[asterisk-users] No reply to our critical packet

[Danny Nicholas]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Roman Odaisky
Sent: Friday, March 13, 2009 9:57 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] No reply to our critical packet

Hi,

I've installed Asterisk for use as a SIP server. I can call people, but one 
strange thing happens: if I call someone with a SIP account outside my
server 
(for example, sip:enum-echo-test at sip.nemox.net) everything is fine, if I
call 
any Asterisk extension it also works, but the call gets disconnected in
about 
20 seconds. To be exact, audio is turned off but the SIP client still thinks

it's connected.

Logs say "no reply to our critical packet". tcpdump shows that the packet
does  
arrive at the destination.

sip set debug shows this is what the packet contains:

Retransmitting #6 (NAT) to 77.239.189.223:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
77.239.189.223;branch=z9hG4bK-d8754z-db899ced94cc7fd3-1---d8754z-;received=7
7.239.189.223
From: "Roma"<sip:roma at qwertty.com;transport=UDP>;tag=01785d5e
To: <sip:echo at qwertty.com;transport=UDP>;tag=as068592d2
Call-ID: ZTkzNjYxNzZmOWMzY2ZhOTdjMWIwYTEwZTYxZmUyZTY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:echo at 78.46.49.80>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 25952 25952 IN IP4 78.46.49.80
s=session
c=IN IP4 78.46.49.80
t=0 0
m=audio 30606 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

There's NAT: computer (192.168.1.2) behind a router (77.239.189.223), the 
server (78.46.49.80) doesn't have any NAT. I have even set DMZ host to 
192.168.1.2, so I'm sure all packets reach it.

As far as I understand, Asterisk expects the SIP client to reply to that 
packet with an ACK, the client receives the packet but does not reply. What 
have I configured incorrectly? In sip.conf I have nat=yes (otherwise I don't

hear anything), whatever I do with NAT settings of SIP clients does not
help. 
Maybe there's something wrong with the headers of the packet that makes the 
client think the packet is misaddressed? Twinkle says, "you have the 
following registrations <sip:roma at 192.168.1.2>" while I'd expect 
<sip:roma at qwertty.com>. So how do I make sure the client sends its ACK?

-- 
TIA
Roman.

--> 

Two thoughts (both could be wrong)
1. Do you have the incoming 10000-20000 holes in your firewall so the remote
server can get it's reply back to *?
2. If #1 is ok, try putting an Answer command in front of your Dial Command.

Danny Nicholas