Hello,
I'm an Asterisk newbie. But I somehow managed to setup a queue where calls
go to an extension 20. I want to be able to overflow this queue to the point
that the 5 agents keep getting calls after calls. All I wanna do is send
lots of calls to extension 20.
For testing this, I would like to try sipp.
I followed the following links to get a basic idea of things, but it's still
hard with practically no documentation for new people, or any kind of help
other than XML scenarios, other than the wiki and documentation (which is
still a little hard to understand by a newb):
http://preview.tinyurl.com/26lpxx
http://preview.tinyurl.com/yqd2qz
http://preview.tinyurl.com/2cddnn (liked this one)
And I got this error 9 out of 10 times:
Unable to bind audio RTP socket (IP=208.69.32.130, port=5607), errno = 99
(Cannot assign requested address).
(The IP belongs do OpenDNS, which is the DNS server company for the server)
If someone can please give me the steps involved to be able to achieve my
goal, that would be splendid. But I would also like to learn what I'm doing
wrong.
Please, please help.
Thanks,
Shoeb Ahmed.
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