[asterisk-users] SIP Options Reply Ignored

[Ian Clough]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alex Balashov
Sent: 03 June 2007 18:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Options Reply Ignored

On Sun, 3 Jun 2007, Ian Clough wrote:

> The problem comes when I try to use a SIP phone at home (also behind a 
> NAT router). The phone registers correctly and I can see the SIP OPTONS 
> packets being sent to the phone (SNOM 190).  I can see an OK reply being 
> received by Asterisk (using SIP DEBUG). However the OK reply appears to 
> be ignored and a retransmission is made and the phone is marked as 
> UNREACHABLE and will not accept any calls.

Alex>   Wait, so this is the phone registering to Asterisk?  Any
inconsistencies 
Alex> in the source/destination ports vis-a-vis the NAT state pinholes?

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671

Yes once the phone registers OK asterisk sends the OPTIONS packets
(qualify=yes)
This is an example. It shows asterisk reading a reply from by phone to
transmission #3 and then sending retransmission #4

intechdial*CLI>
<--- SIP read from xxx.xxx.xxx.xxx:2057 ---> SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.13.251.93:5060;branch=z9hG4bK24ffe501;rport=40804
From: "asterisk" <sip:asterisk at 207.13.251.93>;tag=as35e434f3
To: <sip:665 at xxx.xxx.xxx.xxx:2057;line=15aykp4d>
Call-ID: 053d0ce438527a5309581e6f53e3176a at 207.13.251.93
CSeq: 102 OPTIONS
Contact: <sip:665 at 192.168.1.50:2057;line=15aykp4d>
User-Agent: snom190/3.60x
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Retransmitting #4 (NAT) to xxx.xxx.xxx.xxx:2057:
OPTIONS sip:665 at xxx.xxx.xxx.xxx:2057;line=15aykp4d SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK24ffe501;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at yyy.yyy.yyy.yyy>;tag=as35e434f3
To: <sip:665 at xxx.xxx.xxx.xxx:2057;line=15aykp4d>
Contact: <sip:asterisk at yyy.yyy.yyy.yyy>
Call-ID: 053d0ce438527a5309581e6f53e3176a at yyy.yyy.yyy.yyy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-trunk-r63567
Date: Thu, 31 May 2007 11:01:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Where xxx.xxx.xxx.xxx is the external address of my home router
Yyy.yyy.yyy.yyy is the external address of the office router
207.13.251.93 is the internal address of the asterisk server
192.168.1.50 is the internal address of the phone

Ian C