[SIPForum-discussion] Server Error 503

[amirul ahsan]

plz send wireshark/ethereal trace. 

--- On Thu, 12/11/08, Robert Vos <robert at dcomt.com> wrote:
From: Robert Vos <robert at dcomt.com>
Subject: [SIPForum-discussion] Server Error 503
To: discussion at sipforum.org
Date: Thursday, December 11, 2008, 6:44 PM



 
 

Hi,
 
I am very new to SIP coding, and am stuck with a 
problem
 
When I try to terminate a call, using the BYE 
method, I am getting a Server error 503 message back from Asterisk.  this 
only happens, however, if both my softphone and the device i'm calling are using 
the same codec.  
If my softphone uses ULaw and the device ALaw, this 
does not occur.
 
Here is the Bye message and the 
response:
 
 
BYE sip:1112 at 192.168.1.202 SIP/2.0
Via: 
SIP/2.0/UDP 
192.168.0.22:5075;branch=z9hG4bK90ED68A545F54E0E9DD61D04BCC2E71C;uas-addr=192.168.1.202;rport=5075;received=192.168.0.22
Contact: 
<sip:softphone1 at 192.168.0.22:5075>
To: 
<sip:1112 at 192.168.1.202>;tag=as37940f45
From: 
<sip:softphone1 at 192.168.1.202>;tag=83CA16CF1
Call-ID: 
97494774C0DC4BFE99C55EC5C50F33AE
CSeq: 33 BYE
Content-Length: 
0

 
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 
192.168.0.22:5075;branch=z9hG4bK90ED68A545F54E0E9DD61D04BCC2E71C;uas-addr=192.168.1.202;rport=5075;received=192.168.0.22;received=192.168.0.22
From: 
<sip:softphone1 at 192.168.1.202>;tag=83CA16CF1
To: 
<sip:1112 at 192.168.1.202>;tag=as37940f45
Call-ID: 
97494774C0DC4BFE99C55EC5C50F33AE
CSeq: 33 BYE
User-Agent: Dcom Network 
Technology
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY
Contact: <sip:1112 at 192.168.1.202>
Content-Length: 
0
X-Asterisk-HangupCause: Normal Clearing
 
I would appreciate any help on this.
 
Regards,
Robert Vos_______________________________________________
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