22 Mar
2012
22 Mar
'12
1:35 p.m.
I don;t think there is any difference in configuration working for asterisk
and not for anything esle..REGFWD is just a piece of code which can forward
registration to any IP:PORT where SIP is enabled and working.
If you carefully deploy that configurations keeping in mind only your
software (replace-all asterisk with portaone) won't that help !!
Like I said before its a starting point, you may need to put one or two
more things/modules/functions like Path() to have this thing working for
you.
On Wed, Mar 21, 2012 at 4:42 PM, ICI-Robert <robert(a)inteli-core.com> wrote:
I'll look however:
( a ) not working with Asterisk - working with a commercial Softswitch
(PortaOne)
( b ) need NAT support
Sent from my iPhone 4S
On Mar 21, 2012, at 12:23 AM, SamyGo <govoiper(a)gmail.com> wrote:
Hi Robert,
There can't be any simpler start then using the
configurations(kamailio.cfg file) mentioned in this blog-post by Daniel.
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
You can see a *route[REGFWD]* and in that route you can relay the
REGISTRATIONS to your desired SIP-Server!!
This is by far the easiest and simplest start I've found without any
headache of going deeper into the SIP packets and techniques.
See this is the concerned route.
# Forward REGISTER to *YOUR_SIP_REGISTRAR_SERVER*
route[REGFWD] {
if(!is_method("REGISTER"))
{
return;
}
$var(rip) = $sel(cfg_get.asterisk.bindip); #CHANGE THE
asterisk.bindip/asterisk.bindport on the top
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" +
$sel(cfg_get.asterisk.bindport);
$uac_req(furi)="sip:" + $au + "@" + $var(rip);
$uac_req(turi)="sip:" + $au + "@" + $var(rip);
$uac_req(hdrs)="Contact: <sip:" + $au + "@"
+ $sel(cfg_get.kamailio.bindip)
+ ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; # Change
Kamailio.bindip/port as well.
if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) +
"\r\n";
else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) +
"\r\n";
uac_req_send();}#!endif
Make sure you do DB tables and Variables adjustments accordingly.
Regards,
Sammy.
NOTE: Thanks Sir Daniel for these wonderful and very helpful blogposts.
On Wed, Mar 21, 2012 at 3:41 AM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hello,
On 3/20/12 10:36 PM, Robert wrote:
I've spent a significant amount of time now jacking with configurations…
Maybe I should re-state what I'm trying to do…
I'm trying to establish an SBC / UpperRegistration SBC…
The goal – is to listen on a port other than 5060 and relay the
registration to the actual proxy.
I.e. IP Phone ---> Kamailio (port 8080) ----> Sip proxy (port 5060)
I've tried all combinations of rewritehostport / t_replicate etc…
perhaps t_relay_to(...) is enough, depending on what the r-uri address
contain (should be registrar server address, if K on port 8080 is used as
outbound proxy).
You have to use path module and the sip proxy on 5060 has to understand
the path headers (it is not clear that proxy on 5060 is Kamailio, if it is,
then it support path).
The best I get is an attempt to register followed by an unauthorized
with request for digest auth..
I don't want to store any credentials in Kamailio – I just want to act
as an offset port for users who cannot reach 5060…
Forwarding registrations is fine, no need to authenticate them on K port
8080. Perhaps providing the relevant part of your config along with a ngrep
SIP trace on K port 8080 will help to give you the right directions to get
it work.
Cheers,
Daniel
OpenSBC was working for this – however the registrations were timing
out and not renewing
From: Reda Aouad <reda.aouad(a)gmail.com>
Reply-To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER)
- Users Mailing List" <sr-users(a)lists.sip-router.org>
Date: Fri, 9 Mar 2012 17:16:14 +0100
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users
Mailing List" <sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Proxy Registrations offset port
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(a)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(a)gmail.com>
Reply-To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER)
- Users Mailing List" <sr-users(a)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(a)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(a)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(a)gmail.com>
Reply-To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER)
- Users Mailing List" <sr-users(a)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(a)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(a)inteli-core.com> wrote:
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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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--
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin,
Germanyhttp://www.asipto.com/index.php/kamailio-advanced-training/
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