> After fighting with a "Unable to create/find channel" [1] [2], I gave up > on my previous installation and rebuild my asterisk from CVS-Head. I > guess the Debian package available today is broken somewhere (after a > previous broken release made with an old libpri package), but now I'm > having another issue with my 7960 registration (SIP v. 7.1). > > The call is being (silent) rejected by asterisk, and the "sip debug" is > showing: > [...] > Retransmitting #5 (NAT): > SIP/2.0 407 Proxy Authentication Required > [...] > SIP/2.0 401 Unauthorized > > Even with "set verbose 9" no message is displayed on console regarding > invalid context, password, call attempt... > > Digging the list, I found a message suggesting to "remove" the password > from the sip.conf [3]. I did it and now the calls can be placed (I was > always able to receive calls, even with the broken debian package I had > before). > > Is there *any* reason to this very strange behavior? > > The specific extension sip.conf entry is: > [1234] > type=friend > host=dynamic > qualify=1500 > username=1234 > secret=yeah > auth=md5 > context=cisco > nat=yes > disallow=all > allow=g729 > > I also tried some different approaches, like removing the "auth=md5" tag > and lately removing the password also. Only when no password is set I > was able to place calls. I'm sure the password is the same in the phone > and the sip.conf > > In any scenery, I'm always seeing: > sip show peers > Name/username Host Dyn Nat ACL Mask > 1234/1234 1.2.3.4 D N 255.255.255.255 > Port Status > 63415 OK (982 ms) > > which, I guess, means that the phone is registered with * and the > password has been accepted.
Looks like a couple of problems here. I don't believe the Cisco phone handles md5, so remove that line. In your sip.conf you have "nat=yes", but in the sip show peers it is saying "Nat=N". That would imply that you need to "stop" asterisk and restart it after making such changes. Reload does _not_ reread all such changes, so don't use that until you have a solid understanding of its use. The remainder of your sip.conf definitions look okay other then sooner or later you'll probably want "mailbox=1234" in there to handle voicemail. _______________________________________________ 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
