[asterisk-bugs] [Asterisk 0018348]: No answer to OPTIONS packet because Asterisk not looking for 's' in default context

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18348 
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Reported By:                shmaize
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18348
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.0 
JIRA:                       SWP-2639 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-22 02:54 CST
Last Modified:              2011-02-02 06:17 CST
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Summary:                    No answer to OPTIONS packet because Asterisk not
looking for 's' in default context
Description: 
Hello,

asterisk-1.8.0 installed through yum on CentOS 5.5 32bit. 

My SIP provider is checking if I'm alive with OPTIONS. I must answer to
that request with "SIP/2.0 200 OK". The default behavior is to check if I
have a peer with that IP, then check for "s" in it's context. If not check
for "s" in the guests context (context=XX from general section of
sip.conf). Or I'm missing something?

With 1.8.0 the extension part is missing. Some debug logs:
1.1.1.1 is the SIP provider, 2.2.2.2 is my IP. 

/var/log/asterisk/full:
[Nov 18 16:38:05] DEBUG[6942] chan_sip.c: = Looking for  Call ID:
269dd7c535d89368dbb1770a86cd13df at 1.1.1.1 (Checking From) --From tag 24178
--To-tag   
[Nov 18 16:38:05] DEBUG[6942] acl.c: For destination '1.1.1.1', our source
address is '2.2.2.2'.
[Nov 18 16:38:05] DEBUG[6942] chan_sip.c: Setting SIP_TRANSPORT_UDP with
address 2.2.2.2:5060
[Nov 18 16:38:05] DEBUG[6942] chan_sip.c: Allocating new SIP dialog for
269dd7c535d89368dbb1770a86cd13df at 1.1.1.1 - OPTIONS (No RTP) 
[Nov 18 16:38:05] DEBUG[6942] chan_sip.c: **** Received OPTIONS (3) -
Command in SIP OPTIONS
[Nov 18 16:38:05] DEBUG[6942] chan_sip.c: Trying to put 'SIP/2.0 404' onto
UDP socket destined for 1.1.1.1:5080
[Nov 18 16:38:05] DEBUG[6942] chan_sip.c: SIP message could not be
handled, bad request: 269dd7c535d89368dbb1770a86cd13df at 1.1.1.1


<--- SIP read from UDP:1.1.1.1:5080 ---> 
OPTIONS sip:2.2.2.2:5060 SIP/2.0
Call-ID: 7ee61f3291443916e4d631a411c231b5 at 1.1.1.1
CSeq: 100 OPTIONS
From: "Voxbone Monitoring" <sip:voxmon at 1.1.1.1>;tag=95055
To: <sip:2.2.2.2:5060>
Max-Forwards: 30
Route: <sip:2.2.2.2:5060>
Via: SIP/2.0/UDP
1.1.1.1:5080;branch=z9hG4bKfabde850548854ad76efa0335e4bfe82
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Looking for  in guests (domain 2.2.2.2:5060)

<--- Transmitting (no NAT) to 1.1.1.1:5080 ---> 
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
1.1.1.1:5080;branch=z9hG4bKfabde850548854ad76efa0335e4bfe82;received=1.1.1.1
From: "Voxbone Monitoring" <sip:voxmon at 1.1.1.1>;tag=95055
To: <sip:2.2.2.2:5060>;tag=as5967e5ef
Call-ID: 7ee61f3291443916e4d631a411c231b5 at 1.1.1.1
CSeq: 100 OPTIONS
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog
'7ee61f3291443916e4d631a411c231b5 at 1.1.1.1' in 32000 ms (Method: OPTIONS)

In the part "Looking for  in guests" it should be "Looking for s in
guests". 

sip.conf
[general]

bindport=5060           ; Port to bind to (SIP is 5060)
srvlookup=yes
bindaddr=2.2.2.2
disallow=all                                                         
allow=alaw                                                          
dtmfmode=rfc2833
allowguest=no
canreinvite=no
context=guests ; Send unknown SIP callers to this context          
callerid=Unknown
allowoverlap=no                 ; Disable overlap dialing support.
(Default is yes)
allowtransfer=no
useragent=Asterisk PBX
externip=2.2.2.2


When testing with 1.6.2.14 (installed from yum) it's OK:
<------------->
--- (9 headers 0 lines) ---
Looking for s in guests (domain 2.2.2.2)

<--- Transmitting (no NAT) to 1.1.1.1:5080 --->
SIP/2.0 200 OK

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---------------------------------------------------------------------- 
 (0131386) shmaize (reporter) - 2011-02-02 06:17
 https://issues.asterisk.org/view.php?id=18348#c131386 
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What is the status of this issue? Any progress? 
I think it's a big problem. Many VoIP providers are checking if the peer
is alive with OPTIONS packets and considering "dead" if no response is send
back. So right now asterisk 1.8 DO NOT support that kind of trunks. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-02 06:17 shmaize        Note Added: 0131386                          
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