[Asterisk-Users] 488 Not Acceptable Here

[Nabeel Jafferali]

I have a whole bunch of remote devices connected to my Asterisk box,
including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only
rolled out recently and I am having a problem that is intermittent and
inconsistent.

It happens to some users but not other users on the same ISP. It happens to
users in 2 different countries where the Internet setup (NAT issues) are
completely different. It happens to some users intermittently and some users
on every call. Note all the PAP2-NAs are running the same and latest
firmware.

Am I missing something completely obvious? Is there a way to see why
Asterisk is sending 488 (i.e. what is not acceptable?). sip debug peer and
sip.conf is below:

Sip read: 
INVITE sip:800 at sip.x2n.net SIP/2.0
v: SIP/2.0/UDP 216.252.155.227:11288;branch=z9hG4bK-8079155a;rport
f: Bode Dar <sip:3255 at sip.x2n.net>;tag=faf9295d7d8c85e6o0
t: <sip:800 at sip.x2n.net>
i: 1db23be9-a4adb4ca at 192.168.2.3
CSeq: 103 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="3255",realm="asterisk",nonce="********",uri="sip:800 at sip.x2n.net",
algorithm=MD5,response="********************************"
m: Bode Dar <sip:3255 at 216.252.155.227:11288>
Expires: 240
User-Agent: Linksys/PAP2-2.0.14(LSVa)
l: 402
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
k: x-sipura
c: application/sdp

v=0
o=- 13600 13600 IN IP4 216.252.155.227
s=-
c=IN IP4 216.252.155.227
t=0 0
m=audio 14274 RTP/AVP 18 0 2 8 96 97 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

15 headers, 18 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 216.252.155.227:11288;branch=z9hG4bK-8079155a
From: Bode Dar <sip:3255 at sip.x2n.net>;tag=faf9295d7d8c85e6o0
To: <sip:800 at sip.x2n.net>;tag=as0be177dc
Call-ID: 1db23be9-a4adb4ca at 192.168.2.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:800 at 38.116.194.7>
Content-Length: 0

 to 216.252.155.227:11288


The user's sip.conf entry is:

[3255]
type=friend            
username=3255          
secret=******          
accountcode=3255     
callerid="Bode Dar" <>
context=clients-int
host=dynamic 
qualify=yes
nat=yes
disallow=all
allow=g729
allow=ulaw


--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990