Re: [SR-Users] port forwarding

[Vladyslav Zakhozhai]

Hi Anrzej,

Right question is 50% of solution. Unfortunately the question is wrong.
Almost nothing is clear.

You can register your phones (clients) and you even can start or receive
calls but there is an issue with audio/video? Is it correct?

In this case this problem is not related to kamailio. It is related to
pbx/media server that is possibly behind kamailio. But maybe there is no
any pbx behind?

Without additional configuration audio/video traffic will go end-to-end
which is a problem with external clients with no public IP.

And yes, you are right about 5060/udp. But everything depends on your
configuration which is not clear. Maybe you need additionaly 5060/tcp,
5061/tcp (TLS).

But ports range for media traffic depends on media server/pbx/rtp proxy
configuration and nobody can answer this question but you.

I hope this tips will help you a little bit.


On Tue, Oct 17, 2017 at 1:40 AM, Andrzej Kaczmarczyk 
wrote:

> Hello,
>
> This is probably trivial, but I'm quite new in kamailio and trying to
> handle it for my personal purposes (so - no trainings, no support etc :) )
> - for few (up to 20) users and rare connections, more to learn something
> than to have production system. No plans to go outside my own server.
>
> I have LAN. One of machines in this LAN is server with kamailio onboard.
> It works, but only with local hostname (user@hostname).
> Now I want to go outside (with clients).
>
> As temporarty solution I have only dynamic DNS (no-ip.org) and I didn't
> expect it working. But tried with IP adress (u...@wan.ip.add.ress).
>
> And... I don't know which ports to redirect on router.
> Now there are redirected 5060 and 1-3 (maybe it is not necessary
> but first I want it working) - taken from some on-line info.
>
> I can:
> - log in to all my accounts,
>
> - depends on client - call to another user and in some cases I can receive
> the call (so it is probably issue on my clients, because it is possible to
> make a call at all).
>
> But I cannot transmit any data: connections are completely silent in any
> case and I cannot start video - although cameras are findable.
>
>
>
> Which ports in fact are necessary?
>
> I want audio and video communication, presence info and probably nothing
> more. Which ports should I redirect?
>
> --
> Pozdrawiam
> Andrzej Kaczmarczyk
>
>
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>



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Re: [SR-Users] call disconnects after 32 seconds

[Vladyslav Zakhozhai]

Hi Adesh

Nobody can help you but you :)

It seems that you have issues with NATed endpoints. Is it correct? Is there
NAT?
In this case nathelper module may help you.

Resolution of this configuration issue depends on your infrastructure (how
thins are connected to each other).

On Wed, Sep 27, 2017 at 8:10 AM, Adesh Pandey 
wrote:

> Hi Alex,
> Could you please help me to fix that or inspect that?
>
> On Wed, Sep 27, 2017 at 10:20 AM Alex Balashov 
> wrote:
>
>> That sounds a lot like an end-to-end ACK that isn't making it from
>> caller to callee.
>>
>> On Wed, Sep 27, 2017 at 04:46:45AM +, Adesh Pandey wrote:
>>
>> > Hi,
>> > I have set up the Kamailio as load balancer using dispatcher module but
>> all
>> > the calls get disconnected after 30 seconds (call is forwarded to
>> Asterisk
>> > Box and answered as well). I have used example configuration from the
>> below
>> > URL:
>> >
>> > http://kamailio.org/docs/modules/4.3.x/modules/
>> dispatcher.html#idp51014772
>> >
>> > Please help me to understand what is wrong and what shall I do it to
>> fix.
>> >
>> > Thank You!!
>> > Adesh P.
>> > --
>> >
>> > Adesh Pandey
>> >
>> > Sr. Software Developer
>> > [image: phone]
>> >
>> > +91 92129 92129 Ext: 56 <+91%2092129%2092129>
>> > [image: moible]
>> >
>> > +91 8527384897 <+91%2085273%2084897>
>> > [image: Profile Pic]
>> >
>> > [image: Facebook] [image:
>> Twitter]
>> > [image: Twitter]
>> > [image: Linkedin]
>> > [image:
>> Google+]
>> > 
>> > [image: Logo] 
>>
>> > ___
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>> > sr-users@lists.kamailio.org
>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Alex Balashov | Principal | Evariste Systems LLC
>>
>> Tel: +1-706-510-6800 <(706)%20510-6800> / +1-800-250-5920
>> <(800)%20250-5920> (toll-free)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
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>>
> --
>
> Adesh Pandey
>
> Sr. Software Developer
> [image: phone]
>
> +91 92129 92129 Ext: 56 <+91%2092129%2092129>
> [image: moible]
>
> +91 8527384897 <+91%2085273%2084897>
> [image: Profile Pic]
>
> [image: Facebook] [image:
> Twitter] [image: Twitter]
> [image: Linkedin]
> [image: Google+]
> 
> [image: Logo] 
>
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>
>


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Re: [SR-Users] Kamailio / Asterisk and multidomain

[Vladyslav Zakhozhai]

Hi,

I think that this issue is more specific for asterisk than kamailio.

Maybe configuration of outbound proxy may help here but I am not sure.

On Tue, Aug 22, 2017 at 3:50 PM, Cyrille Demaret  wrote:

> Hi,
>
> I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or
> more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime
> Integration" tutorials, I would like to let Kamailio handle registrations,
> calls between users and other basic functionalities. Asterisk will only
> handle advanced features like voicemail, advanced dialplan configuration,.
> My problem is that It needs to be multidomain.
>
> My network will be as follow with 2 network interfaces for Kamailio and a
> private lan between Kamailio and Asterisk :
>
> Public interface ip 1.2.3.4 > [Kamailio] <- Private interface
> 192.168.100.10 <> 192.168.100.11 [Asterisk]
>
> When I send a call to asterisk, the domain is sent in the from field of the
> INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP
> domain using the ${SIPDOMAIN} variable). My problem is when I need to send
> a
> call back to Kamailio for example to reach another user of the domain.
>
> I'm using Asterisk 14 with PJSIP with the following config :
>
> [kamailio]
> type=endpoint
> transport=transport-udp
> context=from-kamailio
> disallow=all
> allow=ulaw
> aors=kamailio
>
> [kamailio]
> type=aor
> contact=sip:192.168.100.10:5060
>
> [kamailio]
> type=identify
> endpoint=kamailio
> match=192.168.100.10
>
> If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio
> /sip:2...@testdomain.com", Asterisk contact directly testcomain.com without
> going through the local IP of my Kamailio.
>
> I can't send a domain to Kamailio in the INVITE request.
>
> Does anyone can help me on this or maybe simply tell me that I'm not going
> to the good direction? :)
>
> Thank you,
>
> Cyrille
>
>
>
>
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>



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