Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Daniel Tryba]

On Tuesday 23 June 2015 21:22:26 Javier Aristizábal wrote:
> On my cfg file I have this for the asterisk registration:
> 
> route[REGFWD] {
>   if(!is_method("REGISTER"))
>   {
> return;
>   }
>   $var(rip) = $sel(cfg_get.asterisk.bindip);
>   $uac_req(method)="REGISTER";
...
>   uac_req_send();
> }

I'm confused, why are you creating a new register request instead of 
forwarding the REGISTER to the asterisk server? And where are you handling 
this uac_send()? 

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] linking math.h while compiling a module

[Shouvanik Chakrabarti]

Hello,
I have developed a Kamailio module to handle group messaging and am
using certain functions from the libm.so library. Compilation happens fine,
but Kamailio subsequently fails to start because the math.h symbols are
unlinked/undefined. Could someone please suggest a workaround for this.

Thanks,
Shouvanik.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

I see. How can forwarding the REGISTER to asterisk? Apparently I don't
handling the uac_send

On Wed, Jun 24, 2015 at 10:15 AM, Daniel Tryba  wrote:

> On Tuesday 23 June 2015 21:22:26 Javier Aristizábal wrote:
> > On my cfg file I have this for the asterisk registration:
> >
> > route[REGFWD] {
> >   if(!is_method("REGISTER"))
> >   {
> > return;
> >   }
> >   $var(rip) = $sel(cfg_get.asterisk.bindip);
> >   $uac_req(method)="REGISTER";
> ...
> >   uac_req_send();
> > }
>
> I'm confused, why are you creating a new register request instead of
> forwarding the REGISTER to the asterisk server? And where are you handling
> this uac_send()?
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

[Vitaliy Aleksandrov]

Hi,

I would also add that if you see partial packets you can try to remove 
any transport protocol (udp/tcp) and port filters. It will help if you 
are dealing with IP fragmentation. Otherwise sniffer won't catch IP 
fragments since they don't have transport level headers.


Best Regards,
Vitaliy

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Hi Daniel, I think that in fact I am doing the forwarding register to
asterisk with the uac_req_send();

On Wed, Jun 24, 2015 at 10:58 AM, Javier Aristizábal <
javieraristiza...@gmail.com> wrote:

> I see. How can forwarding the REGISTER to asterisk? Apparently I don't
> handling the uac_send
>
> On Wed, Jun 24, 2015 at 10:15 AM, Daniel Tryba  wrote:
>
>> On Tuesday 23 June 2015 21:22:26 Javier Aristizábal wrote:
>> > On my cfg file I have this for the asterisk registration:
>> >
>> > route[REGFWD] {
>> >   if(!is_method("REGISTER"))
>> >   {
>> > return;
>> >   }
>> >   $var(rip) = $sel(cfg_get.asterisk.bindip);
>> >   $uac_req(method)="REGISTER";
>> ...
>> >   uac_req_send();
>> > }
>>
>> I'm confused, why are you creating a new register request instead of
>> forwarding the REGISTER to the asterisk server? And where are you handling
>> this uac_send()?
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> --
> Javier Aristizábal
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

[Daniel-Constantin Mierla]

On 23/06/15 19:06, Andrey Utkin wrote:
> 2015-06-23 18:49 GMT+03:00 Daniel-Constantin Mierla :
>> Have you grabbed the sip trace on client side to see what it is
>> receiving? Are the clients reporting errors?
> Yes, see this https://gist.github.com/krieger-od/c9fe6ea4bb64fac82cda
> this is taken on Linux box running Jitsi desktop app.
> It doesn't report anything, or I haven't seen the log, ut just doesn't
> show the incoming call.

Jisti has options to enable logging -- search on the web how to enable
and where is located the log file.

It would be interesting to see if it receives any packet and prints any
error message there.

Have you tried with tls? That will rule out eventual packet mangling on
the way done by providers/routers ALGs.

Cheers,
Daniel
>
>
>> If you have a snom phone, you can easily see the received sip packets
>> via web interface. Perhaps the desktop phones will have also some logs
>> printing what is happening that can be accessed easily.
>>
>> Eventually you can try to run a kamailio locally, near the client, using
>> it as an intermediate proxy between the phone and the main sip server.
>>
>> The timestamps I checked in previous traces were not following the sip
>> retransmissions intervals (0.5sec, 1sec, 2sec, ...), a clear indication
>> that it is not kamailio transaction layer doing retransmissions.
>
>> As I said before, ngrep is not a source to trust when dealing with large
>> packets. Also, it can happen that it prints the same packet twice.
> But what sniffer should I try instead of ngrep to have more details
> and confidence?
>
> Also I guess you mean this to be an issue of Linux kernel on any side,
> or possibly of routing hardware somewhere in the route?
>

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Daniel Tryba]

On Wednesday 24 June 2015 13:12:48 Javier Aristizábal wrote:
> Hi Daniel, I think that in fact I am doing the forwarding register to
> asterisk with the uac_req_send();

You generate a new REGISTER to the asterisk server, but:
-what will you do with the response from asterisk?
-what response will go to the endpoint of the original REGISTER?

In other words: do you have a trace for a register?

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Heh...
Well, I still have troubles with my configuration. And in SDP media adress
is Amazon public interface - but rtpengine has replace-origin
replace-session-connection session, so it must be local address.
Any ideas?
Asterisk log http://pastebin.com/MFt9V9qK
Kamailio log http://pastebin.com/jZceP2Rn
Javascript log http://pastebin.com/4ZLePyKz


2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:

> Well.. Guys, sorry, it was totally my fault. I just used VPN.
>
> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>
>> I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
>> works on other then Amazon EC2 environment and I still get this error.
>> Maybe it is somehow related to NAT traversal?
>>
>> Kamailio log: http://pastebin.com/jZceP2Rn
>> javascript log: http://pastebin.com/9Y4Pv43W
>>
>>
>> 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> Here is it
>>> http://pastebin.com/JkkM4M5m
>>>
>>> 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla :
>>>
  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:

> Here's the trace on port which I use for ws server. Don't look at
> fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
> establish a ws connection properly. Client is SIPML5 demo phone
> http://pastebin.com/LvAk2HkP
>
> 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>
>> I solved the SIP voice trouble, but WebRTC problem still exists. What
>> kind of trace I must do to make my post more informative?
>>
>> 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla <
>> mico...@gmail.com>:
>>
>>>  Hello,
>>>
>>> On 23/06/15 04:10, Alexandru Covalschi wrote:
>>>
>>>  Hello. I'm trying to set up this (v 4.2 stable):
>>>  peer <--> ec2 <--kamailio+rtpengine--> asterisk
>>>  scheme
>>>
>>>  I use advertised adress for SIP and WS connections.
>>>  The problem is that on SIP I get one way audio - I can receive
>>> audio from asterisk, but I can't transmit audio there - my SIP UA tries 
>>> to
>>> send data to Kamailio-s local EC2 IP.
>>>
>>>
>>>  you should grab a ngrep trace on server to see what happens in the
>>> signaling in order to be able to provide some hints on solving it.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>In case of WebRTC I get lot's of erros:
>>>
>>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: 
>>> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
>>> WebSocket could not be found
>>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: 
>>> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create 
>>> Via
>>> header
>>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: 
>>> [forward.c:584]: forward_request(): building failed
>>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
>>> [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
>>> terribly sorry, server error occurred (1/SL)
>>>
>>>  The call reaches Asterisk, but not vice-versa. No media is being
>>> transferred.
>>>
>>>  Rtpengine flags I use:
>>>  For SIP:  rtpengine_manage("trust-adress replace-origin
>>> replace-session-connection RTP/AVP");
>>>  For WS:  rtpengine_manage("trust-address replace-origin
>>> replace-session-connection ICE=force RTP/AVP");
>>>
>>>  Do you have any ideas how ti fix that? I also make REGFWD's to
>>> Asterisk
>>>  --
>>>  Alexandru Covalschi
>>> ABRISS-Solutions
>>> VoIP engineer and system administrator
>>> phone: +37367398493
>>> web: http://abs-telecom.com/
>>>
>>>
>>>  ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
>>> http://www.linkedin.com/in/miconda
>>> Book: SIP Routing With Kamailio - http://www.asipto.com
>>>
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>> list
>>> sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Daniel-Constantin Mierla]

Can you specify exactly which side received what IP and what you would
expect there? It is not easy to digests lots of logs and also guess what
would you expect to happen...

Cheers,
Daniel

On 24/06/15 15:14, Alexandru Covalschi wrote:
> Heh...
> Well, I still have troubles with my configuration. And in SDP media
> adress is Amazon public interface - but rtpengine has replace-origin
> replace-session-connection session, so it must be local address.
> Any ideas?
> Asterisk log http://pastebin.com/MFt9V9qK
> Kamailio log http://pastebin.com/jZceP2Rn
> Javascript log http://pastebin.com/4ZLePyKz
>
>
> 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568...@gmail.com
> >:
>
> Well.. Guys, sorry, it was totally my fault. I just used VPN.
>
> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com
> >:
>
> I used https://github.com/caruizdiaz/kamailio-ws configuration
> that 100% works on other then Amazon EC2 environment and I
> still get this error. Maybe it is somehow related to NAT
> traversal?
>
> Kamailio log: http://pastebin.com/jZceP2Rn
> javascript log: http://pastebin.com/9Y4Pv43W
>
>
> 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi
> <568...@gmail.com >:
>
> Here is it
> http://pastebin.com/JkkM4M5m
>
> 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla
> mailto:mico...@gmail.com>>:
>
> There are no major changes in 4.3 comparing with 4.2
> in regards to websocket -- the implementation is quite
> mature for a long time.
>
> Looks like websocket connection is not available. Can
> you look at javascript debug console in the browser to
> see what is printing?
>
> Daniel
>
>
> On 23/06/15 17:23, Alexandru Covalschi wrote:
>> without fix_nated_contact error behaviour is the same
>> maybe I should upgrade to 4.3 ?
>>
>> 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi
>> <568...@gmail.com >:
>>
>> Here's the trace on port which I use for ws
>> server. Don't look at fix_nated_contact, I'll fix
>> later - now the trouble is that Kamailio can't
>> establish a ws connection properly. Client is
>> SIPML5 demo phone
>> http://pastebin.com/LvAk2HkP
>>
>> 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi
>> <568...@gmail.com >:
>>
>> I solved the SIP voice trouble, but WebRTC
>> problem still exists. What kind of trace I
>> must do to make my post more informative?
>>
>> 2015-06-23 10:46 GMT+03:00 Daniel-Constantin
>> Mierla > >:
>>
>> Hello,
>>
>> On 23/06/15 04:10, Alexandru Covalschi wrote:
>>> Hello. I'm trying to set up this (v 4.2
>>> stable):
>>> peer <--> ec2 <--kamailio+rtpengine-->
>>> asterisk
>>> scheme
>>>
>>> I use advertised adress for SIP and WS
>>> connections.
>>> The problem is that on SIP I get one way
>>> audio - I can receive audio from
>>> asterisk, but I can't transmit audio
>>> there - my SIP UA tries to send data to
>>> Kamailio-s local EC2 IP.
>>
>> you should grab a ngrep trace on server
>> to see what happens in the signaling in
>> order to be able to provide some hints on
>> solving it.
>>
>> Cheers,
>> Daniel
>>
>>> In case of WebRTC I get lot's of erros:
>>>
>>> Jun 23 01:58:57 kamailio
>>> /usr/sbin/kamailio[18325]: WARNING:
>>>  [msg_translator.c:2778]:
>>> via_builder(): TCP/TLS connection (id:
>>> 0) for WebSocket could not be found
>>> Jun 23 01:58:57 kamailio
>>> /usr/sbin/kamailio[18325]: ERROR: 
>>> [msg_translator.c:1996]:
>>> build_req_buf_from_sip_req(): could not
>>>   

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Also, an interesting thing - if you can see in Kamailio log, a check of the
proto of user "300" is being made. But 300 is $tU, and $tU proto is being
checked only if source IP is asterisks IP.

Here's the part of config where rtpengine is engaged (in NATmanage route)

if((src_ip==10.0.0.87))
{
xlog("L_NOTICE","== select proto from sipusers where
name=$tU");
sql_xquery("ca_asterisk", "select proto from sipusers where
name=$tU", "ra");
 xlog("L_NOTICE","= $tU has proto $xavp(ra=>proto)");
if ($xavp(ra=>proto)=="ws")
{
 xlog("L_NOTICE","= $tU has WEBSOCKETS");

rtpengine_manage("trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF");
}
else
{
xlog("L_NOTICE","= $tU has NO fucken WEBSOCKETS");
rtpengine_manage("trust-address replace-origin
replace-session-connection");
}
} else {
xlog("L_NOTICE","== select proto from sipusers where
name=$fU");
   sql_xquery("ca_asterisk", "select proto from sipusers where
name=$fU", "ra");
  if ($xavp(ra=>proto)=="ws")
{

xlog("L_NOTICE","= $fU has WEBSOCKETS");
rtpengine_manage("trust-address replace-origin
replace-session-connection ICE=force RTP/AVP");
}
else
{
xlog("L_NOTICE","= $fU has NO WEBSOCKETS");
rtpengine_manage("replace-origin
replace-session-connection RTP/AVP");
}

}


2015-06-24 16:14 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:

> Heh...
> Well, I still have troubles with my configuration. And in SDP media adress
> is Amazon public interface - but rtpengine has replace-origin
> replace-session-connection session, so it must be local address.
> Any ideas?
> Asterisk log http://pastebin.com/MFt9V9qK
> Kamailio log http://pastebin.com/jZceP2Rn
> Javascript log http://pastebin.com/4ZLePyKz
>
>
> 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>
>> Well.. Guys, sorry, it was totally my fault. I just used VPN.
>>
>> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> I used https://github.com/caruizdiaz/kamailio-ws configuration that
>>> 100% works on other then Amazon EC2 environment and I still get this error.
>>> Maybe it is somehow related to NAT traversal?
>>>
>>> Kamailio log: http://pastebin.com/jZceP2Rn
>>> javascript log: http://pastebin.com/9Y4Pv43W
>>>
>>>
>>> 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>>
 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla 
 :

>  There are no major changes in 4.3 comparing with 4.2 in regards to
> websocket -- the implementation is quite mature for a long time.
>
> Looks like websocket connection is not available. Can you look at
> javascript debug console in the browser to see what is printing?
>
> Daniel
>
>
> On 23/06/15 17:23, Alexandru Covalschi wrote:
>
>  without fix_nated_contact error behaviour is the same
>  maybe I should upgrade to 4.3 ?
>
> 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>
>> Here's the trace on port which I use for ws server. Don't look at
>> fix_nated_contact, I'll fix later - now the trouble is that Kamailio 
>> can't
>> establish a ws connection properly. Client is SIPML5 demo phone
>> http://pastebin.com/LvAk2HkP
>>
>> 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> I solved the SIP voice trouble, but WebRTC problem still exists.
>>> What kind of trace I must do to make my post more informative?
>>>
>>> 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla <
>>> mico...@gmail.com>:
>>>
  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer <--> ec2 <--kamailio+rtpengine--> asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive
 audio from asterisk, but I can't transmit audio there - my SIP UA 
 tries to
 send data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: 
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id:

Re: [SR-Users] linking math.h while compiling a module

[Daniel-Constantin Mierla]

Hello,

what are the errors you get? Maybe it is more that libm that needs to be
linked.

Cheers,
Daniel

On 24/06/15 10:46, Shouvanik Chakrabarti wrote:
> Hello,
> I have developed a Kamailio module to handle group messaging and
> am using certain functions from the libm.so library. Compilation
> happens fine, but Kamailio subsequently fails to start because the
> math.h symbols are unlinked/undefined. Could someone please suggest a
> workaround for this.
>
> Thanks,
> Shouvanik.
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Ok, so my scheme.
Kamailio and Asterisk are in Amazon EC2
Kamailio externip=54.197.230.121 localip=10.145.45.103
Asterisk localip=10.145.45.103, externip doesn't matter

Call should flow like that:
webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip
but now it's webrtc --> kamailio-externip --> kamailio--localip -->
asterisk-localip --> kamailio-externip --> peer

I have the voice, but it's wrong scheme, and Asterisk drops call because of
retransmissions failure


2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla :

>  Can you specify exactly which side received what IP and what you would
> expect there? It is not easy to digests lots of logs and also guess what
> would you expect to happen...
>
> Cheers,
> Daniel
>
>
> On 24/06/15 15:14, Alexandru Covalschi wrote:
>
>  Heh...
>  Well, I still have troubles with my configuration. And in SDP media
> adress is Amazon public interface - but rtpengine has replace-origin
> replace-session-connection session, so it must be local address.
>  Any ideas?
>  Asterisk log http://pastebin.com/MFt9V9qK
>  Kamailio log http://pastebin.com/jZceP2Rn
>  Javascript log http://pastebin.com/4ZLePyKz
>
>
> 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>
>> Well.. Guys, sorry, it was totally my fault. I just used VPN.
>>
>> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>>  I used https://github.com/caruizdiaz/kamailio-ws configuration that
>>> 100% works on other then Amazon EC2 environment and I still get this error.
>>> Maybe it is somehow related to NAT traversal?
>>>
>>>  Kamailio log: http://pastebin.com/jZceP2Rn
>>>  javascript log: http://pastebin.com/9Y4Pv43W
>>>
>>>
>>> 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>>
 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla 
 :

>  There are no major changes in 4.3 comparing with 4.2 in regards to
> websocket -- the implementation is quite mature for a long time.
>
> Looks like websocket connection is not available. Can you look at
> javascript debug console in the browser to see what is printing?
>
> Daniel
>
>
> On 23/06/15 17:23, Alexandru Covalschi wrote:
>
>  without fix_nated_contact error behaviour is the same
>  maybe I should upgrade to 4.3 ?
>
> 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>
>> Here's the trace on port which I use for ws server. Don't look at
>> fix_nated_contact, I'll fix later - now the trouble is that Kamailio 
>> can't
>> establish a ws connection properly. Client is SIPML5 demo phone
>> http://pastebin.com/LvAk2HkP
>>
>> 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> I solved the SIP voice trouble, but WebRTC problem still exists.
>>> What kind of trace I must do to make my post more informative?
>>>
>>> 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla <
>>> mico...@gmail.com>:
>>>
  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer <--> ec2 <--kamailio+rtpengine--> asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive
 audio from asterisk, but I can't transmit audio there - my SIP UA 
 tries to
 send data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: 
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: 
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not 
 create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: 
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage("trust-adress replace-origin
 replace-session-connection RTP/AVP");
  For WS:  rtpengine_manage("trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP");

  Do you have any ideas how ti

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Asterisk localip=10.0.0.87, sorry

2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:

> Ok, so my scheme.
> Kamailio and Asterisk are in Amazon EC2
> Kamailio externip=54.197.230.121 localip=10.145.45.103
> Asterisk localip=10.145.45.103, externip doesn't matter
>
> Call should flow like that:
> webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip
> but now it's webrtc --> kamailio-externip --> kamailio--localip -->
> asterisk-localip --> kamailio-externip --> peer
>
> I have the voice, but it's wrong scheme, and Asterisk drops call because
> of retransmissions failure
>
>
> 2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla :
>
>>  Can you specify exactly which side received what IP and what you would
>> expect there? It is not easy to digests lots of logs and also guess what
>> would you expect to happen...
>>
>> Cheers,
>> Daniel
>>
>>
>> On 24/06/15 15:14, Alexandru Covalschi wrote:
>>
>>  Heh...
>>  Well, I still have troubles with my configuration. And in SDP media
>> adress is Amazon public interface - but rtpengine has replace-origin
>> replace-session-connection session, so it must be local address.
>>  Any ideas?
>>  Asterisk log http://pastebin.com/MFt9V9qK
>>  Kamailio log http://pastebin.com/jZceP2Rn
>>  Javascript log http://pastebin.com/4ZLePyKz
>>
>>
>> 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> Well.. Guys, sorry, it was totally my fault. I just used VPN.
>>>
>>> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>>
  I used https://github.com/caruizdiaz/kamailio-ws configuration that
 100% works on other then Amazon EC2 environment and I still get this error.
 Maybe it is somehow related to NAT traversal?

  Kamailio log: http://pastebin.com/jZceP2Rn
  javascript log: http://pastebin.com/9Y4Pv43W


 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:

> Here is it
> http://pastebin.com/JkkM4M5m
>
> 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla  >:
>
>>  There are no major changes in 4.3 comparing with 4.2 in regards to
>> websocket -- the implementation is quite mature for a long time.
>>
>> Looks like websocket connection is not available. Can you look at
>> javascript debug console in the browser to see what is printing?
>>
>> Daniel
>>
>>
>> On 23/06/15 17:23, Alexandru Covalschi wrote:
>>
>>  without fix_nated_contact error behaviour is the same
>>  maybe I should upgrade to 4.3 ?
>>
>> 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> Here's the trace on port which I use for ws server. Don't look at
>>> fix_nated_contact, I'll fix later - now the trouble is that Kamailio 
>>> can't
>>> establish a ws connection properly. Client is SIPML5 demo phone
>>> http://pastebin.com/LvAk2HkP
>>>
>>> 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>>>
 I solved the SIP voice trouble, but WebRTC problem still exists.
 What kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla <
 mico...@gmail.com>:

>  Hello,
>
> On 23/06/15 04:10, Alexandru Covalschi wrote:
>
>  Hello. I'm trying to set up this (v 4.2 stable):
>  peer <--> ec2 <--kamailio+rtpengine--> asterisk
>  scheme
>
>  I use advertised adress for SIP and WS connections.
>  The problem is that on SIP I get one way audio - I can receive
> audio from asterisk, but I can't transmit audio there - my SIP UA 
> tries to
> send data to Kamailio-s local EC2 IP.
>
>
>  you should grab a ngrep trace on server to see what happens in
> the signaling in order to be able to provide some hints on solving it.
>
> Cheers,
> Daniel
>
>In case of WebRTC I get lot's of erros:
>
> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING:
>  [msg_translator.c:2778]: via_builder(): TCP/TLS connection 
> (id: 0)
> for WebSocket could not be found
> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: 
> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not 
> create Via
> header
> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: 
> [forward.c:584]: forward_request(): building failed
> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
> [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
> terribly sorry, server error occurred (1/SL)
>
>  The call reaches Asterisk, but not vice-versa. No media is being
> transferred.
>
>  Rtpengine flags I use:
>  For SIP:  rtpengine_

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Hi Daniel,

this is the trace: http://pastebin.com/ncwsptz0

Thanks!

On Wed, Jun 24, 2015 at 3:05 PM, Daniel Tryba  wrote:

> On Wednesday 24 June 2015 13:12:48 Javier Aristizábal wrote:
> > Hi Daniel, I think that in fact I am doing the forwarding register to
> > asterisk with the uac_req_send();
>
> You generate a new REGISTER to the asterisk server, but:
> -what will you do with the response from asterisk?
> -what response will go to the endpoint of the original REGISTER?
>
> In other words: do you have a trace for a register?
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Daniel-Constantin Mierla]

Do you have proper routing rules between the local ips of kamailio and
asterisk? Why aren't you use only external IPs if they are on different
servers? Asterisk has also the option to set external ip. It can reduce
the complexity of doing bridging of signaling and rtp. Once you get that
working you can start adding bridging step by step.

Cheers,
Daniel

On 24/06/15 15:25, Alexandru Covalschi wrote:
> Asterisk localip=10.0.0.87, sorry
>
> 2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568...@gmail.com
> >:
>
> Ok, so my scheme.
> Kamailio and Asterisk are in Amazon EC2
> Kamailio externip=54.197.230.121 localip=10.145.45.103
> Asterisk localip=10.145.45.103, externip doesn't matter
>
> Call should flow like that:
> webrtc <--> kamailio-externip <--> kamailio-localip <-->
> asterisk-localip
> but now it's webrtc --> kamailio-externip --> kamailio--localip
> --> asterisk-localip --> kamailio-externip --> peer
>
> I have the voice, but it's wrong scheme, and Asterisk drops call
> because of retransmissions failure
>
>
> 2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla
> mailto:mico...@gmail.com>>:
>
> Can you specify exactly which side received what IP and what
> you would expect there? It is not easy to digests lots of logs
> and also guess what would you expect to happen...
>
> Cheers,
> Daniel
>
>
> On 24/06/15 15:14, Alexandru Covalschi wrote:
>> Heh...
>> Well, I still have troubles with my configuration. And in SDP
>> media adress is Amazon public interface - but rtpengine has
>> replace-origin replace-session-connection session, so it must
>> be local address.
>> Any ideas?
>> Asterisk log http://pastebin.com/MFt9V9qK
>> Kamailio log http://pastebin.com/jZceP2Rn
>> Javascript log http://pastebin.com/4ZLePyKz
>>
>>
>> 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi
>> <568...@gmail.com >:
>>
>> Well.. Guys, sorry, it was totally my fault. I just used
>> VPN.
>>
>> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi
>> <568...@gmail.com >:
>>
>> I used https://github.com/caruizdiaz/kamailio-ws
>> configuration that 100% works on other then Amazon
>> EC2 environment and I still get this error. Maybe it
>> is somehow related to NAT traversal?
>>
>> Kamailio log: http://pastebin.com/jZceP2Rn
>> javascript log: http://pastebin.com/9Y4Pv43W
>>
>>
>> 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi
>> <568...@gmail.com >:
>>
>> Here is it
>> http://pastebin.com/JkkM4M5m
>>
>> 2015-06-23 18:53 GMT+03:00 Daniel-Constantin
>> Mierla > >:
>>
>> There are no major changes in 4.3 comparing
>> with 4.2 in regards to websocket -- the
>> implementation is quite mature for a long time.
>>
>> Looks like websocket connection is not
>> available. Can you look at javascript debug
>> console in the browser to see what is printing?
>>
>> Daniel
>>
>>
>> On 23/06/15 17:23, Alexandru Covalschi wrote:
>>> without fix_nated_contact error behaviour is
>>> the same
>>> maybe I should upgrade to 4.3 ?
>>>
>>> 2015-06-23 14:08 GMT+03:00 Alexandru
>>> Covalschi <568...@gmail.com
>>> >:
>>>
>>> Here's the trace on port which I use for
>>> ws server. Don't look at
>>> fix_nated_contact, I'll fix later - now
>>> the trouble is that Kamailio can't
>>> establish a ws connection properly.
>>> Client is SIPML5 demo phone
>>> http://pastebin.com/LvAk2HkP
>>>
>>> 2015-06-23 14:03 GMT+03:00 Alexandru
>>> Covalschi <568...@gmail.com
>>> >:
>>>
>>> I solved the SIP voice trouble, but
>>> WebRTC problem still exists. What
>>> kind of trace I must do to make my
>>> post more informative?
>>>
>>> 2015-06-23 10:46 GMT+03:00
>>> Daniel-C

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

I got bridging working well on internal interfaces in case of simple SIP
calls on a bit other configuration. But editing this config to support
WebRTC causes same problems. I need internal interfaces on asterisk to
completely close external ones (Security etc.).
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Daniel Tryba]

On Wednesday 24 June 2015 15:26:17 Javier Aristizábal wrote:
> this is the trace: http://pastebin.com/ncwsptz0

Okee, I think I understand the solution. 

Endpoint registers on kamailio via challenge/response. When register is 
succesful a new register is created for asterisk, but kamailio failes to 
respond to eht 401 unauthorized. So you either need to handle the 
challenge/response from asterisk, or tell asterisk to not authenticate 
registers from kamailio.

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Hola Daniel!

Could you please help me with your idea? :)

this is my Routing logic: http://pastebin.com/bdWK1JW3

Thanks!

On Wed, Jun 24, 2015 at 3:48 PM, Daniel Tryba  wrote:

> On Wednesday 24 June 2015 15:26:17 Javier Aristizábal wrote:
> > this is the trace: http://pastebin.com/ncwsptz0
>
> Okee, I think I understand the solution.
>
> Endpoint registers on kamailio via challenge/response. When register is
> succesful a new register is created for asterisk, but kamailio failes to
> respond to eht 401 unauthorized. So you either need to handle the
> challenge/response from asterisk, or tell asterisk to not authenticate
> registers from kamailio.
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Daniel, is this using proxy_challenge? from the auth module?

On Wed, Jun 24, 2015 at 3:55 PM, Javier Aristizábal <
javieraristiza...@gmail.com> wrote:

> Hola Daniel!
>
> Could you please help me with your idea? :)
>
> this is my Routing logic: http://pastebin.com/bdWK1JW3
>
> Thanks!
>
> On Wed, Jun 24, 2015 at 3:48 PM, Daniel Tryba  wrote:
>
>> On Wednesday 24 June 2015 15:26:17 Javier Aristizábal wrote:
>> > this is the trace: http://pastebin.com/ncwsptz0
>>
>> Okee, I think I understand the solution.
>>
>> Endpoint registers on kamailio via challenge/response. When register is
>> succesful a new register is created for asterisk, but kamailio failes to
>> respond to eht 401 unauthorized. So you either need to handle the
>> challenge/response from asterisk, or tell asterisk to not authenticate
>> registers from kamailio.
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> --
> Javier Aristizábal
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Daniel Tryba]

On Wednesday 24 June 2015 16:02:58 Javier Aristizábal wrote:
> Daniel, is this using proxy_challenge? from the auth module?

You need uac_auth():
http://kamailio.org/docs/modules/stable/modules/uac.html#uac.f.uac_auth%28%29

How you disable authentication for realtime users in asterisk, I don't know. 
My guess is to have an empty (NULL) secret in the asterisk table/view.

-- 

Telefoon: 088 0100 700
Sales: sa...@pocos.nl | Service: serviced...@pocos.nl
http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 
17097024


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

[Andrey Utkin]

Thanks for all the suggestions.
Have taken pcap dumps with commands
tcpdump -i any -w /tmp/sip_tcp.pcap host 

http://whdd.org/sip_tcp_server.pcap
http://whdd.org/sip_tcp_client.pcap

The only obviously bad thing I see is "[TCP Previous segment not
captured]" mark on the packet client side dump which is the ending of
INVITE request body.
Googling on this message, I've found some advice at StackOverflow to
tune TCP socket TCP_MAXSEG option.


http://whdd.org/sip_tcp_mtu800_server.pcap
http://whdd.org/sip_tcp_mtu800_client.pcap

For this test I have set MTU to 800 on the server's only actual net
interface, and restarted Kamailio. The situation stays the same,
despite the packets show differently in wireshark.
On client side, there's still "[TCP Previous segment not captured]",
and on server side there's a mark "[TCP segment of a reassembled
PDU]".

Any review is greatly appreciated!

-- 
Andrey Utkin

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Yes, I think is with NULL in that field.

So, if I leave the secret field with NULL, I assume that I can  manage the
authentication with uac_auth(), right?

Or I can do it with ACL?

Thanks!

On Wed, Jun 24, 2015 at 4:20 PM, Daniel Tryba  wrote:

> On Wednesday 24 June 2015 16:02:58 Javier Aristizábal wrote:
> > Daniel, is this using proxy_challenge? from the auth module?
>
> You need uac_auth():
>
> http://kamailio.org/docs/modules/stable/modules/uac.html#uac.f.uac_auth%28%29
>
> How you disable authentication for realtime users in asterisk, I don't
> know.
> My guess is to have an empty (NULL) secret in the asterisk table/view.
>
> --
>
> Telefoon: 088 0100 700
> Sales: sa...@pocos.nl | Service: serviced...@pocos.nl
> http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel
> 17097024
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Daniel Tryba]

On Wednesday 24 June 2015 16:31:31 Javier Aristizábal wrote:
> So, if I leave the secret field with NULL, I assume that I can  manage the
> authentication with uac_auth(), right?
> 
> Or I can do it with ACL?

It is an or, no secret -> no need for uac_auth,

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Hola Daniel,

I just put NULL on the secret field and I got the status OK.. strange isn't?


Name/username HostDyn
Forcerport ComediaACL Port Status  Description
Realtime

101/101   192.168.1.156D  Yes
  Yes5060 OK (590 ms)
Cached RT


On Wed, Jun 24, 2015 at 4:31 PM, Javier Aristizábal <
javieraristiza...@gmail.com> wrote:

> Yes, I think is with NULL in that field.
>
> So, if I leave the secret field with NULL, I assume that I can  manage the
> authentication with uac_auth(), right?
>
> Or I can do it with ACL?
>
> Thanks!
>
> On Wed, Jun 24, 2015 at 4:20 PM, Daniel Tryba  wrote:
>
>> On Wednesday 24 June 2015 16:02:58 Javier Aristizábal wrote:
>> > Daniel, is this using proxy_challenge? from the auth module?
>>
>> You need uac_auth():
>>
>> http://kamailio.org/docs/modules/stable/modules/uac.html#uac.f.uac_auth%28%29
>>
>> How you disable authentication for realtime users in asterisk, I don't
>> know.
>> My guess is to have an empty (NULL) secret in the asterisk table/view.
>>
>> --
>>
>> Telefoon: 088 0100 700
>> Sales: sa...@pocos.nl | Service: serviced...@pocos.nl
>> http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel
>> 17097024
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> --
> Javier Aristizábal
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Daniel Tryba]

On Wednesday 24 June 2015 16:46:01 Javier Aristizábal wrote:
> I just put NULL on the secret field and I got the status OK.. strange isn't?

This is intended behavior.

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Well, but in this behavior, anybody can register because the secret field
is empty, how can manage a password for each extension?

On Wed, Jun 24, 2015 at 4:59 PM, Daniel Tryba  wrote:

> On Wednesday 24 June 2015 16:46:01 Javier Aristizábal wrote:
> > I just put NULL on the secret field and I got the status OK.. strange
> isn't?
>
> This is intended behavior.
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Javier Aristizábal]

Hello Daniel,

I think I understand now.. thanks a lot for your help and input :-)

On Wed, Jun 24, 2015 at 5:03 PM, Javier Aristizábal <
javieraristiza...@gmail.com> wrote:

> Well, but in this behavior, anybody can register because the secret field
> is empty, how can manage a password for each extension?
>
> On Wed, Jun 24, 2015 at 4:59 PM, Daniel Tryba  wrote:
>
>> On Wednesday 24 June 2015 16:46:01 Javier Aristizábal wrote:
>> > I just put NULL on the secret field and I got the status OK.. strange
>> isn't?
>>
>> This is intended behavior.
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> --
> Javier Aristizábal
>



-- 
Javier Aristizábal
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + RTPEngine + Asterisk

[Daniel Tryba]

On Wednesday 24 June 2015 17:14:03 Javier Aristizábal wrote:
> I think I understand now.. thanks a lot for your help and input :-)

I'm really wondering what tutorial you followed?

http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
has the same route[REGFWD]. But also mentions:
>sipusers is the standard table required by Asterisk to store SIP user 
>profile, with one extra column sippasswd where will be stored the password 
>for SIP authentication. By default, Asterisk uses the column secret for SIP 
>user password, but if that is filled in, Asterisk will ask for authentication 
>again, resulting in double-authentication which we want to avoid.

and 

>Be sure you configure Asterisk to not authenticate SIP requests coming from 
>Kamailio. 

But if you are worried about exposing asterisk to the world, if you make it 
communicate through kamailio you could bind it to localhost only.

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users