[Asterisk-Users] Incoming SIP from Cisco 7206
Scott Miller
scoscmil at imap.iu.edu
Wed Aug 3 14:09:17 MST 2005
I am running an Asterisk server through a Cisco 7206 PSTN gateway. I am
able to make outgoing SIP calls without a problem, though incoming calls
have been somewhat of a problem. I am not sure exactly how sip.conf should
look in such a scenario.
I believe most Cisco gateways are just managed through ACL's, with no
authentication, so I think I have the outgoing "peer" statement right, but I
have no idea where to start on the incoming "user" statement. Here's my
sip.conf (configured through AMP).
[gk02-inbound]
type=user
host=10.0.106.10
context=from-pstn
[gk01]
type=peer
host=10.0.50.10
When a call comes it, about every second I get this..
Aug 1 11:53:49 DEBUG[4076]: Stopping retransmission on
'495FF45C-1DB11DA-8E67BEF0-CE47D9F7 at 10.0.106.10' of Response 101: Found
Aug 1 11:53:49 DEBUG[4076]: Check for res for
Aug 1 11:53:49 DEBUG[4076]: is not a local user
Aug 1 11:53:49 DEBUG[4076]: is not a local user
Any help would be appreciated..
Thanks,
Scott Allen Miller
Research Assistant
Telecommunications
University Information Technology Services
Indiana University
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