[Asterisk-Users] Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

[Rich Adamson]

> 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.