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I have a test box setup and I can make outbound
calls on the PSTN thru the diguim card, however I can not make a sip user to sip
user call by dialing the extensions. I am getting the following
error.
-- Called cisco7960
-- Got SIP response 482 "Loop Detected" back from 208.218.14.123 == No one is available to answer at this time CLI> sip show
peers
Name/username Host Dyn Nat ACL Mask Port Status cisco7960/5052
208.218.14.123 D N
255.255.255.255 5060 OK (1 ms)
garycarr/5011
208.218.14.123 D N
255.255.255.255 5060 OK (1 ms)
sip.conf statements
[cisco7960]
type=friend host=dynamic nat=yes qualify=200 dtmfmode=rfc2833 canreinvite=no mailbox=5052 callerid="Cisco 7960" context=local [garycarr]
type=friend host=dynamic nat=yes qualify=200 dtmfmode=rfc2833 canreinvite=no mailbox=5011 callerid="Gary Carr" context=local extensions.conf statements
exten =>
5011,1,dial(SIP/garycarr,20,tr)
exten => 5052,1,dial(SIP/cisco7960,20,tr) Is this a possible nat issue? I can make a good
call from behind the firewall doing sip to pstn so it seems 2 way traffic thru
the firewall is working.
I am still sifting thru the sip debug info but
anyone has any ideas that would be great.
Gary
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