That was coming from the register statements in the sip.conf file. Once I removed those and restarted the sip clients everything started to work.
Thanks! Gary > 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 > > > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
