asterisk users - Mar 2005 - Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

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.

Any ideas?

[1] 
http://lists.digium.com/pipermail/asterisk-users/2005-February/090364.html
[2] http://lists.digium.com/pipermail/asterisk-users/2005-March/092083.html
[3]
http://lists.digium.com/pipermail/asterisk-users/2004-September/064998.html

> 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.