URI / Requester Ports

1. Brekeke Product Name and version: Brekeke SIP Server , Version 2.1.6.6 Standard

2. Java version: 1.5.0_14

3. OS type and the version: Windows 2003 Entreprise

4. UA (phone), gateway or other hardware/software involved: Grandstream HT-486

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : 9

6. Your problem: Noticed that sometimes UA registers with different ports for SIP-URI and REQUESTER. That makes me losing incoming traffic from PSTN (DID). Any help will be deeply appreciated.

Are UA clients behind NAT?
Disable STUN at UA settings.

Yes, behind NAT. But, no STUN configured.

go to server's GUI -> [Configuration] -> [SIP] -> [NAT traversal].
And select [Add 'rport' parameter (Receive)] = "true".

Try it.

It did not work out...

Try the latest beta version..
http://www.brekeke.com/download/download_sip_beta.php

It has improved NAT detection feature.

No success after upgrade. Worst: now ALL DID's can't receive calls. BSS sends error 486 (busy here) for ALL incoming calls. Please help.

Are you using Brekeke PBX ?

Yes, I am.

Do you have a user in the PBX?
If not, you need to create certain users in the PBX.

We have both: PBX and BSS users.

Hi damatta,

Can you paste 486 error response packet here?
Or, can you see"User-Agent" header in the response packet?

Incoming traffic issue solved (NAT problem, not).Thanks.

Try this DialPlan rule.
=========================================
[Matching Patterns]
$request = ^REGISTER
$globaladdr( Contact ) = true
$addr = (.+)
$globaladdr( "%1" ) = true
$port = (.+)

[Deploy Patterns]
&register.contact.addrport = %1:%2
&register.contact.nat = true
$continue = true
=========================================

After applying this Dial Plan, some lines became RED. So, I assume there are synthax errors.

Matching Patterns:

$globaladdr(Contact)=true
$globaladdr("%1")=true

Deploy Patterns:

&register.contact.addrport=%1:%2
&register.contact.nat=true

Hi all,

Harold's DialPlan seems no problem.

>> After applying this Dial Plan, some lines became RED. So, I assume there are synthax errors.

Don't worry. you can use these variables.
These variables are new from version 2.2 beta.. so they are not listed in the DialPlan's spell checker.

Fine now

Wow..

Does Harold's DialPlan solve the issue??

Well, partially... Signalling issue was solved (can receive calls) but not media (no audio).

Did you set "RTP-Relay=on"??
If not, try it.

Yes, RTP Relay setting is ON and RTP relay (UA on this machine) is AUTO.