Hi David,
Between 1 and 2, you need to decide what should be the relation between
opensips and freeswitch.
If you consider opensips "serving" the SIP domain of FS (and have the
aliasing), then you need to configure the logic for handling
registrations into OpenSIPS - instead of locally processing the
REGISTER's, it should simply forward them to FS. Also, all the calls
targeting the served / local domains should be relayed to FS.
If you want opensips to see FS as an external entity (and consider the
SIP domain of FS an something remote), you need to re-configure the call
routing logic - to recognize the SIP domain used by FS and to accept to
do relay for it (as a trusted peer).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.02.2016 10:44, David Wafula wrote:
---------- Forwarded message ----------
From: *David Wafula* <[email protected] <mailto:[email protected]>>
Date: Tue, Feb 2, 2016 at 10:43 AM
Subject: Re: [OpenSIPS-Users] Opensips as outbound proxy to Freeswitch
To: Bogdan-Andrei Iancu <[email protected] <mailto:[email protected]>>
Yes, the opensips.cfg was indeed messed up. i was tinkering with it.
Anyway, i regenerated a clean one, put in the the alias, and happily
got registrations going through. And one way audio. Except that i
think am missing the big picture.
My goal is to have Opensips front Freeswitch:
Client <-> OpenSips <-> Freeswitch
To act as outbound proxy at this stage. As i learn more about it, i
hope to get into NATing, loadbalancing etc. Am afraid i have not
succeeded in achieving this goal unfortunately, yet, though i think
am very close. This is what i have done.
1. Added the alias as advised. The registrations fail. Then when i
create users in opensips, registration work perfectly, and one of the
users can call the other. Except that in this case it appears all is
happening in Opensips and freeswitch is not in the loop here. Infact,
i verified, that obviously registrationds are done on Opensips and not
freeswitch.
2. I removed the alias, deleted the users from opensips. This time the
registrations goes through successfully onto freeswitch. Very happy,
as this is what i want. But now obviously INVITES fail with 403 Rely
forbidden.
It appears to me, there is more configuration that i must do, beyond
setting alias, to get Opensips to act as outboundproxy to freeswitch.
I need registrations and calls to happen on Freeswitch, but via Opensips.
So after again searching the internet, i can see that may be
something like dynamic routing can be done
(http://www.taitclarridge.com/techlog/2012/02/opensips-dynamic-routing.html),
or i can see whole lot of other way of doing it
(http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration).
What is the recommended, possibly simplest way to do this: to get
opensips proxy registrations, invites , messaging to freeswitch.
Many thanks.
On Mon, Feb 1, 2016 at 5:30 PM, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>> wrote:
David,
Your opensips cfg is bogus - the test for the presence of username
in RURI should not be done for a REGISTER requests (as they do not
have a username part).
Where have you taken the script from ? have you changed it by
yourself ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.02.2016 15:06, David Wafula wrote:
Thank you Bogdan. Ok i made the alias change, and now am getting
"Address Incomplete" reply. Please see the log below, am not
sure what could be now causing this:
REQUEST
============
02-01 14:58:53.144: I/System.out(14726): REGISTER
sip:192.168.0.46 SIP/2.0
02-01 14:58:53.144: I/System.out(14726): Via: SIP/2.0/UDP
10.0.1.175:30802;rport;branch=z9hG4bKPjGfzHRCN4S.QyGFhBBv9njI9Amj2oU7ko
02-01 14:58:53.144: I/System.out(14726): Route:
<sip:192.168.4.248;lr>
02-01 14:58:53.144: I/System.out(14726): Max-Forwards: 70
02-01 14:58:53.144: I/System.out(14726): From:
<sip:[email protected]>;tag=eNVm.45KTH3OkXAjSikua1cwsZ.XdeRE
02-01 14:58:53.144: I/System.out(14726): To: <sip:[email protected]>
02-01 14:58:53.144: I/System.out(14726): Call-ID:
9TVBTDFtUPz05axr9qvyynhFWhJetb6m
02-01 14:58:53.144: I/System.out(14726): CSeq: 64838 REGISTER
02-01 14:58:53.144: I/System.out(14726): User-Agent: Pjsua2
Android 2.4.5
02-01 14:58:53.144: I/System.out(14726): Contact:
<sip:[email protected]:34708;ob>
02-01 14:58:53.144: I/System.out(14726): Expires: 300
02-01 14:58:53.144: I/System.out(14726): Allow: PRACK, INVITE,
ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
02-01 14:58:53.144: I/System.out(14726): Content-Length: 0
02-01 14:58:53.144: I/System.out(14726):
02-01 14:58:53.144: I/System.out(14726): --end msg--
REPLY
=============
02-01 14:58:53.585: I/System.out(14726): SIP/2.0 484 Address
Incomplete
02-01 14:58:53.585: I/System.out(14726): Via: SIP/2.0/UDP
10.0.1.175:30802;received=10.0.1.175;rport=30802;branch=z9hG4bKPjGfzHRCN4S.QyGFhBBv9njI9Amj2oU7ko
02-01 14:58:53.585: I/System.out(14726): From:
<sip:[email protected]>;tag=eNVm.45KTH3OkXAjSikua1cwsZ.XdeRE
02-01 14:58:53.585: I/System.out(14726): To:
<sip:[email protected]>;tag=a0a925d2eca49498ea7382b7b1f63f38.d365
02-01 14:58:53.585: I/System.out(14726): Call-ID:
9TVBTDFtUPz05axr9qvyynhFWhJetb6m
02-01 14:58:53.585: I/System.out(14726): CSeq: 64838 REGISTER
02-01 14:58:53.585: I/System.out(14726): Server: OpenSIPS (2.1.2
(x86_64/linux))
02-01 14:58:53.585: I/System.out(14726): Content-Length: 0
On Mon, Feb 1, 2016 at 1:07 PM, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>> wrote:
Hi David,
I see. The problem is all the SIP traffic contains references
to this 192.168.0.46, but opensips has no idea how to handle
consider this IP (like local or foreign domain).
It should handle it as local domain, so add in your cfg:
alias="192.168.0.46"
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.02.2016 12:21, David Wafula wrote:
Hi Bogdan,
OpenSips is listening on 192.168.4.248, which running in
totally different network from freeswitch. Freeswitch is
running on 192.168.0.46.
And, my test user [email protected]
<mailto:[email protected]>, is created on freeswitch. I do
not have any local users on opensips at this stage.
Regards.
On Mon, Feb 1, 2016 at 11:48 AM, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>> wrote:
Hi David,
The "relay forbidden" case happens when neither the FROM
URI, nor the request URI contain a SIP domain
served/local to OpenSIPS. In your case I see that the
INVITE has in FROM and RURI the 192.168.0.46 SIP domain.
Is your OpenSIPS actually listening on this IP ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 30.01.2016 17:25, David Wafula wrote:
Hello list,
Am quite new to opensips. So i want opensips to act as
an outbound proxy to freeswitch. I followed the
tutorial on the opensips site and set up a running
opensips instance (vanilla). I too have a running
freeswitch instance.
so, on my softphone, when i set outbound proxy as
opensips, am able to register the phone successfully
to freeswitch via opensips. But when i attempt to make
a call, i get:
send_reply("403","Rely forbidden");
--
David Wafula
--
David Wafula
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