[asterisk-bugs] [Asterisk 0012921]: Asterisk 1.4.21 breaks realtime sip on 'sip reload'

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Jun 26 00:38:03 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12921 
====================================================================== 
Reported By:                Nuitari
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12921
Category:                   PBX/pbx_realtime
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.21-rc1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             06-23-2008 20:59 CDT
Last Modified:              06-26-2008 00:38 CDT
====================================================================== 
Summary:                    Asterisk 1.4.21 breaks realtime sip on 'sip reload'
Description: 
Using Asterisk 1.4.21 realtime becomes useless after a sip reload is done.


The dynamic information is cleared, however it doesn't get reloaded from
the database when the friend is doing some activity. The only way to make
the friend show again is to force the phone to register again, usually
though a reboot.

The module is res_mysql, from asterisk-addons 1.4.7, works as expected
with Asterisk 1.4.20.
====================================================================== 

---------------------------------------------------------------------- 
 Nuitari - 06-26-08 00:38  
---------------------------------------------------------------------- 
Another thing that I've found. Qualify still works after a "sip reload",
however the peer doesn't show in "sip show peers"


*CLI> Reliably Transmitting (no NAT) to 10.0.2.197:5060:
OPTIONS sip:4000 at 10.0.2.197 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.11:5060;branch=z9hG4bK1c21f394;rport
From: "asterisk" <sip:asterisk at 10.0.2.11>;tag=as62f6ae98
To: <sip:4000 at 10.0.2.197>
Contact: <sip:asterisk at 10.0.2.11>
Call-ID: 60f7f44364e90baf02c365042376c2ef at 10.0.2.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 26 Jun 2008 05:43:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 10.0.2.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.2.11:5060;branch=z9hG4bK1c21f394;rport
From: "asterisk" <sip:asterisk at 10.0.2.11>;tag=as62f6ae98
To: <sip:4000 at 10.0.2.197>;tag=1C9774B5-7B303CA8
CSeq: 102 OPTIONS
Call-ID: 60f7f44364e90baf02c365042376c2ef at 10.0.2.11
Contact: <sip:4000 at 10.0.2.197>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '60f7f44364e90baf02c365042376c2ef at 10.0.2.11'
Method: OPTIONS
Reliably Transmitting (no NAT) to 10.0.2.197:5060:
OPTIONS sip:4000 at 10.0.2.197 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.11:5060;branch=z9hG4bK55eb7519;rport
From: "asterisk" <sip:asterisk at 10.0.2.11>;tag=as33cf8f72
To: <sip:4000 at 10.0.2.197>
Contact: <sip:asterisk at 10.0.2.11>
Call-ID: 770e37ee562cacfb46a9a2e550961a40 at 10.0.2.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 26 Jun 2008 05:43:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 10.0.2.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.2.11:5060;branch=z9hG4bK55eb7519;rport
From: "asterisk" <sip:asterisk at 10.0.2.11>;tag=as33cf8f72
To: <sip:4000 at 10.0.2.197>;tag=2641EB4C-7271F3C7
CSeq: 102 OPTIONS
Call-ID: 770e37ee562cacfb46a9a2e550961a40 at 10.0.2.11
Contact: <sip:4000 at 10.0.2.197>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '770e37ee562cacfb46a9a2e550961a40 at 10.0.2.11'
Method: OPTIONS 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-26-08 00:38  Nuitari        Note Added: 0089259                          
======================================================================




More information about the asterisk-bugs mailing list