The reason for my question is that we are 2 mvno's who wants to share a
single sip mvno trunk - and want to have our individual customers routed
according to who owns them - but also have the dispatcher functionallity
for "load-balancing".
On Tue, Dec 15, 2015 at 1:23 PM, Michael
Hello everyone,
I know kamailio can be configured into almsot everything for SIP - and that
the dispatcher function is already build for dispatching between multiple
sbc's etc...
Is it possible to use the dispatcher for routing between multiple sbc's
accroding to called subscriber?
So I can make
Yes, the endpoints communicate with different IPs.
Client 1 with sip1.my-domain.com and client 2 with sip2.my-domain.com.
On Thu, Nov 26, 2015 at 4:51 PM, Daniel Tryba wrote:
> On Thursday 26 November 2015 16:41:52 Michael Nielsen wrote:
> > I'm using Amason's Route 53 DNS s
e set usrloc db_mode = 3 so there's no cache in location of users in
memory...
But NAT seems to be an issue even though WITH_NAT is enabled :(
On Thu, Nov 26, 2015 at 4:37 PM, Daniel Tryba wrote:
> On Thursday 26 November 2015 16:00:09 Michael Nielsen wrote:
> > I've setup a
And I do have usrloc db_mode = 3.
On Thu, Nov 26, 2015 at 4:00 PM, Michael Nielsen
wrote:
> I've installed Kamailio on 2 different servers.
> They share the same PostgreSQL database for users etc.
>
> I've setup a single domain, sip.my-domain.com, with latency respo
I've installed Kamailio on 2 different servers.
They share the same PostgreSQL database for users etc.
I've setup a single domain, sip.my-domain.com, with latency responds to the
2 servers.
So a user will always land on the server with the fastest responds in a
respective area of the world.
Howe
I've installed RTPProxy and Kamailio and would like to have them start up
correct on boot.
First RTPProxy and then Kamailio.
How to accomplish this?
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
I would end up with 2 or more PBX's...
On Fri, Sep 4, 2015 at 2:47 PM, Daniel Tryba wrote:
> On Friday 04 September 2015 14:30:34 Michael Nielsen wrote:
> > My initial thought is to, right after route(pstn); to put:
> > route(DISPATCHER);
> >
> > and then h
xit;
}
Would that do it?
On Fri, Sep 4, 2015 at 1:11 PM, Michael Nielsen
wrote:
> I'm running the standard Debian 7 Kamailio 4.3 package.
>
> I've added the following to my kamailio.cfg:
>
> #!define WITH_MYSQL
>
> #!define WITH_AUTH
>
> #!define WITH_USR
I'm running the standard Debian 7 Kamailio 4.3 package.
I've added the following to my kamailio.cfg:
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_NAT
#!define WITH_TLS
#!define WITH_ANTIFLOOD
After authentication, nat etc. is done is there an easy way to tran
Yes.
On Fri, Sep 4, 2015 at 10:19 AM, Daniel Tryba wrote:
> On Thursday 03 September 2015 19:19:44 Michael Nielsen wrote:
> > Is it something to do with codec or could it be something else?
>
> Are you using an rtp proxy?
>
> ___
This is my setup:
Kamailio -> FreeSWITCH (for voicemail)
...SIP gateway for connecting GSM
So every call is routed directly via Kamailio - between SIP clients and
in/out from the GSM space.
Testing with a cellular phone as one client and X-Lite on Mac OS X as the
other - everything works.
Testi
Hello everyone...
I'm testing various ways of setting up Kamailio and FreeSWITCH (as
VAS-handler).
I'm struggling a bit on what Kamailio should do and FreeSWITCH should do.
My basic setup is to handle voicemails, internal calls between subscribers
and outgoing calls to a SIP gateway to GSM.
My
Well, I'm using a PSTN (sip) gateway to call out from, and what to change
my caller id here.
On Wed, Sep 2, 2015 at 3:55 PM, Alex Balashov
wrote:
> On 09/02/2015 09:30 AM, Michael Nielsen wrote:
>
> The same way I can change called party id with $rU, is there also a way
>>
The same way I can change called party id with $rU, is there also a way to
change the caller id of the person making the call?
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-route
ubscriber table, it will be loaded from database whent he
> authentication is done (i.e., in the same query with the password).
>
> Cheers,
> Daniel
>
>
> On 01/09/15 17:37, Michael Nielsen wrote:
>
> What if I just want to lookup the country code IF the country code is no
luck...
On Tuesday, September 1, 2015, Daniel Tryba wrote:
> On Tuesday 01 September 2015 16:43:42 Michael Nielsen wrote:
> > If I want to load a user specific country code, using avp, how can I do
> > this?
> >
> > I've added kamctl avp add MY-SUBSCRIBER countr
;t seem to give me +44?
Thank you once again.
On Tue, Sep 1, 2015 at 3:59 PM, Daniel Tryba wrote:
> On Tuesday 01 September 2015 13:45:16 Michael Nielsen wrote:
> > Is it possible to, if a subscriber has not entered either + or 00, to
> > append the value from country_code of
I've added a custom column in my subscriber table in MySQL - called
country_code.
This contains +44 etc. for subscribers.
Is it possible to, if a subscriber has not entered either + or 00, to
append the value from country_code of the specific subscriber to the
dialled number?
And if the user has
based proxy for authentication.
>
> Cheers,
> Daniel
>
>
> On 01/09/15 09:50, Michael Nielsen wrote:
>
> So sorry, but do you know of any tutorial using the auc module for
> authenticating against a PSTN gateway?
>
>
> On Mon, Aug 31, 2015 at 8:57 AM, Michael
}
# user location service
route(LOCATION);
-
Is it correctly understood that it first checks if the subscriber is local,
if not, it routes to PSTN otherwise it routes locally?
Or does it do both local and PSTN route if the subscriber is local?
On Mon, Aug 31, 2015
So sorry, but do you know of any tutorial using the auc module for
authenticating against a PSTN gateway?
On Mon, Aug 31, 2015 at 8:57 AM, Michael Nielsen
wrote:
> Perfect, thanks.
>
> On Sun, Aug 30, 2015 at 9:37 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
ble does it return.
>> What I'd suggest is to check if call is coming not from PSTN (if it comes
>> from PSTN - it's for sure must be routed to PBX) and if TRUE, then first
>> send call to PBX and if answer is not 180/183 200 etc. (you can catch that
>> in a sp
I'm trying to use a PSTN gateway with Kamailio for outbound - and incoming
- calls.
Everything seems to be routed correctly, except the PSTN gateway denies the
call due to missing authentication.
How do I add credentials for my PSTN gateway in a secure way?
Best regards,
Michael
gt; On Fri, Aug 21, 2015 at 8:04 AM, Michael Nielsen
> wrote:
>
>> I've installed from Debian package?
>>
>> On Fri, Aug 21, 2015 at 1:57 PM, E. Schmidbauer
>> wrote:
>>
>>> You need to install kamailio's rtpproxy module.
>>
I've installed from Debian package?
On Fri, Aug 21, 2015 at 1:57 PM, E. Schmidbauer
wrote:
> You need to install kamailio's rtpproxy module.
> if you compiled kamailio make sure you have it listed like:
> make include_modules="rtpproxy"
>
>
>
> On Fri,
Hello everyone,
I'm new to Kamailio and are trying this tutorial:
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
I've installed everything and are trying to launch Kamailio, but it fails
with the following error:
-
systemctl status kamailio.service
● kamailio.servic
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