No wonder your signallin is not reaching the final client.

I didnt't suggest to throw OpenSips. NAT handling is delicate woth proxies.

Start by looking at some wireshark dump to see what's wrong.

Reda

On 9 mars 2012, at 16:57, Robert <robert@inteli-core.com> wrote:

Two different tests I completed:

Test #1 – it showed the opensips server

Test #2 – it showed the client's public ip address

The issue is not as much RTP – but I'm not getting SIP signaling back to the client.. Which leads me to believe the NAT problem. I'm not even getting ringing…

Maybe I should go ahead and install Kamailio and throw out the OpenSIPs stuff…

From: Reda Aouad <reda.aouad@gmail.com>
Reply-To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users@lists.sip-router.org>
Date: Fri, 9 Mar 2012 16:41:57 +0100
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users@lists.sip-router.org>
Subject: Re: [SR-Users] Proxy Registrations offset port

Kamailio's default conf is really a good place to start, you just have to disable the authentication, by commenting the corresponding route.

You can start by looking at a wireshark/tcpdump dump on your PortaOne server to know where the rtp stream is going, and then troubleshoot accordingly..

On your PortaOne server, what is the location of registered users? Is it their IP@, or that of OpenSips server?

Reda

On 9 mars 2012, at 16:28, Robert <robert@inteli-core.com> wrote:

Thank you so much for answering.. The folks at the other place don't want to help.

We have a PortaOne SIP platform… It does everything… It's listening on port 5060 – public ip..

We want to deploy Kamailio or some derivative of SER to listen on PUBLIC port – other than 5060 – for our customers who cannot reach 5060.

Using a very simple rewritehostport statement – I was successful in getting the registration to pass through. I.e. Registered my Aastra and X-Lite to port 8080 and saw registration in PortaOne.

I do not want K/SER to do actual auth / registration – I.e. I do not care who tries to use it – I want final registration to be PortaOne… That's why I think the simple rewritehostport worked well. OpenSIPS was not doing any auth – just simply passing it on.

Our PortaOne was handling RTP Proxy – so during an outbound call I was getting 2 way audio – no problem.

However when an inbound call would come in from PortaOne – it was being delivered to OpenSIPS and looks like Opensips was reporting 401 Unauthorized.

The Aastra / X-Lite are NAT… It's possible that either I need to look at nathlper closer – or that I misconfigured something somewhere.

I guess I'm not fluent enough in the configs and options to get a good config.

From: Reda Aouad <reda.aouad@gmail.com>
Reply-To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users@lists.sip-router.org>
Date: Fri, 9 Mar 2012 15:45:14 +0100
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users@lists.sip-router.org>
Subject: Re: [SR-Users] Proxy Registrations offset port

Hi Robert,

I had the same feeling about O***sips community.....

I have a similar setup. A proxy that listens on port other than 5060, forwarding to other servers on 5060.

If you can better define better your problem with your setup, maybe we can help you better.

Is your server listening only on one port 8080? Is your server behind NAT or does it have a public IP @ on the interface it's listening on? What is the exact problem with inbound calls? Are you doing record-route? Do you forward only registrations, or are your handling invites as well? What server listens on port 5060? Another Kamailio server? Does it handle clients behind NAT?

You should know that Kamailio is a SIP server, not a PBX/sofswitch/SBC as Asterisk or Freeswitch.

If you define better your problem, and send me your config file (by private email if you like), I may look at it.

Reda

On Fri, Mar 9, 2012 at 15:32, Robert <robert@inteli-core.com> wrote:

I've been playing with OpenSIPS and am willing to change over if it means I'll get a more informed answer from this community..

I need to set up a proxy forwarding server that can listen on a port other than 5060 – I.e. 8080 and forward the registration requests to another Voip platform we have that is running on 5060.

I've been able to this under OpenSIPS but I'm having issues with inbound calls. It could be a NAT issue.

Basically listen on 8080 – forward any and all registrations to x:5060 — which I've done using rewritehostport

Anyone have any good example configs for this? I'm a n00b at this OpenSER stuff – but have been around asterisk / freeswitch / portaone for a while.

Thanks in advance

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