[asterisk-users] Huh? IP address ending with 611

[Mike]

The weird thing is that the phone actually works for now, but I want to
proactively fix anything that may go wrong (this phone _has_ to work until
Saturday)

A SIP debug gives me this:


---
Scheduling destruction of call 'b5aedb83-54d7a3f5-6340ca5a at 192.168.10.100'
in 15000 ms
hd-t3143cl*CLI> sip 
<-- SIP read from 66.38.177.61:5060: 
REGISTER sip:pbx.test.ca SIP/2.0
Via: SIP/2.0/UDP 66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701
From: "notarius-phone-1"
<sip:notarius-phone-1 at pbx.test.ca>;tag=28A4D21C-80220BAB
To: <sip:notarius-phone-1 at pbx.test.ca>
CSeq: 1803 REGISTER
Call-ID: 4e7c1af8-56c2c3aa-fb7af919 at 192.168.0.192
Contact: <sip:notarius-phone-1 at 66.38.177.611:5060>;methods="INVITE, ACK,
BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
REFER"
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Authorization: Digest username="notarius-phone-1", realm="asterisk",
nonce="20d11a72", uri="sip:pbx.test.ca",
response="dbcfab79977a81ea3681bbe574bd1c37", algorithm=MD5
Max-Forwards: 70
Expires: 30
Content-Length: 0


--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Apr 13 08:28:26 WARNING[20348]: chan_sip.c:7036 check_via: '66.38.177.611'
is not a valid host
Transmitting (NAT) to 66.38.177.61:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701;received=66.38.177.61
From: "notarius-phone-1"
<sip:notarius-phone-1 at pbx.test.ca>;tag=28A4D21C-80220BAB
To: <sip:notarius-phone-1 at pbx.test.ca>
Call-ID: 4e7c1af8-56c2c3aa-fb7af919 at 192.168.0.192
Seq: 1803 REGISTERp 
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:notarius-phone-1 at 15.123.76.34>
Content-Length: 0


---
Transmitting (NAT) to 66.38.177.61:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701;received=66.38.177.61
From: "notarius-phone-1"
<sip:notarius-phone-1 at pbx.test.ca>;tag=28A4D21C-80220BAB
To: <sip:notarius-phone-1 at pbx.test.ca>;tag=as42c283c1
Call-ID: 4e7c1af8-56c2c3aa-fb7af919 at 192.168.0.192
CSeq: 1803 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 30
Contact: <sip:notarius-phone-1 at 66.38.177.611:5060>;expires=30
Date: Fri, 13 Apr 2007 12:28:26 GMT
Content-Length: 0 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alex Balashov
Sent: Thursday, April 12, 2007 23:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Huh? IP address ending with 611


Can you do a packet capture and see what the actual contact (Via) in fact
says right before it hits Asterisk?

On Thu, 12 Apr 2007, Mike said something to this effect:

> Hi,
>
> I`m getting this (from one of my registered phone that has been 
> installed at some location I can`t access at the moment) in the 
> Asterisk CLI.  Notice the last 3 digits of the IP address in the error
message:
>
> Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611'
> is not a valid host
>
> Of course it's not a valid host!  But, when using "sip show peers", 
> the phone is actually listed with IP address 67.39.117.61 (which makes 
> alot more sense, but then again I shouldn`t be getting any warning in this
case).
>
> Where is that error coming from then? Are there any consequences?
>
> Using 1.2.13.
>
> Mike
>

--
Alex Balashov <sasha at presidium.org>
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