I have tried following rule but somehow opensips challenging it from
authentication
route[3]{
if ( $ru =~ "^sip:011[0-9]*@") {
rewritehostport("65.65.65.65:5065");
xlog("Redirecting to SIP Provider... $ru\n");
exit;
};
}
U 198.198.198.1
rewritehost() sucessfully work with UAC. But As I know
1) It statless function
2) It read only string argumetns, and do not read variables
2014-08-21 14:43 GMT+04:00 Satish Patel :
> I will give it a try again today, can you please make sure my t_relay()
> syntax is correct?
>
> So t_relay will
I will give it a try again today, can you please make sure my t_relay() syntax
is correct?
So t_relay will rewrite my host past right and send call to trunk.
While ago I was using rewritehost() function but I think it's not working with
UAC registrant module.
Sent from my iPhone
On Aug 21,
On Thursday 21 August 2014 05:56:46 Satish Patel wrote:
> if ( $rU =~ "sip:1[0-9]@*") {
Try $ru instead, $rU only contains the dialled "number". So
$ru =~ "sip:1[0-9]@*"
or
$rU =~ "1[0-9]"
But note the regexp, that only matches the exact numbers 10 to 19, if you are
trying to mat
This is what i did but its not working, getting error SIP/2.0 403
Forbidden, it is thinking number i am dialing is local and checking in
local DB . by the way SIP provider Trunk is already registered using UAC
module. I am using Multi-domain setup.
# do lookup with method filtering
if (!
On Wednesday 20 August 2014 14:54:42 Satish Patel wrote:
> I am new in Kamailio so could you please give me code example how to use
> t_relay() to forward traffic to Provide, I know how to use rewritehost()
> but i never use t_relay() function
Well, my guess is your routing ends with t_relay().
B
My example don`t help you. you must read about t_relay there.
t_relay() is not central thing. I must stop your attension at SIP invite
that goes to provider. t_relay simple to use- just customise your INVITE
and call t_relay() from your route.
http://kamailio.org/docs/modules/devel/modules/tm.htm
I am new in Kamailio so could you please give me code example how to use
t_relay() to forward traffic to Provide, I know how to use rewritehost()
but i never use t_relay() function
On Wed, Aug 20, 2014 at 8:22 AM, Yuriy Gorlichenko
wrote:
> You can use t_relay() too. One thing that you need - t
You can use t_relay() too. One thing that you need - to have right packet,
that will be relays to Provider. I have multiple providers and manually
change packets that will send to provider.
2014-08-20 15:57 GMT+04:00 Satish Patel :
> Great! I registered remote Trunk using UAC module. so now i ca
Great! I registered remote Trunk using UAC module. so now i can just use
following function to forward my call right?
rewritehost()
On Wed, Aug 20, 2014 at 12:33 AM, Yuriy Gorlichenko
wrote:
> Use UAC module for this
> 20.08.2014 7:40 пользователь "Satish Patel"
> написал:
>
>> We have setup
Use UAC module for this
20.08.2014 7:40 пользователь "Satish Patel" написал:
> We have setup Kamailio front and SIP Proxy, Now i want to Trunk it with
> other SIP provide they gave me IP, Username/Password. How do i configure
> username/password on Kamailio SIP Proxy?
>
>
We have setup Kamailio front and SIP Proxy, Now i want to Trunk it with
other SIP provide they gave me IP, Username/Password. How do i configure
username/password on Kamailio SIP Proxy?
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Hi,
So when I use the uac_reg_request_to, for some reason it sends an invite
and not a register?? This means I'm not getting a 407 back to the use
sac_auth in the failure route.
Any ideas?
Thanks
Keith
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SIP Express Router (SER) and Kamailio (OpenSER
On Thursday 19 June 2014 10:42:49 Keith wrote:
> Thanks Daniel, however don't I need to send a register to get the 401 back?
> If so how do I do this bit?
Your message sounded like need to authenticate outbound calls, registrations
will direct inbound calls to your machine. Never tried it, but fo
Hi,
Thanks Daniel, however don't I need to send a register to get the 401 back?
If so how do I do this bit?
Thanks
Keith
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http://lists.sip-router.or
On Wednesday 18 June 2014 10:42:02 Keith wrote:
> I have a carrier who requires SIP authentication on calls, can anyone point
> me in the right direction? Normally in Asterisk or Freeswitch you can
> authenticate a peer, but can't see how to do this in Kamailio.
You need the uac module and handle
Hi,
I have a carrier who requires SIP authentication on calls, can anyone point
me in the right direction? Normally in Asterisk or Freeswitch you can
authenticate a peer, but can't see how to do this in Kamailio.
Thanks
Keith
___
SIP Express Router (SER
On 08 May 2014, at 17:20, Joli Martinez wrote:
> Hello.
>
> What I would like to know is where do I setup a sip trunk in kamailio. Are
> there any examples as to how to set one up?
The term "sip trunk" means at least X*Y^Z different things. It's not a
technichal definition of something.
Th
Hello.
What I would like to know is where do I setup a sip trunk in kamailio. Are
there any examples as to how to set one up?
Sent from my iPhone
> On May 8, 2014, at 3:05 AM, Daniel-Constantin Mierla
> wrote:
>
> Hello,
>
> you have to explain in more details what that means for you 'sip
Hello,
you have to explain in more details what that means for you 'sip trunk
registered to kamailio' for more specific hints.
Otherwise, kamailio doesn't care of who is doing the registration as
long as it presents the credentials based on username and password.
Cheers,
Daniel
On 07/05/14
Hello,
I am new to Kamailio. I need to get a SIP trunk registered to Kamailio. Can
someone explain how to set this up.
Thanks
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http://lists.sip
Hello,
I am setting up Kamailio as class4 softswitch for wholesale minutes. I am
trying to figure out how to add a SIP trunk to kamailio.
thanks,
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> On 09.05.2011 22:44, JR Richardson wrote:
> > Would there be any usage examples of dialog module? I'm not sure if I
> > really need profiles and [values] and when to set and unset. A push
> > in the right direction would be helpful.
>
> Hi,
> here is an example I've used for incoming and outgo
On 09.05.2011 22:44, JR Richardson wrote:
Would there be any usage examples of dialog module? I'm not sure if I
really need profiles and [values] and when to set and unset. A push
in the right direction would be helpful.
Hi,
here is an example I've used for incoming and outgoing calls trackin
>> I recently interconnected to an upstream carrier using Kamailio 3.0,
>> working fine. We have configure 2 SIP trunks for failover/redundnacy.
>> I'm using dispatcher module to round robin calls to the carrier. I
>> wanted to monitor trunk usage between us. I was reading in the devel
>> 3.2
On 05/06/2011 03:43 PM, JR Richardson wrote:
Hi All,
I recently interconnected to an upstream carrier using Kamailio 3.0,
working fine. We have configure 2 SIP trunks for failover/redundnacy.
I'm using dispatcher module to round robin calls to the carrier. I
wanted to monitor trunk usage bet
Hi All,
I recently interconnected to an upstream carrier using Kamailio 3.0,
working fine. We have configure 2 SIP trunks for failover/redundnacy.
I'm using dispatcher module to round robin calls to the carrier. I
wanted to monitor trunk usage between us. I was reading in the devel
3.2 about s
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