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Summary:ASTERISK-15080: [patch] out of dialog SIP_NOTIFY with event='keep-alive'
Reporter:under (under)Labels:
Date Opened:2009-11-05 03:36:25.000-0600Date Closed:2009-11-13 15:01:10.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) notify.diff
Description:Subj. is not handled by *.

[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Invalid SIP message - rejected , no callid, len 438
[Nov  5 18:44:44] VERBOSE[84521] /usr/ports/net/smartswitch/work/smartswitch-5.2.6520/sbc/asterisk/main/logger.c:
<--- SIP read from 150.101.182.62:5063 --->
NOTIFY sip:voip002.dc1.syd.au.ponw.net SIP/2.0
Via: SIP/2.0/UDP 192.168.100.65:5063;branch=z9hG4bK-fa2620d9
From: "user126" <sip:user126@voip002.dc1.syd.au.ponw.net>;tag=1d99345272944056o3
To: <sip:voip002.dc1.syd.au.ponw.net>
Call-ID: cf1f062-c96c3386@192.168.100.65
CSeq: 91 NOTIFY
Max-Forwards: 70
Contact: "user126" <sip:user126@192.168.100.65:5063>
Event: keep-alive
User-Agent: Linksys/SPA962-6.1.5(a)
Content-Length: 0


<------------->
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 0: NOTIFY sip:voip002.dc1.syd.au.ponw.net SIP/2.0 (46)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.100.65:5063;branch=z9hG4bK-fa2620d9 (60)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 2: From: "user126" <sip:user126@voip002.dc1.syd.au.ponw.net>;tag=1d99345272944056o3 (80)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 3: To: <sip:voip002.dc1.syd.au.ponw.net> (37)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 4: Call-ID: cf1f062-c96c3386@192.168.100.65 (40)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 5: CSeq: 91 NOTIFY (15)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 6: Max-Forwards: 70 (16)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 7: Contact: "user126" <sip:user126@192.168.100.65:5063> (52)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 8: Event: keep-alive (17)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 9: User-Agent: Linksys/SPA962-6.1.5(a) (35)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 10: Content-Length: 0 (17)
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: Header 11:  (0)
[Nov  5 18:44:44] VERBOSE[84521] /usr/ports/net/smartswitch/work/smartswitch-5.2.6520/sbc/asterisk/main/logger.c: --- (11 headers 0 lines) ---
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: = No match Their Call ID: 51f5d2c8-2c84056@192.168.100.65 Their Tag 7da234725f923eb6o4 Our tag: as2a6b34a7
[Nov  5 18:44:44] DEBUG[84521] asterisk/channels/chan_sip.c: = No match Their Call ID: b1911f06-4943130@192.168.100.65 Their Tag 1d99345272944056o3 Our tag: as03e9ec0a
[Nov  5 18:44:44] VERBOSE[84521] /usr/ports/net/smartswitch/work/smartswitch-5.2.6520/sbc/asterisk/main/logger.c: Sending to 192.168.100.65 : 5063 (no NAT)
[Nov  5 18:44:44] VERBOSE[84521] /usr/ports/net/smartswitch/work/smartswitch-5.2.6520/sbc/asterisk/main/logger.c:
<--- Transmitting (no NAT) to 192.168.100.65:5063 --->
SIP/2.0 489 Bad event
v: SIP/2.0/UDP 192.168.100.65:5063;branch=z9hG4bK-fa2620d9;received=150.101.182.62
f: "user126" <sip:user126@voip002.dc1.syd.au.ponw.net>;tag=1d99345272944056o3
t: <sip:voip002.dc1.syd.au.ponw.net>;tag=as5dda1ac1
i: cf1f062-c96c3386@192.168.100.65
CSeq: 91 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0

But this is essential to some SPA gateways, which blink ON/OFF as if SIP gateway to which they are connected to reboots.

****** ADDITIONAL INFORMATION ******

seems to be relevant to all current head branches
Comments:By: Leif Madsen (lmadsen) 2009-11-05 08:11:30.000-0600

License pending.
By: Russell Bryant (russell) 2009-11-13 15:01:09.000-0600

Thank you very much for the contribution.

This functionality is actually already present in Asterisk trunk.  Since this qualifies as a new feature for chan_sip, it was not put into Asterisk 1.4.