[SIPForum-discussion] Server Error 503

[Robert Vos]

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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