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given that AAH is asterisk + amp + bunch of other stuff, can't you just
setup incoming routing and outgoing routing within AMP?
Ira wrote:
At 09:04 PM 03/16/2006, you wrote:
6 different companies with 6 different IVR's and different ring groups
you get ping time in the status page if your extension.conf has qualify=yes
Quoting Samy Antoun [EMAIL PROTECTED]:
--- Sergey Okhapkin [EMAIL PROTECTED] wrote:
Hmm.. What is the output of sip show users and sip show peers?
sip show users
Username Def.Context ACL NAT
200
Hi All,
I am having trouble getting speex to work on asterisk. I downloaded
1.0.4 from speex.org, download libogg from vorbis. ./configure, make and
make install for both and then recompile my asterisk 1.0.5, yet I am getting
error when trying to load codec_speex.so. Below is the error from
