Hi guys,
I'm having a really strange problem, which I'm pretty sure has only
appeared since my last upgrade (1.2.12.1) .
It's about NAT and Qualify. I'm using Asterisk to register with some
external SIP providers. However, they're always marked as UNREACHABLE,
when they weren't before!
A typical debug looks like this:
hera*CLI> sip reload
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
== Parsing '/etc/asterisk/sip_notify.conf': Found
<registration>
Reliably Transmitting (no NAT) to 195.189.173.10:5060:
OPTIONS sip:sip.voipfone.co.uk SIP/2.0
Via: SIP/2.0/UDP 87.194.194.249:5060;branch=z9hG4bK07c29ff6;rport
From: "asterisk" <sip:asterisk@87.194.194.249>;tag=as38a9e906
To: <sip:sip.voipfone.co.uk>
Contact: <sip:asterisk@87.194.194.249>
Call-ID: 7dd0587b016684785b7bda1e6f1b2478@87.194.194.249
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 31 Oct 2006 23:22:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
hera*CLI>
<-- SIP read from 195.189.173.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;received=10.0.0.8;rport=65509
Record-Route: <sip:195.189.173.10:5060;lr=on>
From: "asterisk" <sip:asterisk@10.0.0.8>;tag=as38a9e906
To: <sip:sip.voipfone.co.uk>;tag=as7165a192
Call-ID: 7dd0587b016684785b7bda1e6f1b2478@10.0.0.8
CSeq: 102 OPTIONS
User-Agent: Voipfone Sip Network
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:195.189.173.16>
Accept: application/sdp
Content-Length: 0
<repeats three times...... >
--- (12 headers 0 lines)---
Destroying call '7dd0587b016684785b7bda1e6f1b2478@10.0.0.8'
Retransmitting #4 (no NAT) to 195.189.173.10:5060:
OPTIONS sip:sip.voipfone.co.uk SIP/2.0
Via: SIP/2.0/UDP 87.194.194.249:5060;branch=z9hG4bK07c29ff6;rport
From: "asterisk" <sip:asterisk@87.194.194.249>;tag=as38a9e906
To: <sip:sip.voipfone.co.uk>
Contact: <sip:asterisk@87.194.194.249>
Call-ID: 7dd0587b016684785b7bda1e6f1b2478@87.194.194.249
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 31 Oct 2006 23:22:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Oct 31 23:22:23 NOTICE[30434]: chan_sip.c:11613 sip_poke_noanswer: Peer
'duncVF_proxy-out' is now UNREACHABLE! Last qualify: 0
Destroying call '7dd0587b016684785b7bda1e6f1b2478@87.194.194.249'
hera*CLI>
<-- SIP read from 195.189.173.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;received=10.0.0.8;rport=65509
Record-Route: <sip:195.189.173.10:5060;lr=on>
From: "asterisk" <sip:asterisk@10.0.0.8>;tag=as38a9e906
To: <sip:sip.voipfone.co.uk>;tag=as300cbe8d
Call-ID: 7dd0587b016684785b7bda1e6f1b2478@10.0.0.8
CSeq: 102 OPTIONS
User-Agent: Voipfone Sip Network
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:195.189.173.16>
Accept: application/sdp
Content-Length: 0
--- (12 headers 0 lines)---
Destroying call '7dd0587b016684785b7bda1e6f1b2478@10.0.0.8'
hera*CLI> sip no debug
As you can see, the 200 OK's appear to be being ignored... and no amount
of fiddling seems to fix it...
The SIP config is as follows:
type=peer
username=******
fromuser==******
secret==******
fromdomain=sip.voipfone.co.uk
host=sip.voipfone.co.uk
call-limit=5
insecure=very
dtmfmode=rfc2833
nat=yes
qualify=yes
canreinvite=no
context=voipfone-in
disallow=all
allow=g729
allow=ulaw
Any insight would be very much appreciated.
Cheers,
Dave
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