[asterisk-users] outgoing sip calls work; incoming calls fail

[listmail at websage.ca]

Hi all,

After running for months without issue I've got a situation where
incoming SIP calls to my asterisk server are failing while outbound
calls appear to be working as expected.

The server is a gateway between my home LAN and a broadband cable
connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk
1.6.0.15 (built from ports) and registers to my ISTP no problem.
Outgoing calls can be made successfully and no error messages or
warnings are reported by Asterisk.

However, incoming calls appear to enter my dialplan as desired and go so
far as to start ringing my SIP phone (Grandstream GXP-2000) but drop
after two rings. The caller gets a busy tone and that's it. If I answer
the call before the two rings I just get a moment of dead air and it
drops in the same way.

In the asterisk console (and log file) I see these messages at the fail
point:

[Oct  9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum
retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a
for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt.
[Oct  9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging
up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical
packet (see doc/sip-retransmit.txt)

Okay, so I verified that my firewall is properly accepting traffic on
the range of SIP and RTP ports as specified by my ITSP.

After sending them a sip debug trace my provider said this:

"It appears that your machine is not receiving replies when it tries to
acknowledge the incoming call back to our server.  This could be a
firewall issue or potentially something else that changed without your
knowledge."

Furthermore, they suggested I might try registering and connecting
directly to their Asterisk using only the Grandstream phone. I tried
this and...surprise! Both inbound and outbound calls work fine but
leave me without voicemail or any other services my PBX would be
providing.

Right, so now I'm thinking there must be something wrong with my
Asterisk configuration yet I've made no config changes that would
account for the sudden (and consistent) incoming call failures.

Here's the relevant portions of my sip.conf if it helps (with
credentials and ips replaced by Xs):

[general]
alwaysauthreject=yes
dtmfmode=auto
disallow=all
allow=ulaw

register => XXXX:XXXX at XXX.XXX.XXX.XXX:5060
register => XXXX:XXXX at XXX.XXX.XXX.XXX:5060

[101]
type=friend
context=websage
host=dynamic
deny=0.0.0.0/0
permit=XXX.XXX.XXX.XXX/24
qualify=yes
secret=XXXX
mailbox=101 at default
accountcode=101


I'm now at a complete loss for how to proceed trying to resolve this
and hoping someone with more experience than I on the list might have
some ideas or suggestions.

Any and all advice is warmly appreciated.

Cheers,

GM

-- 
   
Greg Maruszeczka