[SIPForum-discussion] Server Error 503

[Murali Subramanian]

Usually 503 error message will get display when the user is not at all
logged in or SIP account is not registered yet for the user.
Can you check it ?

Regards,
Murali

2008/12/12 amirul ahsan <panna108 at yahoo.com>

>   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%3A1112 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 <sip%3A1112 at 192.168.1.202>>;tag=as37940f45
> From: <sip:softphone1 at 192.168.1.202 <sip%3Asoftphone1 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 <sip%3Asoftphone1 at 192.168.1.202>
> >;tag=83CA16CF1
> To: <sip:1112 at 192.168.1.202 <sip%3A1112 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 <sip%3A1112 at 192.168.1.202>>
> Content-Length: 0
> X-Asterisk-HangupCause: Normal Clearing
>
> I would appreciate any help on this.
>
> Regards,
> Robert Vos
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>


-- 
With thanks & regards,
Murali Subramanian.,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20081213/4c2c1d8e/attachment-0002.html>