Hi all.
I need to route calls to different counties to different pstn gateways.
But in the config file, i can only set one pstn gw ip:port in pstn section. How
should i do it? Can i use carrierroute module to do this? If yes, can anyone
paste a config sample?
many thanks
Jesse
hello David,
I want the user to register to the kamailio server which is closest to the
user. Each user(aka subscriber) has a unique domain, but my kamailio servers
have 10 different domains which are their public ip. Let's say, based on user's
ip, the closest kamailio server's domain is
Hi everyone,
I'm new to kamailio and rtpproxy and the problem's been bothering me for
days. Any help would highly appreciated.
My network topology is as follows:
kamailio5.0 and rtpproxy2.0 are running on the same server. The server
is behind NAT, its private ip is
n 06.06.17 10:48, 赵国杰 wrote:
Hi Daniel,
I solved the problem by add "transport=udp". Thanks anyway.
Cheers
Jesse
At 2017-06-06 15:36:22, "Daniel-Constantin Mierla" <mico...@gmail.com> wrote:
Hello,
if you set r-uri to a UDP address, then you can just us
Hello,
The uac_replace_from works like a charm. However, preferred/asserted
identity are not supported by some of the pstn providers.
Thanks
Jesse
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Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
and may not work with
all clients.
There are a bunch of considerations - for example, what is “closest” ? I don’t
expect you to answer that to me, but, you will need to have a clear definition
before you can embark on this project.
On 9/06/2017, at 6:30 PM, 赵国杰 <zhaoguojie2...@163.com> wro
I'm very sorry. I just found I mis-configured the advertise ip. It's very
stupid of me. sorry
在 2017-06-27 18:37:11，"赵国杰" <zhaoguojie2...@163.com> 写道：
I just found that when client register to server B, client will append another
Contact header after Authorization hea
.
At 2017-09-18 18:05:46, "Daniel Tryba" <d.tr...@pocos.nl> wrote:
>On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
>> I want the caller to play a short audio(like "the number your are
>> calling is busy") when the callee declines the call. How can i do
quot;Sergey Safarov" <s.safa...@gmail.com> wrote:
You can add this example to dialplan and make test
ср, 20 сент. 2017 г. в 10:14, 赵国杰 <zhaoguojie2...@163.com>:
Hello Sergey,
I installed freeswitch, what should i do n
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided
Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2
(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2...@163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($rU=="12345") {
route(RELAY);
exit;
}
}
in freeswitch/conf/dialplan/default.xml, i added
Jurijs
On Fri, Sep 22, 2017 at 10:54 AM, 赵国杰 <zhaoguojie2...@163.com> wrote:
Hi guy.
sorry for the confusion. I'll try t
f/vars.xml and change the default_password."
2) You are calling into Freeswitch with encryption on and probably of this your
call is failing, maybe you can try first to try without SRTP and if it works,
then you can try to make it work with SRTP
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at
Hi guys,
I want the caller to play a short audio(like "the number your are calling
is busy") when the callee declines the call. How can i do that?
Best Regards
Jesse___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
Hello,
In a standard sip flow, the call goes like: sip user A --> kamailio -->
pstn --> landline user B. However, when user A has a bad internet access, the
audio is broken. So what I want is to let sip user A send a invite to kamailio
first, then kamailio send invite to user A and B's
returns, the final destiation is determined, say
C. Then send the INVITE to C.
However, after append_branch(), then original INVITE is sent to B anyway.
How do I block the A to B INVITE?
Thanks
At 2017-12-01 15:13:10, "赵国杰" <zhaoguojie2...@163.com> wrote:
Hello Sammy,
Again, it depends on why you want it?
Regards,
Sammy
On Wed, Nov 29, 2017 at 11:13 PM, Brandon Armstead <bran...@cryy.com> wrote:
if(is_method(“INVITE”)){ .
sl_send_reply(“183”, “Ringing”);
}
On Wed, Nov 29, 2017 at 6:44 PM 赵国杰
ompt/announcement pre-encoded with the
makeann command from the RTPproxy distribution."
Am 15.11.2017 07:59 schrieb "赵国杰" <zhaoguojie2...@163.com>:
Hello guys,
I googled the problem and came up with the following solution
request_route {
...
if (rtpproxy_of
API query in parallel. As
I see it you're trying to append_branch and then doing a t_newtran->HTTP ASYNC
query, causing the whole blockage and that explains if you remove the last 4
line it works.
Regards,
Sammy
On Fri, Dec 1, 2017 at 3:14 AM, 赵国杰 <zhaoguojie2...@163.com> wrote:
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