[asterisk-bugs] [Asterisk 0017790]: Missing semicolon in SIP-Notify
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Aug 25 09:50:18 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=17790
======================================================================
Reported By: denzs
Assigned To: qwell
======================================================================
Project: Asterisk
Issue ID: 17790
Category: Channels/chan_sip/Subscriptions
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Target Version: 1.8.0
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:
======================================================================
Date Submitted: 2010-08-04 04:01 CDT
Last Modified: 2010-08-25 09:50 CDT
======================================================================
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)
======================================================================
----------------------------------------------------------------------
(0126307) denzs (reporter) - 2010-08-25 09:50
https://issues.asterisk.org/view.php?id=17790#c126307
----------------------------------------------------------------------
In ast_full.log you can see a snippet from the full log which should
provide some debug information.
first i thought the problem would occur after registration of a peer,
but during a test it occured after restarting the asterisk while the
appropriate phone was offline.
right now iam checkin out the latest trunk and will do some further
tests...
Issue History
Date Modified Username Field Change
======================================================================
2010-08-25 09:50 denzs Note Added: 0126307
======================================================================
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