[SIPForum-discussion] Server Error 503

[Mark Holloway]

One thing that can cause 503 Service Unavailable is when media ports are not
available for the bearer traffic.   What if you setup two softphones, does
it work as expected?  If so, there could be something wrong with the
"device" that you are calling.  A wireshark capture would be useful to see
the SDP.

 

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Murali Subramanian
Sent: Friday, December 12, 2008 8:44 PM
To: panna108 at yahoo.com
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Server Error 503

 

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 <mailto:sip%3A1112 at 192.168.1.202>  SIP/2.0
Via: SIP/2.0/UDP
192.168.0.22:5075;branch=z9hG4bK90ED68A545F54E0E9DD61D04BCC2E71C;uas-addr=19
2.168.1.202;rport=5075;received=192.168.0.22
Contact: <sip:softphone1 at 192.168.0.22:5075
<http://sip:softphone1@192.168.0.22:5075/> >
To: <sip:1112 at 192.168.1.202 <mailto:sip%3A1112 at 192.168.1.202>
>;tag=as37940f45
From: <sip:softphone1 at 192.168.1.202 <mailto: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=19
2.168.1.202;rport=5075;received=192.168.0.22;received=192.168.0.22
From: <sip:softphone1 at 192.168.1.202 <mailto:sip%3Asoftphone1 at 192.168.1.202>
>;tag=83CA16CF1
To: <sip:1112 at 192.168.1.202 <mailto: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 <mailto: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

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-- 
With thanks & regards,
Murali Subramanian.,

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