Re: [SR-Users] [commercial announcement]

[Henning Westerholt]

Hello Dario,

this type of content is not appropriate for the sr-users lists, please post it 
here.

Please use our dedicated business list for this kind of commercial 
announcements.

https://lists.kamailio.org/cgi-bin/mailman/listinfo/business

Regards,

Henning

Am 17.01.20 um 06:58 schrieb Dario Vargas:
commercial announcement


___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users


--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com
Kamailio Merchandising - https://skalatan.de/merchandising/
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] UCaaS Platform

[Dario Vargas]

Take the LEAD in Unified Communications as a Service! We can help you get
started in the next Telco Disruptive Business.

White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to
offer all of the below products under your own brand -

Hosted Contact Center (CCaaS)
Hosted Unified Communications (UCaaS)
Hosted PBX (Full Feature Set)
SIP Trunk
Ai Chat Bots
Caller Profile Solutions (Display your clients' corporate identity on cell
phones) to improve phone answer ratio.
Video Surveillance (VSaaS integrated with AWS)
Web Conferencing
Virtual FAX
Business SMS

Software developed in the USA to have all the routing features required to
connect with US and Canadian carriers for inbound and outbound (LRN, LNP,
TF, SMS, Push, Video etc).
Use our existing contracts and pricing for telco services or BYOC (bring
your own carrier model).
Alternately we can provide you with introductions to the carriers of choice
below to get started -

Nuwave.com - Number 1 in TF 8XX Inbound
Voipinvite.com - 1000s of MPS and CPS for SMS and Voice Broadcast
Voxbone.com - World Wide DiDs - Low Monthly minimum
Peerlessnetwork.com  - Outbound / Inbound Business calling
Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum
volumes required
Verizon.com - Outbound and Inbound enterprise business calls - minimum
volumes required

Dario Vargas
i...@dnl-switch.com
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] http_client interface

[Fred Posner]

On 1/16/20 4:56 PM, Daniel-Constantin Mierla wrote:
> The libcurl seems to offer an option for that as well:
> 
>   - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html
> 
> So it can be a small patch to the module and support it in the future.
> 
> If you look for a solution on current stable, maybe one of the embedded
> scripting languages can help, you can execute inline a small script in
> Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the
> http client from lua to do more complex http api queries (e.g., with
> specific headers, ...).
> 
> Cheers,
> Daniel
> 

Thank you Daniel,

I implemented a work around (little proxy in go) but would love if the
option was added for the future.

Fred Posner
f...@qxork.com
https://qxork.com
Direct/SMS: +1 (336) 439-3733

Need Fred? Call Fred. 336-HEY-FRED
Matrix: @fred:matrix.lod.com

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT

[Richard Fuchs]

On 16/01/2020 12.29, Nuno Miguel Reis wrote:

Hi again.

Thanks for all the help and suggestions. I realized the issue happens 
if using kernel forwarding only. If I change rtpengine to start at 
userspace without the kernel module enabled everything works fine as 
expected.
Do you have any hints on why this could be happening with the kernel 
module?


I'm running rtpengine like this:

usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 
--listen-ng=127.0.0.1:2223  --tos=184 --sip-source


kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 
--listen-ng=127.0.0.1:2223  --tos=184 
--no-fallback --sip-source


The kernel module receives its instructions from the userspace daemon, 
so there's no reason there should be a difference. Can you post 1) logs 
and 2) the flags you use for your offers/answers?


Cheers

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Question about xhttp module kamailio 5.2.x

[Jimmy Rodley]

Hi,
I am on kamailio 5.2.4 and want to use the xhttp module for certain management 
operations.
The request involves some DB queries . The DB is remote and via ODBC and might 
have some delay.The documentation on the xhttp module states about latency 
http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the 
request is handled in an asynchronous manner. 
However, the JSONRPC-S module documentation says that this module implements 
the support for asynchronous RPC commands only for HTTP and HTTPS transports.

https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm1049416972

I am confused about the process under whose context the xttp request . Can 
there be a dedicated process to handle certain management requests which do 
some custom instructions like certain DB operations or initiate certain http 
requests ?
Thanks,Rupesh

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Trouble with 302 Redirect

[Jim Leahy]

Hi All,
   I'm having some issues trying to properly send a '302 Moved Temporarily'
reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486
status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server
when UAC2 is in DND.
Here are the pertinent parts of my config:

route {
# drop some things right away
route(DROP);

# perform sanity check
route(SANITYCHECK);

# handle registrations
if(is_method("REGISTER")) {
route(REGISTRATION);
exit;
}

# handle invites
if(is_method("INVITE")) {
route("INVITE");
exit;
}

route(RELAY);
}

route[INVITE] {
# add this proxy to the record-route so it stays in the loop
record_route();

# if we can find this user in the location DB...
if(lookup("location")) {
# relay the invite to the new dURI
t_on_failure("FOURDIGITFAIL");
t_on_reply("FOURDIGITREPLY");
route(RELAY);
# go back to the main route
return;
}

onreply_route[FOURDIGITREPLY] {
if(t_check_status("4[0-9]{2}")) {
t_reply("302", "Moved Temporarily");
}
}

I'm having issues with the t_reply in the onreply_route. It successfully
sends the reply, but I can't seem to set the Contact header properly. I'm
trying to set it to the location of the user's mailbox on the voicemail
sever (ie sip:1...@vmpbx.domain.com), but nothing I try works. I've tried:
 remove_hf("Contact");
 insert_hf("Contact: sip:1...@vmpbx.domain.com");
 t_reply("302", "Moved Temporarily");

But as I understand it, that just changes the header in the reply from
UAC2, not the reply I'm sending to UAC1

Then I tried:
 append_to_reply("Contact: sip:1...@vmpbx.domain.com" );
 t_reply("302", "Moved Temporarily");

But I can't use append_to_reply in an onreply block.
How can I change the Contact header in the reply that's sent? Am I just
going about this the wrong way? Any help would be appreciated. Thanks
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio stop to process incoming SIP traffic via TCP.

[adeykunov]

Daniel,

Got the same issue on 5.3.1 with openssl1.1, debian9.
After 3 working days of tests (about ~30-50 wss clients), suddenly we've got
a lot of connections stucked in CLOSE_WAIT state. Kamailio called
sig_alarm_abort() when we try to reboot.

Thanks,
Andrey





--
Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] http_client interface

[Daniel-Constantin Mierla]

The libcurl seems to offer an option for that as well:

  - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html

So it can be a small patch to the module and support it in the future.

If you look for a solution on current stable, maybe one of the embedded
scripting languages can help, you can execute inline a small script in
Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the
http client from lua to do more complex http api queries (e.g., with
specific headers, ...).

Cheers,
Daniel

On 13.01.20 17:00, Fred Posner wrote:
> I'd never had a reason to before, but on a deployment I need http
> messages to go out a specific interface. Curl has an --interface option,
> but I don't see how to do the same with http_client. Unless I'm missing
> something...
>
> Anyone ever do something like this?
>
-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - April 27-29, 2020, in Berlin -- 
www.kamailioworld.com


___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio/RTC dinner at Fosdem 2020

[Federico Santulli]

Hi Daniel,

We would like to join the dinner, 3 people from US.

Kind regards.

Federico Santulli

NHM - S.R.L.
Via Raffaello Sanzio, 88
81031 Aversa (CE)
Italy


> Il giorno 9 gen 2020, alle ore 09:52, Daniel-Constantin Mierla 
>  ha scritto:
> 
> Hello,
> 
> Fosdem 2020 is approaching, there will be a talk from Henning about
> Kamailio, many other friends and related projects are presenting in the
> RTC Devroom (Giacomo Vacca and Federico Cabiddu, Wazo, Asterisk, Homer,
> Janus, Jitsi, CGRateS, Linphone, ...):
> 
>   * https://fosdem.org/2020/schedule/track/real_time_communications/
> 
> I plan to be at the event and wondering if there are enough interested
> participants for having the traditional Kamailio/RTC dinner. If yes,
> Torrey will help to book a place to accommodate us (first choice could
> be the same restaurant like the last year, if available).
> 
> Reply if you want to join the dinner and say how many other people are
> joining you.
> 
> Cheers,
> Daniel
> 
> -- 
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio World Conference - April 27-29, 2020, in Berlin -- 
> www.kamailioworld.com
> 
> 
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users


___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT

[Nuno Miguel Reis]

Hi again.

Thanks for all the help and suggestions. I realized the issue happens if
using kernel forwarding only. If I change rtpengine to start at userspace
without the kernel module enabled everything works fine as expected.
Do you have any hints on why this could be happening with the kernel module?

I'm running rtpengine like this:

usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 --listen-ng=
127.0.0.1:2223 --tos=184 --sip-source

kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100
--listen-ng=127.0.0.1:2223 --tos=184 --no-fallback --sip-source


when using kernel module, I'm also setting this rules on chain INPUT under
iptables:

-A INPUT -p udp -m udp --dport 5000:16383 -j RTPENGINE --id 0
-A INPUT -p udp -m udp --dport 16384:32768 -j RTPENGINE --id 0
-A INPUT -p udp -m udp --dport 49152:65535 -j RTPENGINE --id 0


Looking forward to hear from you guys.

Regards,

--
Nuno Miguel Reis
Departamento de Engenharia Informática
Faculdade de Ciências e Tecnologia
Universidade de Coimbra


On Thu, Jan 16, 2020 at 1:23 PM Richard Fuchs  wrote:

> On 15/01/2020 13.39, Nuno Miguel Reis wrote:
>
> Hi guys.
>
> I'm replacing a environment where I was using kamailio + freeswitch by
> another where I'm adding rtpengine to the mix.
> One of the issues I'm having now is when I have a SIP Client behind NAT:
>
> When I send the INVITE from the SIP Client, the SDP is using the private
> LAN IP address + the advertised media port, let's admit it's
> 10.10.10.10:5000 when the call establishes with a public server running
> kamailio + rtpengine, the actual RTP arrives from the home router public IP
> on a natted port, let's admit it's 100.100.100.100:65100, event though,
> RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000.
> I'm using the '--sip-source' with RTPENGINE to make it use the received ip
> address instead of the private IP coming in the SDP but I'm not finding
> anything to make RTPENGINE adapt an start sending the RTP FLOW to the port
> where he starts receiving the RTP flow from, discarding the SDP media port
> information.
>
> Rtpengine does this automatically, unless the `asymmetric` flag is used.
> You should see a log message about a `confirmed peer address` as soon as
> the first RTP is received on a port.
>
> Cheers
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Get destination IP from forwarded invite

[Joel Serrano]

Hello,

If I understand correctly your email, you want the IP/Host of the
destination of the outgoing INVITE that is leaving a Kamailio box?

First of all, how are you forwarding the request? Are you using dispatcher
module?

Here are a couple of options:

1-
https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#du_-_destination_uri
(although
this depends on your config, it might not be set)

2- https://www.kamailio.org/wiki/cookbooks/5.3.x/core#onsend_route +
https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#sndto_name


If you can describe a little more how you are doing things we might be able
to give you more information.

Joel.


On Thu, Jan 16, 2020 at 3:53 AM CUMHUR KARAHAN <
cumhur.kara...@consultant.turkcell.com.tr> wrote:

> Hi all,
>
>
>
> I’m a newbie for Kamailio and working on a feature needs end-users’
> destination IP in the first transferred/forwarded invite message.
>
>
>
> Basically, our system/call flow is like:
>
> ClientA -> Kamailio1 -> Kamailio2 -> ClientB
>
>
>
> Invite from the clientA:
>
>
>
> *tls:212.252.34.165:32980 *
>
> *tls:172.21.193.100:5061 *
>
> *INVITE sip:905364167...@testsip.tvoip.turkcell.com.tr:443
>  SIP/2.0*
>
> *Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport*
>
> *From:  >;tag=WbINy0ypx*
>
> *To: sip:905364167...@testsip.tvoip.turkcell.com.tr
> *
>
> *CSeq: 20 INVITE*
>
> *…*
>
>
>
> Needed Ip (Client B’s IP 10.252.180.51 for this case) is in the INVITE row
> of the forwarded message.
>
>
>
> Invite to the Kamailio2 :
>
>
>
> *udp:172.21.193.100:5060 *
>
> *udp:172.21.193.98:5060 *
>
> *INVITE sip:905364167881@**10.252.180.51**:36945;transport=tls SIP/2.0*
>
> *Record-Route:
> *
>
> *Record-Route:
> *
>
> *Via: SIP/2.0/UDP
> gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f*
>
> *Via: SIP/2.0/TLS
> 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980*
>
> *From:  >;tag=WbINy0ypx*
>
> *To: sip:905364167...@testsip.tvoip.turkcell.com.tr
> *
>
> *CSeq: 21 INVITE*
>
> *…*
>
>
>
> When the first invite message receives, the dialog variables or sel(msg…)
> parsed values belong to this first received message, as far as I understand
> during the tests. However I need the forwarded invite message’s variables.
> Even after t_relay in RELAY route, it looks the message I have is the first
> one. So, I cannot reach this later IP in the forwarded invite message. As
> far as I understand, bear with me here please, this IP manipulation
> executes in t_relay function and in the C side of Kamailio. So, I cannot
> find a way to do that in the our native scripts.
>
>
>
> Is there any way to get it without diving C code’s abyss? Do you have any
> idea about it? I would appreciate it, if I can get any help or any clue.
>
>
>
> Thanks for all...
>
>
>
> S. Cumhur Karahan
>
>
>
>
> 
>
> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi
> tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi
> icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin
> icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz
> kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir.
>
> TURKCELL bu mesajin icerdigi bilgilerin doğruluğu veya eksiksiz oldugu
> konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne
> sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve
> saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen
> kisiye aittir ve TURKCELLin goruslerini yansitmayabilir
>
> Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir.
> --
>
> This e-mail and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they are
> addressed. If you are not the intended recipient you are hereby notified
> that any dissemination, forwarding, copying or use of any of the
> information is strictly prohibited, and the e-mail should immediately be
> deleted.
>
> TURKCELL makes no warranty as to the accuracy or completeness of any
> information contained in this message and hereby excludes any liability of
> any kind for the information contained therein or for the information
> transmission, reception, storage or use of such in any way whatsoever. The
> opinions expressed in this message belong to sender alone and may not
> necessarily reflect the opinions of TURKCELL.
>
> This e-mail has been scanned for all known computer viruses.
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
___
Kamailio (SER) - Users 

Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT

[Richard Fuchs]

On 15/01/2020 13.39, Nuno Miguel Reis wrote:

Hi guys.

I'm replacing a environment where I was using kamailio + freeswitch by 
another where I'm adding rtpengine to the mix.

One of the issues I'm having now is when I have a SIP Client behind NAT:

When I send the INVITE from the SIP Client, the SDP is using the 
private LAN IP address + the advertised media port, let's admit it's 
10.10.10.10:5000  when the call establishes 
with a public server running kamailio + rtpengine, the actual RTP 
arrives from the home router public IP on a natted port, let's admit 
it's 100.100.100.100:65100 , event 
though, RTPENGINE assumes that RTP stream is coming from 
100.100.100.100 :5000.I'm using the 
'--sip-source' with RTPENGINE to make it use the received ip address 
instead of the private IP coming in the SDP but I'm not finding 
anything to make RTPENGINE adapt an start sending the RTP FLOW to the 
port where he starts receiving the RTP flow from, discarding the SDP 
media port information.


Rtpengine does this automatically, unless the `asymmetric` flag is used. 
You should see a log message about a `confirmed peer address` as soon as 
the first RTP is received on a port.


Cheers

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Get destination IP from forwarded invite

[CUMHUR KARAHAN]

Hi all,

I'm a newbie for Kamailio and working on a feature needs end-users' destination 
IP in the first transferred/forwarded invite message.

Basically, our system/call flow is like:
ClientA -> Kamailio1 -> Kamailio2 -> ClientB

Invite from the clientA:

tls:212.252.34.165:32980
tls:172.21.193.100:5061
INVITE sip:905364167...@testsip.tvoip.turkcell.com.tr:443 SIP/2.0
Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport
From: ;tag=WbINy0ypx
To: sip:905364167...@testsip.tvoip.turkcell.com.tr
CSeq: 20 INVITE
...

Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of 
the forwarded message.

Invite to the Kamailio2 :

udp:172.21.193.100:5060
udp:172.21.193.98:5060
INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0
Record-Route: 
Record-Route: 

Via: SIP/2.0/UDP 
gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f
Via: SIP/2.0/TLS 
100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980
From: ;tag=WbINy0ypx
To: sip:905364167...@testsip.tvoip.turkcell.com.tr
CSeq: 21 INVITE
...

When the first invite message receives, the dialog variables or sel(msg...) 
parsed values belong to this first received message, as far as I understand 
during the tests. However I need the forwarded invite message's variables. Even 
after t_relay in RELAY route, it looks the message I have is the first one. So, 
I cannot reach this later IP in the forwarded invite message. As far as I 
understand, bear with me here please, this IP manipulation executes in t_relay 
function and in the C side of Kamailio. So, I cannot find a way to do that in 
the our native scripts.

Is there any way to get it without diving C code's abyss? Do you have any idea 
about it? I would appreciate it, if I can get any help or any clue.

Thanks for all...

S. Cumhur Karahan



[http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] 


Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi 
tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi 
icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin 
icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle 
yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir.

TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu 
konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne 
sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve 
saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye 
aittir ve TURKCELLin goruslerini yansitmayabilir

Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir.



This e-mail and any files transmitted with it are confidential and intended 
solely for the use of the individual or entity to whom they are addressed. If 
you are not the intended recipient you are hereby notified that any 
dissemination, forwarding, copying or use of any of the information is strictly 
prohibited, and the e-mail should immediately be deleted.

TURKCELL makes no warranty as to the accuracy or completeness of any 
information contained in this message and hereby excludes any liability of any 
kind for the information contained therein or for the information transmission, 
reception, storage or use of such in any way whatsoever. The opinions expressed 
in this message belong to sender alone and may not necessarily reflect the 
opinions of TURKCELL.

This e-mail has been scanned for all known computer viruses.
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT

[Sergio Charrua]

Olá Nuno,

I've similar setup, and everything works fine. I can share the setup, if
you wish.
Contact me directly.

Cheers,


*Sérgio Charrua*

*www.voip.pt *
Mobile: +351 91  631 11 44

Email : *sergio.char...@voip.pt *
Esta mensagem e quaisquer ficheiros anexos a ela são confidenciais e
destinam-se a uso exclusivo da pessoa ou entidade a quem são dirigidos. Se,
por lapso, recebeu este e-mail, não o deverá revelar, copiar ou distribuir
a terceiros. É favor alertar de imediato o remetente e eliminar a mensagem
juntamente com os documentos anexados.





On Wed, Jan 15, 2020 at 6:49 PM Nuno Miguel Reis 
wrote:

> Hi guys.
>
> I'm replacing a environment where I was using kamailio + freeswitch by
> another where I'm adding rtpengine to the mix.
> One of the issues I'm having now is when I have a SIP Client behind NAT:
>
> When I send the INVITE from the SIP Client, the SDP is using the private
> LAN IP address + the advertised media port, let's admit it's
> 10.10.10.10:5000 when the call establishes with a public server running
> kamailio + rtpengine, the actual RTP arrives from the home router public IP
> on a natted port, let's admit it's 100.100.100.100:65100, event though,
> RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000.
> I'm using the '--sip-source' with RTPENGINE to make it use the received ip
> address instead of the private IP coming in the SDP but I'm not finding
> anything to make RTPENGINE adapt an start sending the RTP FLOW to the port
> where he starts receiving the RTP flow from, discarding the SDP media port
> information.
>
> In my previous setup with freeswitch alone, freeswitch did this
> transparently. How do you guys solve this issue in RTPENGINE?
>
> Regards,
> --
> Nuno Miguel Reis
> Departamento de Engenharia Informática
> Faculdade de Ciências e Tecnologia
> Universidade de Coimbra
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio Slack

[David Villasmil]

Here

https://join.slack.com/t/kamailio/shared_invite/enQtNDY1NDc5MzMwNDIzLWJlNTFhYmYwOWE4ODcyZmM3OWYyMjEwYmRjMDg3YjU0NTU5YjNkZDdlNTM4NTFmYzYyMmE1YzQxNzNmZDAxNjA


On Wed, 15 Jan 2020 at 13:32, Abdirahman A. Osman 
wrote:

> Hello,
>
> Is the Kamailio Slack open for sr-users , if yes where can I find the
> invite link?
>
> Thanks
>
> Abdirahman
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
-- 
Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users