Strange call out from my trixbox

General Help
1

I have a trunk setup to gizmoproject that seems to work both outbound and inbound. Now when I do SIP deubug peer, I see following at regular intervals along with normal user registration traffic. my.ip.add is my IP address and all other IDs are hacked to protect the inocent. I am behind home nat router.

[code:1]
<-- SIP read from 198.65.166.131:5060:
SIP/2.0 404 Invalid number or user is offline
Via: SIP/2.0/UDP my.ip.add:5060;branch=z9bK13;rport=5060
From: “Unknown” sip:[email protected];tag=as4c
To: sip:proxy01.sipphone.com;tag=21a48342ba40a2e.fdcb
Call-ID: [email protected]
CSeq: 102 OPTIONS
Content-Length: 0
[/code:1]

Who is “Unknown”? sip:[email protected]

I do not have any “unknown” username or extension in my config files.

2

Asterisk uses the OPTIONS request message to check for reachability if
you have qualify=yes configured.

http://www.voip-info.org/wiki-asterisk+sip+qualify

[quote] -----Original Message-----
From: [email protected]
[mailto:[email protected]] On
Behalf Of baysoor
Sent: 03 December 2006 07:53
To: [email protected]
Subject: [Amportal-users] strange call out from my trixbox

I have a trunk setup to gizmoproject that seems to work both
outbound and inbound. Now when I do SIP deubug peer, I see
following at regular intervals along with normal user
registration traffic. my.ip.add is my IP address and all
other IDs are hacked to protect the inocent. I am behind home
nat router.

<-- SIP read from 198.65.166.131:5060: SIP/2.0 404
Invalid number or user is offline Via: SIP/2.0/UDP
my.ip.add:5060;branch=z9bK13;rport=5060
From: “Unknown” sip:[email protected];tag=as4c
To: sip:proxy01.sipphone.com;tag=21a48342ba40a2e.fdcb
Call-ID: [email protected]
CSeq: 102 OPTIONS
Content-Length: 0

Who is “Unknown”? sip:[email protected]

I do not have any “unknown” username or extension in my config files.

[/quote]

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3

[quote=“steve.langstaff at cit…”]Asterisk uses the OPTIONS request message to check for reachability if
you have qualify=yes configured.

http://www.voip-info.org/wiki-asterisk+sip+qualify
[/quote]

Steve,

Thank you for the correct answer. After I removed the line, the calls stopped. I also found out that unknown was me! callerid = unknown in sip.conf.

BTW I read on some forum and also the voip-info wiki that qualify=yes is recommended behind NAT. I am behind NAT and have experienced that for reasons unknown to me, my public IP changes once in a while. I do have dynamic host configuration but somehow trixbox fails to register back to gizmo. The only solution is to reboot router and reboot asterisk machine completely not just amportal restart. I though that qualify would help.