[SR-Users] Kamailio NAT AWS EC2

[Benjamin Marty]

Hi,

I am new to Kamailio and coming a bit from the Asterisk side.

My goal is to set up a bunch of users on a Kamailio Instance on AWS
EC2 and get them able to call each other and talk to each other over
the Internet without P2P media stream.

Is rtpproxy the tool that enables that? As far my knowledge is right
Kamailio isn't able to do any sort of RTP handling.


Regards,

Benjamin
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[SR-Users] is t_flush_flags() really needed?

[Juha Heinanen]

Juha Heinanen writes:

> While testing xflags, i noticed that a regular flag that I set AFTER
> calling t_newtrans() stays set in onreply_route even when I do not
> call t_flush_flags().

I made the same test with xflags and they do require t_flush_xflags()
call if an xflag is set after t_newtrans();  So the behavior is not the
same with flags and xflags.

This is confusing.  The flags should behave the same way and I would
prefer the flags way in order to avoid the flush call.

I added sr-dev to this thread since it now deals also the new feature.

-- Juha

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Re: [SR-Users] Strange auth_challenge behaivor on 407

[Yuriy Gorlichenko]

Hi thx. Yep. NAT handling was moved to branch.
So rport called from this handler and never called fpr the 1-st invite.

On Mon, Mar 26, 2018, 16:05 Aqs Younas  wrote:

> Make sure you are calling force_rport.
>
> Br, Aqs.
>
>
> On 26 March 2018 at 13:57, Yuriy Gorlichenko  wrote:
>
>> Hi
>> I'm using version: kamailio 5.1.2
>>
>> I have strange issue with auth_challenge
>>
>> kamialio works as NAT handler. When Register comes kamailio handles nat
>> and save original ip:port at the received field in location table
>>
>> I found that when local port in contact is different than external port
>> reply 407 on INVITE comes to externalip (from received) : internalport(from
>> contact)
>> And never reaches destination
>>
>> for exmaple
>> Register comes from
>> 192.168.1.10:50555
>>
>> Kamailio saves at the Received
>> 12.1.1.12:1030
>>
>> but when invite comes form 12.1.1.12:1030 kamailio sends reply to
>> 12.1.1.12:50555
>>
>> i don't think that this is an issue in kamailio source code but i can
>> find where i can configure it
>>
>>
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
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Re: [SR-Users] How should i configure kamailio connection freepbx?

[Seanchann Zhou]

this is a big question.so,  i suggest you  read some guide first.

2018-03-26 15:18 GMT+08:00 星昊通技术 :

> hello all:
> I install kamailio 5.1 and siremis 5.1, I want to configure kamailio
> connection freepbx, how should i configure kamailio vai siremis GUI?
>
> Thanks.
>
> --
> Best regards!
> alex
> Hiastar 深圳星昊通科技
> *Sangoma in APAC, 潮流网络IP话机，IPPBX*
> Asterisk cards
> , SBC
> 
> , NetBorder VOIP Gateway
> ,
> Transcoding
> , 潮流网络IP话机
> 
> QQ：2933996052
> website:www.hiastar.com|www.freepbx.cn, contact number: 13574109204
> 关注微信公众帐号：Asterisk-CN
>
>
>
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Re: [SR-Users] To tags in 407 challenges

[Alex Balashov]

On Mon, Mar 26, 2018 at 10:27:35PM +0200, Henning Westerholt wrote:

> I would think this is normal UAC/UAS behavior for SIP request handling:
> 
> RFC 3261, sect. 8.2.6.2:
> "However, if the To
> header field in the request did not contain a tag, the URI in the To
> header field in the response MUST equal the URI in the To header
> field; additionally, the UAS MUST add a tag to the To header field in
> the response (with the exception of the 100 (Trying) response, in
> which a tag MAY be present). This serves to identify the UAS that is
> responding, possibly resulting in a component of a dialog ID. "

Honestly, in ten years of doing SIP, this is the first time I am
confronting this question. I was accustomed to the notion that To tags
are only added to a reply when it is desirable to form an early dialog
(in the case of a non-100 1xx reply). But the passage you quote seems to
settle it.

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] kamailio with media server

[Mack Hendricks]

Hey Atux,

Can you give a little more detail on your use case?  Are you looking for 
Kamailio to:

- route requests to a media server for playing announcements 
- proxy requests between your endpoints and your media server(s)
- distribute calls to your carriers based on some logic

The answer may be all three - this will help us point you in the right 
direction.

Mack Hendricks / Head of Support / dOpenSource
web: http://dopensource.com 
support: +888-907-2085
dSIPRouter  - GUI focused on implementing Kamailio to 
provide SIP Trunking and PBX Hosting Services

> On Mar 26, 2018, at 9:45 AM, Atux Atux  wrote:
> 
> Hi. New to the area of Kamailio.
> i am did install in debian kamailio with rtpproxy and i have created 3 users 
> 1000-10002 (one for each jitsi user) they talk nice between them.
> I have followed this tutorial 
> http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour 
>  and 
> in less than 5 minutes i had my accounts registered.
> i would like to have a media server so the users could hears announcements 
> and stuff. At the end i would like to have kamailio as a test lab PBX where i 
> could connect my SIP trunk providers and my users to route calls.
> 
> 
> Is there any guide on how to setup a media server and the services, please?
> 
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Re: [SR-Users] To tags in 407 challenges

[Henning Westerholt]

Am Montag, 26. März 2018, 16:55:12 CEST schrieb Alex Balashov:
> This may be more a SIP question than a Kamailio question, but I am
> trying to figure out why 407 challenges from Kamailio contain To tags.
> 
> I suppose I've not run across it before, and I can't find an RFC-based
> rationale. A negative reply to an invite transaction without an
> intermediate early dialog-forming reply (e.g. 183 with To tag) does not
> form a dialog, nor takes place within a context in which a dialog was
> created. So what gives?

Hello Alex,

its indeed that common that it also included in the example RFC 3665, section 
3.3. and other.

There is a transaction relationship between the INVITE and the 407. RFC 5057 
mandates that the 407 only refers to the transaction, not to any dialog state. 
But you are right, if there is no provisional response, there is no (early) 
dialog at this state. 

But its also included in the 401 (for REGISTERs).

I would think this is normal UAC/UAS behavior for SIP request handling:

RFC 3261, sect. 8.2.6.2:
"However, if the To
header field in the request did not contain a tag, the URI in the To
header field in the response MUST equal the URI in the To header
field; additionally, the UAS MUST add a tag to the To header field in
the response (with the exception of the 100 (Trying) response, in
which a tag MAY be present). This serves to identify the UAS that is
responding, possibly resulting in a component of a dialog ID. "

Or I am wrong?

Henning

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[SR-Users] To tags in 407 challenges

[Alex Balashov]

Hi,

This may be more a SIP question than a Kamailio question, but I am
trying to figure out why 407 challenges from Kamailio contain To tags. 

I suppose I've not run across it before, and I can't find an RFC-based
rationale. A negative reply to an invite transaction without an
intermediate early dialog-forming reply (e.g. 183 with To tag) does not
form a dialog, nor takes place within a context in which a dialog was
created. So what gives?

Insights appreciated!

-- Alex

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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[SR-Users] kamailio with media server

[Atux Atux]

Hi. New to the area of Kamailio.
i am did install in debian kamailio with rtpproxy and i have created 3
users 1000-10002 (one for each jitsi user) they talk nice between them.
I have followed this tutorial
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour and
in less than 5 minutes i had my accounts registered.
i would like to have a media server so the users could hears announcements
and stuff. At the end i would like to have kamailio as a test lab PBX where
i could connect my SIP trunk providers and my users to route calls.


Is there any guide on how to setup a media server and the services, please?
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Re: [SR-Users] Help on setting UP SIP PRoxy

[Mack Hendricks]

Hey Ayub,

I would take it one step at a time.  Kamailio is a toolkit with a bunch of 
modules.  I find it useful to first figure out what features you need in a 
proxy.   This includes understanding what are you trying to protect your 
backend media servers from, problems you are currently having and/or how are 
you trying to scale your backend servers.  

Assuming you are using Asterisk or FreeSwitch as your backend media servers, 
here are some questions that might help derive the features you want:

- Are the backend servers being overwhelmed by REGISTER requests?
- Are you providing the proxy for phone end points, PBX’s or both
- What’s your High Availability Requirements?  Are the backend media services 
replicated so it doesn’t mater what server gets the original INVITE requests?
- Are you handing registrations for 1 SIP domain or multiple SIP domains?
- Do you want to maintain CDR’s at the proxy or leave it to the backend media 
servers

Mack Hendricks / Head of Support / dOpenSource
web: http://dopensource.com 
support: +888-907-2085
dSIPRouter  - GUI focused on implementing Kamailio to 
provide SIP Trunking and PBX Hosting Services


> On Mar 26, 2018, at 1:26 AM, Mohammed Ayub  wrote:
> 
> HI,
> 
> Myself Ayub from Phonology, Bangalore, India.
> 
> We are trying to set up an SIP proxy server using Kamailio with Siremis as 
> web interface.
> 
> Have installed and configured kamailio and siremis but need help in config if 
> sip domain , users  and etc...
> 
> Please revert.
> 
> Regards
> Ayub
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Re: [SR-Users] Strange auth_challenge behaivor on 407

[Aqs Younas]

Make sure you are calling force_rport.

Br, Aqs.


On 26 March 2018 at 13:57, Yuriy Gorlichenko  wrote:

> Hi
> I'm using version: kamailio 5.1.2
>
> I have strange issue with auth_challenge
>
> kamialio works as NAT handler. When Register comes kamailio handles nat
> and save original ip:port at the received field in location table
>
> I found that when local port in contact is different than external port
> reply 407 on INVITE comes to externalip (from received) : internalport(from
> contact)
> And never reaches destination
>
> for exmaple
> Register comes from
> 192.168.1.10:50555
>
> Kamailio saves at the Received
> 12.1.1.12:1030
>
> but when invite comes form 12.1.1.12:1030 kamailio sends reply to
> 12.1.1.12:50555
>
> i don't think that this is an issue in kamailio source code but i can find
> where i can configure it
>
>
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[SR-Users] Strange auth_challenge behaivor on 407

[Yuriy Gorlichenko]

Hi
I'm using version: kamailio 5.1.2

I have strange issue with auth_challenge

kamialio works as NAT handler. When Register comes kamailio handles nat and
save original ip:port at the received field in location table

I found that when local port in contact is different than external port
reply 407 on INVITE comes to externalip (from received) : internalport(from
contact)
And never reaches destination

for exmaple
Register comes from
192.168.1.10:50555

Kamailio saves at the Received
12.1.1.12:1030

but when invite comes form 12.1.1.12:1030 kamailio sends reply to
12.1.1.12:50555

i don't think that this is an issue in kamailio source code but i can find
where i can configure it
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[SR-Users] Help on setting UP SIP PRoxy

[Mohammed Ayub]

HI,

Myself Ayub from Phonology, Bangalore, India.

We are trying to set up an SIP proxy server using Kamailio with Siremis as
web interface.

Have installed and configured kamailio and siremis but need help in config
if sip domain , users  and etc...

Please revert.

Regards
Ayub
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[SR-Users] How should i configure kamailio connection freepbx?

[??????????]

hello all:
I install kamailio 5.1 and siremis 5.1, I want to configure kamailio connection 
freepbx, how should i configure kamailio vai siremis GUI?


Thanks.



--
Best regards!
alex
Hiastar ??
Sangoma in APAC, IP??IPPBX
Asterisk cards, SBC, NetBorder VOIP Gateway, Transcoding, IP
QQ??2933996052
website:www.hiastar.com|www.freepbx.cn, contact number: 13574109204
??Asterisk-CN___
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