[Asterisk-Users] sip to sip calls thru asterisk

[Gary Carr]

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

register => garycarr at sip.carolina.net/5011
register => cisco7960 at sip.carolina.net/5052

[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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