How do I tell ser to route invites for every sip uri beginning with the number 1 to
another sip server? I would want to route 1000(a)domain.com , 1001(a)domain.com and
1002(a)domain.com all to 5.0.0.10
my ser box is 0.8.12 on solaris 10 (5.0.0.12)
5.0.0.10 is a cisco call manager 4.0 with a sip trunk
this sccp (skinny) to sip bridging is working a little bit but not completely. i get
one-way audio then the calls disconnect after about 10 seconds if i dial sip to sccp. but
if i dial sccp to sip, then the call rings through, but media (rtp) never flows either way
and call drops after ~33 seconds
in debugging a little with the cisco tac, we've seen that the TO: field in a invite to
the ccm box is coming in with the wrong address...it's coming in 1001(a)5.0.0.12 but it
seems that it should say 1001(a)5.0.0.10
is there a way to fix this?
aaron
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