I have 2 asterisk boxes connected via SIP
box 1 sip peer connected to box 2 (ip addresses intentionally removed)
[ast20]
type=friend
host=x.x.x.20
insecure=very
context=subscriber
dtmfmode=inband
qualify=no
canreinvite=no
disallow=all
allow=ulaw
box 2 sip peer connected to box 1
[sbb19]
type=friend
host=64.1.8.19
insecure=very
context=inbound
dtmfmode=inband
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
I then have 2 UAs registed on box 1, both have identical configs with
the exception of username, but one is a Polycom IP501 and the other is a
Linksys PAP2
The IP 501 can call to box 2 with no issues, also calls originated on a
PRI connected to box 1 connect to box 2 with no issues.
The Linksys UA can not call box 2, here is the error (numbers
intentionally removed);
-- Executing dial("SIP/######0850-b6669f58",
"SIP/######7581@ast20")
-- Called ######7581@ast20
Nov 7 07:20:45 NOTICE[21059]: chan_sip.c:9709 handle_response_invite:
Failed to authenticate on INVITE to '"name removed"
<sip:######0850@64.1.8.19>;tag=as38826922'
-- SIP/ast20-09c8b110 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
I have looked at sip debugs from both scenarios, and the invites from
box 1 to box 2 look nearly identical, box 2 never shows the call when it
fails.
I am assuming that there is something that needs to be changed on the
ATA or peer config to get it to be able to call via box1 to box2 without
requiring authentication, but can not figure out what.
Any ideas?
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