603 decline response for invite message

Mahima · Post by **Mahima** » Mon Aug 10, 2009 1:03 am

1. Brekeke Product Name and version:

2. Java version:

3. OS type and the version:

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:

Hi All

When sending the following invite message;

INVITE SIP:6711@192.168.1.102 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.117:5060
Max-Forwards: 70
From:
To:
Call-ID: 14462@192.168.1.117
User-Agent: TotalView 1.0
CSeq: 3 INVITE
Contact:
Content-Type: application/sdp
Content-Length: 0

(where 1.6711 is callee's no IP:192.168.1.110
2.6710 is caller's no IP:192.168.1.117 both are properly registered in brekeke server IP:192.168.1.102).

to the SIP Server a 603 decline message is received:

response is:

SIP/2.0 603 Decline
Via: SIP/2.0/TCP 192.168.1.117:5060;received=192.168.1.102
From:
To: ;tag=b15ca8215s
Call-ID: 8379@192.168.1.117
CSeq: 3 INVITE
Server: Brekeke SIP Server rev.286 Evaluation
Content-Length: 0

Any help for a newbie would be appreciated

Thanks&Regards
Mahima

hope · Post by **hope** » Mon Aug 10, 2009 12:45 pm

is there any dialplan applied when you make the call?
is the TCP-handling set on at BSS/configuration/sip/ ?
is the BSS local sip port changed?

Mahima · Post by **Mahima** » Mon Aug 10, 2009 10:29 pm

1.dialPlan is set to

Matching Patterns: $request=^INVITE
To=sip:6711@

Deploy Patterns: To=sip:6711@192.168.1.110

2. TCP Handling is on in BSS\Configuration\SIP

3.Local Port is set to 5060 in BSS\Configuration\SIP

hope · Post by **hope** » Tue Aug 11, 2009 4:14 pm

if not use dialplan, is there any difference?

Mahima · Post by **Mahima** » Tue Aug 11, 2009 11:09 pm

No there is no change in response message(if dial Plan is not set).

hope · Post by **hope** » Wed Aug 12, 2009 2:27 pm

Is there any other dialplan related to call? if yes, please disable them.

can the call ring the callee when caller and callee use UDP only?

Can callee side handle TCP? and did you set "UDP failover" and " TCP-handling" on at is the at BSS/configuration/sip/ TCP?

from packets, who returned 603 first? callee or BSS?

Mahima · Post by **Mahima** » Thu Aug 13, 2009 4:56 am

There is no other Dial plan related to this call.

At the callee side i am using Adore softphone(which according to the the doc. supports both tcp and udp ).

"UDP failove" is set to "off" and "TCP-handling" is set to "on"

603 decline error is return from server(192.168.1.102)

When i make a call from one sip adore phone to another through "brekeke server" with the same dial plan it works fine

hope · Post by **hope** » Thu Aug 13, 2009 4:53 pm

When i make a call from one sip adore phone to another through "brekeke server" with the same dial plan it works fine

Does the call use tcp?

can you call from Adore softphone to the totalview 6710?

please set "UDP failove" to "on" and "TCP-handling" to "on" ,
save the settings and restart sip server

and create a dialplan as
Matching Patterns:
$request=^INVITE
To=sip:6711@

Deploy Patterns:
$transport=tcp
To=sip:6711@

click "apply rules"

register both caller and callee to the BSS and test the call.

Mahima · Post by **Mahima** » Mon Aug 17, 2009 4:46 am

Call between both adore sip phone is happening through 'UDP'

Does Brekeke SIP Server , Version 2.3.6.0 Evaluation Supports 'TCP'

'*TCP inactive in Personal/Academic Editions ' is mentioned in TCP Handling of 'BSS/Configuration/SIP '

hope · Post by **hope** » Fri Aug 21, 2009 10:32 am

Version 2.3.6.0 Evaluation Supports 'TCP'
maybe Adore phone's problem?
did you try by setting "UDP failover" to "on" and "TCP-handling" to "on" and register both caller and callee to the BSS and test the call?

Andrey · Post by **Andrey** » Fri Aug 21, 2009 3:40 pm

Mahima,

INVITE SIP:6711@192.168.1.102 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.117:5060
Max-Forwards: 70
From:
To:
Call-ID: 14462@192.168.1.117
User-Agent: TotalView 1.0
CSeq: 3 INVITE
Contact:
Content-Type: application/sdp
Content-Length: 0

Is it an actual packet?
It seems there are no From/To/Contact values... did you remove them?

Mahima · Post by **Mahima** » Mon Aug 24, 2009 5:09 am

Hi

I have constructed a new Invite Message :

INVITE SIP:6711@192.168.1.117 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK25504;rport..User-Agent: TotalView 1.0
From: <sip:6710@192.168.1.117>
To: <SIP:6711@192.168.1.117>
Call-ID: 16295
CSeq: 2 INVITE
Contact: <sip:6710@192.168.1.102:5060>
Content-Type: application/sdp
Allow: INVITE,INFO,ACK,CANCEL,BYE,OPTION,NOTIFY,MESSAGE,PING,REFER
Max-Forwards: 70
Content-Length: 0

here:
(192.168.1.117:5060 is Server Machine(Brekeke Server),
192.168.1.102:5060 is Client Machine Where i am trying to run my program ,extension On is 6710,
192.168.1.110 is another Client Machine where a sip Phone is configured ,extension no. is 6711.)

When i am sending invite message to sip Phone through Brekeke server i am getting decline 603 error.

SIP/2.**UnReg**line..
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK28751;rport=50195..
From: <sip:6710@192.168.1.117>..
To: <SIP:6711@192.168.1.117>;tag=b8a33d053s..
Call-ID: 27731..
CSeq: 2 INVITE..
Server: Brekeke SIP Server rev.286 Evaluation..Content-Length: 0....

UDP Socket Connection is getting use to send invite Message .

lakeview · Post by **lakeview** » Mon Aug 24, 2009 6:50 pm

Are you using own made SIP client??

Mahima · Post by **Mahima** » Tue Aug 25, 2009 5:14 am

ya i using own made client ,

Trace taken by sniffer on server machine

Invite message sent from client:

INVITE SIP:6711@192.168.1.117 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK25504;rport..User-Agent: TotalView 1.0
From: <sip:6710@192.168.1.117>
To: <SIP:6711@192.168.1.117>
Call-ID: 16295
CSeq: 2 INVITE
Contact: <sip:6710@192.168.1.102:5060>
Content-Type: application/sdp
Allow: INVITE,INFO,ACK,CANCEL,BYE,OPTION,NOTIFY,MESSAGE,PING,REFER
Max-Forwards: 70
Content-Length: 0

here:
(192.168.1.117:5060 is Server Machine(Brekeke Server),
192.168.1.102:5060 is Client Machine Where i am trying to run my program ,extension On is 6710,
192.168.1.110 is another Client Machine where a sip Phone is configured ,extension no. is 6711.)

While sending invite message to sip Phone through Brekeke server Using my client the response is coming as decline 603 error.

SIP/2.**UnReg**line..
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK28751;rport=50195..
From: <sip:6710@192.168.1.117>..
To: <SIP:6711@192.168.1.117>;tag=b8a33d053s..
Call-ID: 27731..
CSeq: 2 INVITE..
Server: Brekeke SIP Server rev.286 Evaluation..Content-Length: 0....

UDP Socket Connection is getting use to send invite Message .[/b]

lakeview · Post by **lakeview** » Thu Aug 27, 2009 12:47 pm

Are you using any SDK for making your SIP client??

If you are a SIP client developer, contact Brekeke's support team and ask the interoperability test.