It's not clear how you are making the call.
You should be able to call directly from either phone to the other by dialing 5011 or 5012, respectively, if
your context "local" indeed contains the those extensions, which is not clear from your configuration excerpts.
But it seems you are calling another user on carolina.net, who is registered with that provider from your asterisk,
so the call will loop back to Asterisk. SIP does not really have a good way to handle such loopbacks, and
therefore you get the error.
If you want to make this work you need to load a second SIP channel driver on your asterisk listening on a different port,
The changes are not difficult.
It also seems that both phones are sitting on the same IP address and port, how can that be?
Oh, I see the error message is actually coming from the sip phone, and it's because those phones
have the same IP address, and therefore a loop is detected there. Is this just ONE phone with two
proxy-accounts or personalities?
Gary Carr wrote:
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 => [EMAIL PROTECTED]/5011 <mailto:[EMAIL PROTECTED]/5011>
register => [EMAIL PROTECTED]/5052 <mailto:[EMAIL PROTECTED]/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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