[asterisk-bugs] [Asterisk 0017790]: Missing semicolon in SIP-Notify

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=17790 
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Reported By:                denzs
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17790
Category:                   Channels/chan_sip/Subscriptions
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                       SWP-2005 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.8 
SVN Revision (number only!): 280878 
Request Review:              
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Date Submitted:             2010-08-04 04:01 CDT
Last Modified:              2010-08-23 14:04 CDT
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Summary:                    Missing semicolon in SIP-Notify
Description: 
I see a lots of messages like this in the CLI...

It seems like there is a semicolon missing?

NOTIFY sip:1686 at 192.168.51.201:5060user=phone
shouldn it be
NOTIFY sip:1686 at 192.168.51.201:5060;user=phone


Sending to 192.168.51.201:5060 (no NAT)
[Aug  4 11:29:07] ERROR[486]: netsock2.c:245 ast_sockaddr_resolve:
getaddrinfo("192.168.51.201", "5060user=phone", ...): Servname not
supported for ai_socktype
[Aug  4 11:29:07] WARNING[486]: chan_sip.c:12820
__set_address_from_contact: Invalid host name in Contact: (can't resolve in
DNS) : '192.168.51.201:5060user=phone'
Scheduling destruction of SIP dialog
'66e9b8d9258832ce00d06b607e0243e3 at 192.168.51.123:5060' in 32000 ms (Method:
NOTIFY)
Reliably Transmitting (no NAT) to 192.168.51.201:5060:
NOTIFY sip:1686 at 192.168.51.201:5060user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.51.123:5060;branch=z9hG4bK08321305
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.51.123>;tag=as59f03aba
To: <sip:1686 at 192.168.51.201:5060user=phone>
Contact: <sip:asterisk at 192.168.51.123:5060>
Call-ID: 66e9b8d9258832ce00d06b607e0243e3 at 192.168.51.123:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX SVN-trunk-r280810
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 99

Messages-Waiting: no
Message-Account: sip:asterisk at 192.168.51.123:5060
Voice-Message: 0/0 (0/0)

---

<--- Transmitting (no NAT) to 192.168.51.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.51.201:5060;branch=z9hG4bK1523223228;received=192.168.51.201;rport=5060
From:
<sip:1686 at poc.lvmtest.ar.intranet.gonicus.de;user=phone>;tag=766983564
To:
<sip:1686 at poc.lvmtest.ar.intranet.gonicus.de;user=phone>;tag=as5a5cd7fd
Call-ID: 117896864-5060-1 at 192.168.51.201
CSeq: 4420 REGISTER
Server: Asterisk PBX SVN-trunk-r280810
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:1686 at 192.168.51.201:5060;user=phone>;expires=300
Date: Wed, 04 Aug 2010 09:29:07 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '117896864-5060-1 at 192.168.51.201' in
32000 ms (Method: REGISTER)
[Aug  4 11:29:07] ERROR[459]: netsock2.c:245 ast_sockaddr_resolve:
getaddrinfo("192.168.51.201", "5060user=phone", ...): Servname not
supported for ai_socktype
[Aug  4 11:29:07] WARNING[459]: chan_sip.c:12820
__set_address_from_contact: Invalid host name in Contact: (can't resolve in
DNS) : '192.168.51.201:5060user=phone'
Retransmitting https://issues.asterisk.org/view.php?id=1 (no NAT) to
192.168.51.201:5060:
NOTIFY sip:1686 at 192.168.51.201:5060user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.51.123:5060;branch=z9hG4bK074f9545
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.51.123>;tag=as626d156f
To: <sip:1686 at 192.168.51.201:5060user=phone>
Contact: <sip:asterisk at 192.168.51.123:5060>
Call-ID: 5c08a9c229b34d5a23e3183c58953d48 at 192.168.51.123:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX SVN-trunk-r280810
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 99

Messages-Waiting: no
Message-Account: sip:asterisk at 192.168.51.123:5060
Voice-Message: 0/0 (0/0)

---
Retransmitting https://issues.asterisk.org/view.php?id=1 (no NAT) to
192.168.51.201:5060:
NOTIFY sip:1686 at 192.168.51.201:5060user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.51.123:5060;branch=z9hG4bK08321305
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.51.123>;tag=as59f03aba
To: <sip:1686 at 192.168.51.201:5060user=phone>
Contact: <sip:asterisk at 192.168.51.123:5060>
Call-ID: 66e9b8d9258832ce00d06b607e0243e3 at 192.168.51.123:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX SVN-trunk-r280810
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 99

Messages-Waiting: no
Message-Account: sip:asterisk at 192.168.51.123:5060
Voice-Message: 0/0 (0/0)


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Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-23 14:04 qwell          Status                   acknowledged =>
feedback
======================================================================