set verbose 4 ?
HeyItsMethe adtran ta 905 t1
fab5freddyjnc: set debug <level>, what do i use for level?
jnc3 is pretty vocal
I'd use that
in /etc/asterisk/users.conf there should be an "extension" (user, really) set up for your SIP device/softphone
Strom_MHeyItsMe: but what kind of entrance facilities do you have from the telco?

Page: 2 9 16 23 30 37 44 51 58 65 72 

IrcArchive