[asterisk-bugs] [Asterisk 0013024]: Call blocked after 1 minute 45 seconde

[noreply at bugs.digium.com]

The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=13024 
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Reported By:                olivier1010
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13024
Category:                   PBX/General
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0-beta9 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             07-08-2008 09:28 CDT
Last Modified:              07-08-2008 09:28 CDT
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Summary:                    Call blocked after 1 minute 45 seconde
Description: 

When doing calls between two extensions (G711a codec), the call is dropped
after 1m45sec on the called phone.

Then the calling phone hangup the call at about 2m05sec.

After the called phone hangup, issuing a "sip show channels" gives the two
channels actives.

192.168.200.60   41          2bb589e920880fe  0x1000 (g722)    No      
Rx: ACK                   
192.168.200.40   40          a038a43dc9b9160  0x1000 (g722)    No      
Rx: ACK  

After the second phone hangup, the channels are effectively destroyed.

This is perfectly reproductible.

Using Aastra 57i phones firmware 2.2.1. and FreePBX 2.4.1.


i've notified this registration error, this is strange because the phone
does register correctly at beginning :

<--- SIP read from UDP://192.168.200.60:5060 --->
REGISTER sip:192.168.200.240 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bK0eebc1279e9885cab
Max-Forwards: 70
From: <sip:43 at 192.168.200.240>;tag=b9a1ec5bcb
To: <sip:43 at 192.168.200.240>
Call-ID: 099bee27fef189dc
CSeq: 20235 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest
username="43",realm="asterisk",nonce="33b166a6",uri="sip:192.168.200.240",response="eba83d472e4b97fc34e7bf6272926704",algorithm=MD5
Contact: <sip:43 at 192.168.200.60:5060;transport=udp>
User-Agent: Aastra 57i/2.2.1.25
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.200.60 : 5060 (NAT)

<--- Transmitting (no NAT) to 192.168.200.60:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.200.60:5060;branch=z9hG4bK0eebc1279e9885cab;received=192.168.200.60
From: <sip:43 at 192.168.200.240>;tag=b9a1ec5bcb
To: <sip:43 at 192.168.200.240>;tag=as175efbde
Call-ID: 099bee27fef189dc
CSeq: 20235 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="7354d4ea"
Content-Length: 0





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Issue History 
Date Modified   Username       Field                    Change               
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07-08-08 09:28  olivier1010    Asterisk Version          => 1.6.0-beta9     
07-08-08 09:28  olivier1010    SVN Branch (only for SVN checkou => N/A          
  
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