#asterisk - Sat 28 Apr 2007 between 00:40 and 01:27

NY Lost Funds



_pkNewcan some one please tell me how to get "number of incomming call" ??
jazzanovai am getting UNREACHABLE from my vonage-out
i'm a newbie
can someone give a suggestion ?
actionccole is a noob too. I didn't know Aserisk could work with Vonage
ccoleis a noob too. I didn't know Aserisk could work with Vonage
actionccole reads http://www.voip-info.org/wiki/view/Asterisk+and+Vonage
ccolereads http://www.voip-info.org/wiki/view/Asterisk+and+Vonage
russellbjazzanova: set qualify=no, maybe it's not supported by them
jazzanovarussellb: ok, i am going to remove qualify
but I don't know if the connection is working.
it says that the status is "Registered"
russellbhave you tried calling it?
jazzanovayes, and its not getting to the server. i am getting a vonage answering machine.
also, its re-registering very often.
ccoleHow come I can get an outbound call to work with: exten=>_1NXXNXXXXXX,1,Dial... but I cannot get an outbound call to work with: exten=> _NXXNXXXXXX,1,Dial... ?? I do not want to have to type a '1' before dialing a phone number.
mihinomenestyou need to add the "1" to your Dial()
jazzanovaits re-registering every 20 seconds
also, i am getting this in the log:
Apr 27 22:40:15 WARNING[20941]: Unable to get our IP address, Skinny disabled
where do i set the ip address ?
kiwonekagood evening to all
jazzanovarussellb: ok, when I call to my phone, i can see in asterisk sip debug output. but i don't know where the call is routed
vonkleistHi everybody
I live in mexico, and just bought a tdm22b
I installed asterisknow, and now ihave all my extensions working
But what I can do, is to get calls out to my 2 lines
Extensions rings, and rings, but call never get established
also, it seems like calls aren't going out
ummm... tdm400p, I mean
kiwonekawhat a night
i need some netwrking help
i sent grandma a polycom601, now it connects and gets provisioned just fine
but we cant hear grandma
i have tried alot of things, nw i need some professional advice
Nuggethave you turned on qualify= for that sip peer?
sounds like the usual NAT difficulties
kiwonekahere is my entry in sip.conf http://pastebin.ca/462183
should i specify, a value - qualify=500
_pkNewhi every body
i want to make reports of a call center using queue_logs
how do i get the number of calls offered/answered/abondaned ??

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NY Lost Funds