|_pkNew||can some one please tell me how to get "number of incomming call" ??|
|jazzanova||i am getting UNREACHABLE from my vonage-out|
i'm a newbie
can someone give a suggestion ?
|action||ccole is a noob too. I didn't know Aserisk could work with Vonage|
|ccole||is a noob too. I didn't know Aserisk could work with Vonage|
|action||ccole reads http://www.voip-info.org/wiki/view/Asterisk+and+Vonage|
|russellb||jazzanova: set qualify=no, maybe it's not supported by them|
|jazzanova||russellb: ok, i am going to remove qualify|
but I don't know if the connection is working.
it says that the status is "Registered"
|russellb||have you tried calling it?|
|jazzanova||yes, and its not getting to the server. i am getting a vonage answering machine.|
also, its re-registering very often.
|ccole||How 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.|
|mihinomenest||you need to add the "1" to your Dial()|
|jazzanova||its re-registering every 20 seconds|
also, i am getting this in the log:
Apr 27 22:40:15 WARNING: Unable to get our IP address, Skinny disabled
where do i set the ip address ?
|kiwoneka||good evening to all|
|jazzanova||russellb: 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|
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
|kiwoneka||what 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
|Nugget||have you turned on qualify= for that sip peer?|
sounds like the usual NAT difficulties
|kiwoneka||here is my entry in sip.conf http://pastebin.ca/462183|
should i specify, a value - qualify=500
|_pkNew||hi 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 ??