date:20180723

Re: [SR-Users] Kamailio 5.0.X RPM packages/repo broken

2018-07-23 Thread Patrick Wakano

Hello Sergey!
Hope you are doing fine!

I am having this same issue again today
Would you be able to check that?

Thanks very much!
Kind regards,
Patrick Wakano

# yum install kamailio
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
 * base: centos.mirror.serversaustralia.com.au
 * epel: mirror.intergrid.com.au
 * extras: centos.melbourneitmirror.net
 * remi: remi.conetix.com.au
 * remi-safe: remi.conetix.com.au
 * updates: mirror.intergrid.com.au
Resolving Dependencies
--> Running transaction check
---> Package kamailio.x86_64 0:5.0.7-3.el7.centos will be installed
--> Processing Dependency: libkamailio_ims.so.0()(64bit) for package:
kamailio-5.0.7-3.el7.centos.x86_64
--> Running transaction check
---> Package kamailio-ims.x86_64 0:5.0.7-3.el7.centos will be installed
--> Finished Dependency Resolution

Dependencies Resolved

=
 Package  Arch
Version
Repository Size
=
Installing:
 kamailio x86_64
5.0.7-3.el7.centos
home_kamailio_v5.0.x-rpms 5.1 M
Installing for dependencies:
 kamailio-ims x86_64
5.0.7-3.el7.centos
home_kamailio_v5.0.x-rpms 1.2 M

Transaction Summary
=
Install  1 Package (+1 Dependent package)

Total download size: 6.4 M
Installed size: 29 M
Is this ok [y/d/N]: y
Downloading packages:
Delta RPMs disabled because /usr/bin/applydeltarpm not installed.
kamailio-5.0.7-3.el7.centos.x8
FAILED
http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/CentOS_7/x86_64/kamailio-5.0.7-3.el7.centos.x86_64.rpm:
[Errno 14] HTTP Error 416 - Requested Range Not Satisfiable
Trying other mirror.
kamailio-ims-5.0.7-3.el7.cento
FAILED  ]
218 kB/s | 710 kB  00:00:26 ETA
http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/CentOS_7/x86_64/kamailio-ims-5.0.7-3.el7.centos.x86_64.rpm:
[Errno -1] Package does not match intended download. Suggestion: run yum
--enablerepo=home_kamailio_v5.0.x-rpms clean metadata
Trying other mirror.


Error downloading packages:
  kamailio-5.0.7-3.el7.centos.x86_64: [Errno 256] No more mirrors to try.
  kamailio-ims-5.0.7-3.el7.centos.x86_64: [Errno 256] No more mirrors to
try.


On 10 July 2018 at 14:49, Patrick Wakano  wrote:

> Thank you Sergey!!
> Installation was successful now!!
>
> Cheers!
> Patrick Wakano
>
> On 10 July 2018 at 14:24, Sergey Safarov  wrote:
>
>> I was trigered rpm rebuild.
>> Please check after 40 min.
>>
>> вт, 10 июля 2018 г., 6:57 Patrick Wakano :
>>
>>> Hello list,
>>> Hope you all doing fine!
>>>
>>> I've been trying to install Kamailio packages from the opensuse repo (as
>>> per https://www.kamailio.org/wiki/packages/rpms) but the installation
>>> is failing with this message:
>>> Downloading packages:
>>> Delta RPMs disabled because /usr/bin/applydeltarpm not installed.
>>> kamailio-ims-5.0.7-1.el7.cento FAILED
>>>
>>> http://download.opensuse.org/repositories/home%3A/kamailio%3
>>> A/v5.0.x-rpms/CentOS_7/x86_64/kamailio-ims-5.0.7-1.el7.centos.x86_64.rpm:
>>> [Errno 14] HTTP Error 416 - Requested Range Not Satisfiable  --:--:-- ETA
>>> Trying other mirror.
>>> kamailio-5.0.7-1.el7.centos.x8 FAILED
>>> 77% [-
>>> ] 1.0 MB/s | 4.9 MB  00:00:01 ETA
>>> http://download.opensuse.org/repositories/home%3A/kamailio%3
>>> A/v5.0.x-rpms/CentOS_7/x86_64/kamailio-5.0.7-1.el7.centos.x86_64.rpm:
>>> [Errno -1] Package does not match intended download. Suggestion: run yum
>>> --enablerepo=home_kamailio_v5.0.x-rpms clean metadata
>>> Trying other mirror.
>>>
>>> I've successfully installed the 5.0.7 version on Friday (06/07) but I've
>>> noticed the packages have changed last Saturday (07/07/2018) Given this
>>> error message looks like the repository is broken somehow
>>> Is anyone aware of such issue? I haven't tried the 5.1.X repo because my
>>> script was not migrated to 5.1.X yet
>>>
>>> Thanks,
>>> Kind regards,
>>> Patrick Wakano
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Re: [SR-Users] Error when using http_async_client $http_query_id var

2018-07-23 Thread Henning Westerholt

Am Montag, 23. Juli 2018, 13:52:32 CEST schrieb je...@cyberchaos.nl:
> Thanks that works, the docs are referring to $http_query_id
> (https://kamailio.org/docs/modules/5.1.x/modules/http_async_client.html#http
> _async_client.f.http_async_query)

Hi Jan,

I have fixed the error in the docs in master, 5.0 and 5.1 branch.

Best regards,

Henning


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Re: [SR-Users] NATHELPER issue

2018-07-23 Thread Ali Taher

Thank you Konstantin,

 

What I can’t understand is why media packets are sent correctly to the public 
IP , while early media (RBT) are sent on private.

 

Regards,

Ali

From: Konstantin Polyakov  
Sent: Monday, July 23, 2018 1:38 PM
To: Ali Taher 
Subject: Re[4]: [SR-Users] NATHELPER issue

 

Original INVITE  contains private IP in c= parameter, so you need to use STUN 
or TURN or ICE, something which is supported in your system or in your 
softphone.
See my comments in the previous mail.

Best regrads



Понедельник, 23 июля 2018, 12:25 +03:00 от Ali Taher mailto:ata...@vanrise.com> >:

Hi Konstantin,

 

INVITE:

 

INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0

Record-Route: 

X-AUTH-IP: Y.Y.Y.Y

Via: SIP/2.0/UDP 
X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0

Via: SIP/2.0/UDP 
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556

Max-Forwards: 69

Contact: 

To: "9613159370"

From: "+967123456";tag=5269152d

Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO

Content-Type: application/sdp

User-Agent: eyeBeam release 1003s stamp 31159

Content-Length: 434

 

v=0

o=- 5 2 IN IP4 192.168.26.3

s=CounterPath eyeBeam 1.5

c=IN IP4 192.168.26.3

t=0 0

m=audio 8602 RTP/AVP 0 8 101

a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602

a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602

a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602

a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv

a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF

 

 

200 OK

 

200 OK

Via: SIP/2.0/UDP 
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556

Record-Route: 

From: "+967123456" ;tag=5269152d

To: "9613159370" ;tag=agtHmrp3Dccrg

Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.

CSeq: 1 INVITE

Contact: 

User-Agent: FIKAR

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, 
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, 
line-seize, call-info, sla, include-session-description, presence.winfo, 
message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 200

Remote-Party-ID: "Outbound Call" mailto:506251539613159370@X.X.X.X> >;party=calling;privacy=off;screen=no

 

v=0

o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X

s=FIKAR

c=IN IP4 X.X.X.X

t=0 0

m=audio 24496 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

where:

X.X.X.X is Kamailio IP

Y.Y.Y.Y is origination public IP

 

 

Regards,

Ali

 

From: Konstantin Polyakov mailto:piligrim...@mail.ru> > 
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher mailto:ata...@vanrise.com> >
Subject: Re[2]: [SR-Users] NATHELPER issue

 

Hello Ali,

Could you show your latest INVITE and OK?

Check you SDP bodies in those messages, if you still have there private IPs 
then you need to solve usual NAT traversal task in your media software (not 
proxy).
Kamailio doesn't process media as you know. 
Of course you can try fix_nated_sdp  function from nathelper module but it will 
not work for symmetric NAT because you fix SDP with public IP toward proxy but 
not toward UA.
Your media software or hardware should support STUN or ICE protocols to work 
correctly via NAT.

Best regards.

Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher  >:

Thank you for your reply.

 

ACK is now sent correctly to public IP , yet I have another issue with RBT  
where it’s still sent to private IP .Yet the call media is sent correctly to 
the public IP.

 

There is something strange , as I hear two short different ringing tones then 
complete silence till the called party answer the call and RTP flow normally.

 

How can I handle this case?

 

Thanks,

Ali

 

From: Konstantin Polyakov  > 
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org 
 
; Ali Taher  >
Subject: Re: [SR-Users] NATHELPER issue

 

Hello Ali,

ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.

My NATDETECT looks like:

if (nat_uac_test("19")) {

   fix_nated_contact();

}


I call it for requests and responses.

Best regards.
Konstantin

 

Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" mailto:ata...@vanrise.com> >
To: mailto:sr-us...@lists.sip-router.org> >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com 
 >
Content-Type: text/plain; charset="utf-8"

Hello,

 

I'm using Kamailio 4.2 as proxy with nathelper enabled.

 

Yet , the ACK packet sent from the proxy to the origination's private

Re: [SR-Users] Kamailio Calls Concurrency

2018-07-23 Thread Abtin Mir

Hello Henning,
Thanks for your reply.

I understand the IO operations affects to overall performance. That proxy
server is actually a dispatcher (Load Balancer)  to different Asterisk pbx
endpoints and is only going to log INVITEs and BYEs. Since I'm new to SER ,
Not really sure If should have RTP-Proxy as well, or not.(Your advice on
this is appreciated).

 So how you think the concurrency would be now? (Considering 16GB of RAM
and 4 cores of Xeon-Silver 4114) I just need a estimated number by an
expert like you.

Thanks again for your response.

On Sun, Jul 22, 2018 at 12:23 PM, Henning Westerholt 
wrote:

> Am Freitag, 6. Juli 2018, 09:52:16 CEST schrieb Abtin Mir:
> > As a proxy server , How many concurrent calls can be processed by a
> > Kamailio which is installed on a VM with 4 cores and 16GB of RAM?
> >
> > Your answer is much appreciated.
>
> Hello Abtin,
>
> this depends a lot on your configuration, if you use a lot of database
> requests, if the database is local or remote, the database performance,
> the
> amount of logging you are doing locally, do you handle RTP as well and so
> on.
>
> If your server is not exceedingly using a lot of this mentioned IO
> operations,
> you should be able to handle many thousands concurrent calls with one
> server.
>
> But to have a proper answer for your certain scenario, you need to test it.
>
> Best regards,
>
> Henning
>
>
> --
> If you like my work in the Kamailio project, it would be great if you
> could
> consider supporting me on Patreon: https://www.patreon.com/henningw
>

-- 

*Regards,*

*Abtin Mir*
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[SR-Users] messages dumped in wrong order sometime

2018-07-23 Thread Philip Kucheev

 Hello there!
Using kamailio 5.0.1  and msilo module for storing offline messages I
noticed that sometimes, especially when a big number of messages dumping
for one user, some of them can arrive to receiver in wrong order.
Lets say we have 20 messages and the order of receiving could be next
1..18..17..19..20.
That could be a reason of that ?
Thanks,
Philip
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Re: [SR-Users] Error when using http_async_client $http_query_id var

2018-07-23 Thread jenus


Alex,

Thanks that works, the docs are referring to $http_query_id 
(https://kamailio.org/docs/modules/5.1.x/modules/http_async_client.html#http_async_client.f.http_async_query)


Jan



Alex Balashov schreef op 2018-07-23 13:39:

Hi,

$http_query_id does not exist. However, $http_req_id does.

On Mon, Jul 23, 2018 at 01:37:30PM +0200, je...@cyberchaos.nl wrote:


Hello,

I'm trying to build a async http call to a external server and was 
trying to
use the $http_query_id var to identity the http call in the response 
route.


But when i try the following:

$http_req(suspend) = 0;
$http_req(hdr) = "Content-Type: application/json";
$http_req(body) = "{'r_uri':'" + $rU + "', 'f_uri':'" + $fU + "'}";
http_async_query("http://10.10.10.1/;, "ACC_HTTP_REPLY");
xlog("L_WARN", "ACC Query id: $http_query_id");


route[ACC_HTTP_REPLY] {
xlog("L_WARN", "route[ACC_HTTP_REPLY]: received reply for query
$http_query_id\n");
}

I get errors in the kamailio log file :

Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: 
[core/pvapi.c:903]: pv_parse_spec2(): error searching pvar 
"http_query_id"

Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: 
[core/pvapi.c:1107]: pv_parse_spec2(): wrong char [d/100] in
[$http_query_id] at [13 (0)]
Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: xlog 
[xlog.c:513]:
xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received 
reply for

query $http_query_id#012]
Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: 
[core/route.c:1154]: fix_actions(): fixing failed (code=-1) at
cfg://etc/kamailio/kamailio.cfg:5308
Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: 
[core/pvapi.c:903]: pv_parse_spec2(): error searching pvar 
"http_query_id"

Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: 
[core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in
[$(http_query_id)] at [15 (1)]
Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: xlog 
[xlog.c:513]:
xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received 
reply for

query $(http_query_id)#012]
Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: 
[core/route.c:1154]: fix_actions(): fixing failed (code=-1) at
cfg://etc/kamailio/kamailio.cfg:5308


Should i escape the var $http_query_id in some way?

I'm running kamailio 5.1.3.


Thanks,

Jan

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Re: [SR-Users] Error when using http_async_client $http_query_id var

2018-07-23 Thread Alex Balashov

Hi,

$http_query_id does not exist. However, $http_req_id does. :-)

On Mon, Jul 23, 2018 at 01:37:30PM +0200, je...@cyberchaos.nl wrote:

> Hello,
> 
> I'm trying to build a async http call to a external server and was trying to
> use the $http_query_id var to identity the http call in the response route.
> 
> But when i try the following:
> 
> $http_req(suspend) = 0;
> $http_req(hdr) = "Content-Type: application/json";
> $http_req(body) = "{'r_uri':'" + $rU + "', 'f_uri':'" + $fU + "'}";
> http_async_query("http://10.10.10.1/;, "ACC_HTTP_REPLY");
> xlog("L_WARN", "ACC Query id: $http_query_id");
> 
> 
> route[ACC_HTTP_REPLY] {
> xlog("L_WARN", "route[ACC_HTTP_REPLY]: received reply for query
> $http_query_id\n");
> }
> 
> I get errors in the kamailio log file :
> 
> Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: 
> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "http_query_id"
> Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: 
> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [d/100] in
> [$http_query_id] at [13 (0)]
> Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: xlog [xlog.c:513]:
> xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received reply for
> query $http_query_id#012]
> Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: 
> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at
> cfg://etc/kamailio/kamailio.cfg:5308
> Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: 
> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "http_query_id"
> Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: 
> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in
> [$(http_query_id)] at [15 (1)]
> Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: xlog [xlog.c:513]:
> xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received reply for
> query $(http_query_id)#012]
> Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: 
> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at
> cfg://etc/kamailio/kamailio.cfg:5308
> 
> 
> Should i escape the var $http_query_id in some way?
> 
> I'm running kamailio 5.1.3.
> 
> 
> Thanks,
> 
> Jan
> 
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[SR-Users] Error when using http_async_client $http_query_id var

2018-07-23 Thread jenus


Hello,

I'm trying to build a async http call to a external server and was 
trying to use the $http_query_id var to identity the http call in the 
response route.


But when i try the following:

$http_req(suspend) = 0;
$http_req(hdr) = "Content-Type: application/json";
$http_req(body) = "{'r_uri':'" + $rU + "', 'f_uri':'" + $fU + "'}";
http_async_query("http://10.10.10.1/;, "ACC_HTTP_REPLY");
xlog("L_WARN", "ACC Query id: $http_query_id");


route[ACC_HTTP_REPLY] {
xlog("L_WARN", "route[ACC_HTTP_REPLY]: received reply for query 
$http_query_id\n");

}

I get errors in the kamailio log file :

Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR:  
[core/pvapi.c:903]: pv_parse_spec2(): error searching pvar 
"http_query_id"
Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR:  
[core/pvapi.c:1107]: pv_parse_spec2(): wrong char [d/100] in 
[$http_query_id] at [13 (0)]
Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: xlog 
[xlog.c:513]: xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: 
received reply for query $http_query_id#012]
Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR:  
[core/route.c:1154]: fix_actions(): fixing failed (code=-1) at 
cfg://etc/kamailio/kamailio.cfg:5308
Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR:  
[core/pvapi.c:903]: pv_parse_spec2(): error searching pvar 
"http_query_id"
Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR:  
[core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in 
[$(http_query_id)] at [15 (1)]
Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: xlog 
[xlog.c:513]: xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: 
received reply for query $(http_query_id)#012]
Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR:  
[core/route.c:1154]: fix_actions(): fixing failed (code=-1) at 
cfg://etc/kamailio/kamailio.cfg:5308



Should i escape the var $http_query_id in some way?

I'm running kamailio 5.1.3.


Thanks,

Jan

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Re: [SR-Users] Forwarding to backup server

2018-07-23 Thread Daniel Tryba

On Sun, Jul 22, 2018 at 11:11:12AM +0200, Pali Roh??r wrote:
> Kamailio should try to send SIP messages directly to foreign SIP domains
> and when it fails (e.g. user not available or foreign SIP is not
> running), then forward message to that my SIP <--> XMPP gateway which
> will try to deliver it via XMPP.

There are more scenarios you need to keep in mind. A simple one is if
lookup() fails (not registered if you are running a registrar on this
instance) to just change the domain in the R-URI and relay it there:

if(!lookup("location"))
{
$rd="xmpp.example.org";
route(RELAY);
exit;
}

You can use a similar approach in failure routes primed:
failure_route[MANAGE_FAILURE] {
if (t_check_status("[456][0-9][0-9]"))
{
$rd="xmpp.example.org";
route(RELAY);
exit;
}
}

Be sure to use a different failure route (or add some logic to the if to
avoid infinite loops).

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Re: [SR-Users] NATHELPER issue

2018-07-23 Thread Ali Taher

Hi Konstantin,

 

INVITE:

 

INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0

Record-Route: 

X-AUTH-IP: Y.Y.Y.Y

Via: SIP/2.0/UDP 
X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0

Via: SIP/2.0/UDP 
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556

Max-Forwards: 69

Contact: 

To: "9613159370"

From: "+967123456";tag=5269152d

Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO

Content-Type: application/sdp

User-Agent: eyeBeam release 1003s stamp 31159

Content-Length: 434

 

v=0

o=- 5 2 IN IP4 192.168.26.3

s=CounterPath eyeBeam 1.5

c=IN IP4 192.168.26.3

t=0 0

m=audio 8602 RTP/AVP 0 8 101

a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602

a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602

a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602

a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv

a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF

 

 

200 OK

 

200 OK

Via: SIP/2.0/UDP 
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556

Record-Route: 

From: "+967123456" ;tag=5269152d

To: "9613159370" ;tag=agtHmrp3Dccrg

Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.

CSeq: 1 INVITE

Contact: 

User-Agent: FIKAR

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, 
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, 
line-seize, call-info, sla, include-session-description, presence.winfo, 
message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 200

Remote-Party-ID: "Outbound Call" 
;party=calling;privacy=off;screen=no

 

v=0

o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X

s=FIKAR

c=IN IP4 X.X.X.X

t=0 0

m=audio 24496 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

where:

X.X.X.X is Kamailio IP

Y.Y.Y.Y is origination public IP

 

 

Regards,

Ali

 

From: Konstantin Polyakov  
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher 
Subject: Re[2]: [SR-Users] NATHELPER issue

 

Hello Ali,

Could you show your latest INVITE and OK?

Check you SDP bodies in those messages, if you still have there private IPs 
then you need to solve usual NAT traversal task in your media software (not 
proxy).
Kamailio doesn't process media as you know. 
Of course you can try fix_nated_sdp  function from nathelper module but it will 
not work for symmetric NAT because you fix SDP with public IP toward proxy but 
not toward UA.
Your media software or hardware should support STUN or ICE protocols to work 
correctly via NAT.

Best regards.

Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher mailto:ata...@vanrise.com> >:

Thank you for your reply.

 

ACK is now sent correctly to public IP , yet I have another issue with RBT  
where it’s still sent to private IP .Yet the call media is sent correctly to 
the public IP.

 

There is something strange , as I hear two short different ringing tones then 
complete silence till the called party answer the call and RTP flow normally.

 

How can I handle this case?

 

Thanks,

Ali

 

From: Konstantin Polyakov mailto:piligrim...@mail.ru> > 
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org  ; Ali 
Taher mailto:ata...@vanrise.com> >
Subject: Re: [SR-Users] NATHELPER issue

 

Hello Ali,

ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.

My NATDETECT looks like:

if (nat_uac_test("19")) {

   fix_nated_contact();

}


I call it for requests and responses.

Best regards.
Konstantin

 

Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher"  >
To: 
 >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com 

 >
Content-Type: text/plain; charset="utf-8"

Hello,

 

I'm using Kamailio 4.2 as proxy with nathelper enabled.

 

Yet , the ACK packet sent from the proxy to the origination's private IP.

 

The ACK is sent as reply on the following 200 OK sent from the origination :


 

8m2EJN41BN/6WSIP/2.0 200 OK

From: ;tag=XQBQNjvjgp4Ze

To: ;tag=12033368836000

Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN

Record-Route: 

Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX 

 

CSeq: 125698370 INVITE

Accept: application/sdp

Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE

P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XX.XX
;orig-ioi=MXX

Content-Type: application/sdp

Contact:

Re: [SR-Users] NATHELPER issue

2018-07-23 Thread Ali Taher

Hi Konstantin,

 

INVITE:

 

INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0

Record-Route: 

X-AUTH-IP: Y.Y.Y.Y

Via: SIP/2.0/UDP 
X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0

Via: SIP/2.0/UDP 
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556

Max-Forwards: 69

Contact: 

To: "9613159370"

From: "+967123456";tag=5269152d

Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO

Content-Type: application/sdp

User-Agent: eyeBeam release 1003s stamp 31159

Content-Length: 434

 

v=0

o=- 5 2 IN IP4 192.168.26.3

s=CounterPath eyeBeam 1.5

c=IN IP4 192.168.26.3

t=0 0

m=audio 8602 RTP/AVP 0 8 101

a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602

a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602

a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602

a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv

a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF

 

 

200 OK

 

200 OK

Via: SIP/2.0/UDP 
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556

Record-Route: 

From: "+967123456" ;tag=5269152d

To: "9613159370" ;tag=agtHmrp3Dccrg

Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.

CSeq: 1 INVITE

Contact: 

User-Agent: FIKAR

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, 
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, 
line-seize, call-info, sla, include-session-description, presence.winfo, 
message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 200

Remote-Party-ID: "Outbound Call" 
;party=calling;privacy=off;screen=no

 

v=0

o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X

s=FIKAR

c=IN IP4 X.X.X.X

t=0 0

m=audio 24496 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

 

Regards,

Ali

 

 

From: Konstantin Polyakov  
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher 
Subject: Re[2]: [SR-Users] NATHELPER issue

 

Hello Ali,

Could you show your latest INVITE and OK?

Check you SDP bodies in those messages, if you still have there private IPs 
then you need to solve usual NAT traversal task in your media software (not 
proxy).
Kamailio doesn't process media as you know. 
Of course you can try fix_nated_sdp  function from nathelper module but it will 
not work for symmetric NAT because you fix SDP with public IP toward proxy but 
not toward UA.
Your media software or hardware should support STUN or ICE protocols to work 
correctly via NAT.

Best regards.

Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher mailto:ata...@vanrise.com> >:

Thank you for your reply.

 

ACK is now sent correctly to public IP , yet I have another issue with RBT  
where it’s still sent to private IP .Yet the call media is sent correctly to 
the public IP.

 

There is something strange , as I hear two short different ringing tones then 
complete silence till the called party answer the call and RTP flow normally.

 

How can I handle this case?

 

Thanks,

Ali

 

From: Konstantin Polyakov mailto:piligrim...@mail.ru> > 
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org  ; Ali 
Taher mailto:ata...@vanrise.com> >
Subject: Re: [SR-Users] NATHELPER issue

 

Hello Ali,

ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.

My NATDETECT looks like:

if (nat_uac_test("19")) {

   fix_nated_contact();

}


I call it for requests and responses.

Best regards.
Konstantin

 

Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher"  >
To: 
 >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com 

 >
Content-Type: text/plain; charset="utf-8"

Hello,

 

I'm using Kamailio 4.2 as proxy with nathelper enabled.

 

Yet , the ACK packet sent from the proxy to the origination's private IP.

 

The ACK is sent as reply on the following 200 OK sent from the origination :


 

8m2EJN41BN/6WSIP/2.0 200 OK

From: ;tag=XQBQNjvjgp4Ze

To: ;tag=12033368836000

Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN

Record-Route: 

Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX 

 

CSeq: 125698370 INVITE

Accept: application/sdp

Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE

P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XX.XX
;orig-ioi=MXX

Content-Type: application/sdp

Contact: 

Content-Length: 268

 

v=0

o=- 5838243 5838244 IN IP4

Re: [SR-Users] KEMI: accessing "keyword" variables from core

2018-07-23 Thread Enrico Bandiera

Noticed there's KSR.is_myself

for some reason the search functionality on the KEMI documentation didn't
find it when searching for "myself"

Thanks

2018-07-23 10:53 GMT+02:00 Enrico Bandiera :

> Hi Daniel, thank you, any hint for the keyword "myself"?
>
> 2018-07-20 14:59 GMT+02:00 Daniel-Constantin Mierla :
>
>> Hello,
>>
>> all the keywords should have corresponding variables, in these cases:
>>
>>   - dst_port should be $Rp (received on port)
>>
>>   - src_port should be $sp
>>
>> You can look in the pseudo-variables cookbook on kamailio.org/wiki to
>> confirm.
>>
>> Cheers,
>> Daniel
>>
>> On 20.07.18 12:48, Enrico Bandiera wrote:
>>
>> Hi, anyone how to access keyword variables like dst_port or src_port from
>> kemi? (specifically python in our case).
>>
>> Thanks,
>> Enrico.
>>
>>
>> ___
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>>
>>
>> --
>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
>> www.linkedin.com/in/miconda
>> Kamailio World Conference -- www.kamailioworld.com
>>
>>
>
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Re: [SR-Users] KEMI: accessing "keyword" variables from core

2018-07-23 Thread Enrico Bandiera

Hi Daniel, thank you, any hint for the keyword "myself"?

2018-07-20 14:59 GMT+02:00 Daniel-Constantin Mierla :

> Hello,
>
> all the keywords should have corresponding variables, in these cases:
>
>   - dst_port should be $Rp (received on port)
>
>   - src_port should be $sp
>
> You can look in the pseudo-variables cookbook on kamailio.org/wiki to
> confirm.
>
> Cheers,
> Daniel
>
> On 20.07.18 12:48, Enrico Bandiera wrote:
>
> Hi, anyone how to access keyword variables like dst_port or src_port from
> kemi? (specifically python in our case).
>
> Thanks,
> Enrico.
>
>
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>
>
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> www.linkedin.com/in/miconda
> Kamailio World Conference -- www.kamailioworld.com
>
>
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Re: [SR-Users] NATHELPER issue

2018-07-23 Thread Ali Taher

Thank you for your reply.

 

ACK is now sent correctly to public IP , yet I have another issue with RBT  
where it’s still sent to private IP .Yet the call media is sent correctly to 
the public IP.

 

There is something strange , as I hear two short different ringing tones then 
complete silence till the called party answer the call and RTP flow normally.

 

How can I handle this case?

 

Thanks,

Ali

 

From: Konstantin Polyakov  
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org; Ali Taher 
Subject: Re: [SR-Users] NATHELPER issue

 

Hello Ali,

ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.

My NATDETECT looks like:

if (nat_uac_test("19")) {

   fix_nated_contact();

}


I call it for requests and responses.

Best regards.
Konstantin

 

Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" mailto:ata...@vanrise.com> >
To: mailto:sr-us...@lists.sip-router.org> >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com 
 >
Content-Type: text/plain; charset="utf-8"

Hello,

 

I'm using Kamailio 4.2 as proxy with nathelper enabled.

 

Yet , the ACK packet sent from the proxy to the origination's private IP.

 

The ACK is sent as reply on the following 200 OK sent from the origination :


 

8m2EJN41BN/6WSIP/2.0 200 OK

From: ;tag=XQBQNjvjgp4Ze

To: ;tag=12033368836000

Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN

Record-Route: 

Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX 
 

CSeq: 125698370 INVITE

Accept: application/sdp

Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE

P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XX.XX
;orig-ioi=MXX

Content-Type: application/sdp

Contact: 

Content-Length: 268

 

v=0

o=- 5838243 5838244 IN IP4 BC00.XX

s=-

c=IN IP4 172.16.45.144

t=0 0

a=sendrecv

m=audio 47588 RTP/AVP 18 96

c=IN IP4 172.16.45.144

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

a=maxptime:20

 

Following is the header of the sent ACK packet:

 

Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0

Record-Route: 

Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0

Via: SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc

Max-Forwards: 69

From: ;tag=XQBQNjvjgp4Ze

To: ;tag=12033368836000

Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX 
 

CSeq: 125698370 ACK

Content-Length: 0

 

Where X.X.X.X is Kamailio server public IP.

 

Following is part of my config file :

 

route {

route(NATDETECT);

record_route();

 

if(!mf_process_maxfwd_header("10")) {

sl_send_reply("483", "Too Many Hops");

exit;

}

 

# Maybe some sanity_check() here.

 

if(has_totag()) {

   

if(loose_route()) {

route(DLGURI);

if(!t_relay())

sl_reply_error();

 

exit;

} else {

if(is_method("ACK")) {

route(DLGURI);

if(t_check_trans()) {

t_relay();

}

} else

sl_send_reply("403", "Forbidden");

}

exit;

}



.

}



 

route[NATDETECT] {

#!ifdef WITH_NAT

force_rport();

if (nat_uac_test("19")) {

if (is_method("REGISTER")) {

fix_nated_register();

} else {

add_contact_alias();

}

setflag(FLT_NATS);

}

#!endif

return;

}

 

route[DLGURI] {

#!ifdef WITH_NAT

if(!isdsturiset()) {

 handle_ruri_alias();

}

#!endif

return;

} 

 

Can you please check why the ACK is still sent on private IP ?

 

Thanks

Ali Taher



 

"Ali Taher" https://e.mail.ru/compose?To=ata...@vanrise.com> >

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Re: [SR-Users] Selectively enable / disable cdrs using acc module

2018-07-23 Thread M S

I have tested it. It does not work. It only stops sip transaction
accounting but not the cdrs.

I think i have to add support in native code for dialog level flag or
variable to allow the customization of cdrs...humm, lets see.

Thank you.
On Sat, 21 Jul 2018 at 17:40, M S  wrote:

> Ok, let me try and get back to you.
>
> Thank you.
>
>
> On Sat, 21 Jul 2018 at 16:10, Sergiu Pojoga  wrote:
>
>> I suppose it depends how one generates CDRs, directly from Kamailio or
>> using an external process.
>>
>> Still, not setting the acc log_flag should result in no cdrs. Why don't
>> you give it a try.
>>
>> On Sat, Jul 21, 2018 at 2:16 AM M S  wrote:
>>
>>> Humm, isn’t acc log_flag is for transaction logging, rather then cdrs?
>>>
>>> For cdrs, the acc module has just one relevant module parameter, that is
>>> cdr_enable which enables / disables cdrs for ALL calls.
>>> On Sat, 21 Jul 2018 at 00:01, Sergiu Pojoga  wrote:
>>>
 Assuming your acc log_flag is set to FLT_ACC, it should be as simple as
 not setting or resetting the FLT_ACC flag, depending on the script logic.

 You will probably need to do that for both INVITEs and BYEs


 On Fri, Jul 20, 2018 at 2:48 PM M S  wrote:

> Hi,
>
> I have a kamailio v5.0 setup where cdrs are enabled using acc module.
> Is there a way to enable / disable cdrs on per call basis in kamailio
> script? For example i want to allow cdrs only for offnet calls and disable
> them for onnet calls?
>
> May be a dialog flag configurable in module parameters to signal acc
> module whether or not to generate cdr for current dialog at the end of 
> call?
>
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Re: [SR-Users] Module RTPPROXY param Timeout_Socket not Working

2018-07-23 Thread Yu Boot


Hello.


AFAIK rtpproxy timeout socket requires very special version of rtpproxy 
itself and a patch. Personally, I've gived up to make this feature work.



Now I use dialog-based keepalives to detect and tear down dead sessions.




23.07.2018 9:42, Hamid Hashmi пишет:

waiting for suggestions.

Regards


*Hamid R. Hashmi*

__

www.hrhashmi.blogspot.com 

Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com 
; _hamid2kviii@hotmail.com_


Sype: _hamidrhashmi_


*From:* Hamid Hashmi 
*Sent:* Thursday, July 19, 2018 1:05 PM
*To:* Hamid Hashmi; Kamailio (SER) - Users Mailing List
*Subject:* Re: Module RTPPROXY param Timeout_Socket not Working
I started the RTPproxy with the following command.

|/usr/bin/rtpproxy -u root -A [PUBLIC IP] -F -f -T 45 -i -2 -l 
192.168.3.22 -b -m 35000 -M 65000 -s udp:127.0.0.1:7722 -n 
tcp:127.0.0.1:8000 -d DBUG|



Regards


*Hamid R. Hashmi*

__

www.hrhashmi.blogspot.com 

Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com 
; _hamid2kviii@hotmail.com_


Sype: _hamidrhashmi_


*From:* sr-users  on behalf of 
Hamid Hashmi 

*Sent:* Thursday, July 19, 2018 12:59 PM
*To:* sr-users@lists.kamailio.org
*Subject:* [SR-Users] Module RTPPROXY param Timeout_Socket not Working

Kamailio didn't start listening on IP mentioned in Module RTPPROXY 
parameterTimeout_Socket 
.


Below is my configuration file.

|
|loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", 
"udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_tout", 1) 
modparam("rtpproxy", "rtpproxy_retr", 2) modparam("rtpproxy", 
"timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2;) loadmodule 
"xmlrpc.so" modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", 
"mode", 1) modparam("xmlrpc", "url_match", "^/RPC2")|

|

when I type a command

|$ netstat -nlp|

it only shows SIP listening ports. Kamailio should also start 
listening on|127.0.0.1:8000.|



 Below are the RTPproxy logs, *RTPproxy is also unable to bridge the 
media*


|DBUG:get_command:GLOBAL: received command "19083_4 Uc120,101 
9a1060af660d4202976c419ab5300470 58.65.176.42 4000 
1ea06d1eb0a8443697a08e568045598c;1" 
INFO:rtpp_command_ul_handle:GLOBAL: new session 
9a1060af660d4202976c419ab5300470, tag 
1ea06d1eb0a8443697a08e568045598c;1 requested, type strong 
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new 
session on a port 35012 created, tag 
1ea06d1eb0a8443697a08e568045598c;1 
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: 
pre-filling caller's address with 58.65.176.42:4000 
DBUG:rtpc_doreply:GLOBAL: sending reply "35012 51.38.201.3 " 
DBUG:get_command:GLOBAL: received command "19083_5 Uc120,101 
9a1060af660d4202976c419ab5300470 58.65.176.42 4022 
9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 1ea06d1eb0a8443697a08e568045598c;1 
xmlrpc:http://127.0.0.1:8000/RPC2; INFO:rtpp_command_ul_handle:GLOBAL: 
new session 9a1060af660d4202976c419ab5300470, tag 
9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 requested, type strong 
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new 
session on a port 35014 created, tag 
9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 
ERR:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: *invalid 
socket name 1ea06d1eb0a8443697a08e568045598c;1* 
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: 
pre-filling caller's address with 58.65.176.42:4022 
DBUG:rtpc_doreply:GLOBAL: sending reply "35014 51.38.201.3 " 
DBUG:get_command:GLOBAL: received command "19086_4 D 
9a1060af660d4202976c419ab5300470 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq 
1ea06d1eb0a8443697a08e568045598c" 
INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully 
deleting session 1 on ports 35012/0 
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in 
from callee, 0 in from caller, 0 relayed, 0 dropped 
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in 
from callee, 0 in from caller, 0 relayed, 0 dropped 
INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 
35012/0 is cleaned up 
INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully 
deleting session 1 on ports 35014/0 
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in 
from callee, 0 in from caller, 0 relayed, 0 dropped 
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in 
from callee, 0 in from caller, 0 relayed, 0 dropped 
INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 
35014/0 is cleaned up DBUG:rtpc_doreply:GLOBAL: sending reply "0 " 
DBUG:get_command:GLOBAL: received command

Re: [SR-Users] Module RTPPROXY param Timeout_Socket not Working

2018-07-23 Thread Hamid Hashmi

waiting for suggestions.


Regards


Hamid R. Hashmi

__

www.hrhashmi.blogspot.com

Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com; 
hamid2kv...@hotmail.com

Sype:  hamidrhashmi


From: Hamid Hashmi 
Sent: Thursday, July 19, 2018 1:05 PM
To: Hamid Hashmi; Kamailio (SER) - Users Mailing List
Subject: Re: Module RTPPROXY param Timeout_Socket not Working

I started the RTPproxy with the following command.


/usr/bin/rtpproxy -u root -A [PUBLIC IP] -F -f -T 45 -i -2 -l 192.168.3.22 -b 
-m 35000 -M 65000 -s udp:127.0.0.1:7722 -n tcp:127.0.0.1:8000 -d DBUG



Regards


Hamid R. Hashmi

__

www.hrhashmi.blogspot.com

Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com; 
hamid2kv...@hotmail.com

Sype:  hamidrhashmi


From: sr-users  on behalf of Hamid Hashmi 

Sent: Thursday, July 19, 2018 12:59 PM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Module RTPPROXY param Timeout_Socket not Working


Kamailio didn't start listening on IP mentioned in Module RTPPROXY parameter 
Timeout_Socket.

Below is my configuration file.

loadmodule "rtpproxy.so"
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
modparam("rtpproxy", "rtpproxy_tout", 1)
modparam("rtpproxy", "rtpproxy_retr", 2)
modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2;)

loadmodule "xmlrpc.so"
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "mode", 1)
modparam("xmlrpc", "url_match", "^/RPC2")

when I type a command

$ netstat -nlp

it only shows SIP listening ports. Kamailio should also start listening on 
127.0.0.1:8000.

 Below are the RTPproxy logs, RTPproxy is also unable to bridge the media


DBUG:get_command:GLOBAL: received command "19083_4 Uc120,101 
9a1060af660d4202976c419ab5300470 58.65.176.42 4000 
1ea06d1eb0a8443697a08e568045598c;1"
INFO:rtpp_command_ul_handle:GLOBAL: new session 
9a1060af660d4202976c419ab5300470, tag 1ea06d1eb0a8443697a08e568045598c;1 
requested, type strong
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new session on a 
port 35012 created, tag 1ea06d1eb0a8443697a08e568045598c;1
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: pre-filling 
caller's address with 58.65.176.42:4000
DBUG:rtpc_doreply:GLOBAL: sending reply "35012 51.38.201.3
"
DBUG:get_command:GLOBAL: received command "19083_5 Uc120,101 
9a1060af660d4202976c419ab5300470 58.65.176.42 4022 
9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 1ea06d1eb0a8443697a08e568045598c;1 
xmlrpc:http://127.0.0.1:8000/RPC2;
INFO:rtpp_command_ul_handle:GLOBAL: new session 
9a1060af660d4202976c419ab5300470, tag 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 
requested, type strong
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new session on a 
port 35014 created, tag 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1
ERR:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: invalid socket 
name 1ea06d1eb0a8443697a08e568045598c;1
INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: pre-filling 
caller's address with 58.65.176.42:4022
DBUG:rtpc_doreply:GLOBAL: sending reply "35014 51.38.201.3
"

DBUG:get_command:GLOBAL: received command "19086_4 D 
9a1060af660d4202976c419ab5300470 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq 
1ea06d1eb0a8443697a08e568045598c"
INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully deleting 
session 1 on ports 35012/0
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in from 
callee, 0 in from caller, 0 relayed, 0 dropped
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in from 
callee, 0 in from caller, 0 relayed, 0 dropped
INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 35012/0 
is cleaned up
INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully deleting 
session 1 on ports 35014/0
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in from 
callee, 0 in from caller, 0 relayed, 0 dropped
INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in from 
callee, 0 in from caller, 0 relayed, 0 dropped
INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 35014/0 
is cleaned up
DBUG:rtpc_doreply:GLOBAL: sending reply "0
"
DBUG:get_command:GLOBAL: received command "19086_5 D 
9a1060af660d4202976c419ab5300470 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq 
1ea06d1eb0a8443697a08e568045598c"
INFO:handle_command:GLOBAL: delete request failed: session 
9a1060af660d4202976c419ab5300470, tags 
9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq/1ea06d1eb0a8443697a08e568045598c not found
DBUG:rtpc_doreply:GLOBAL: sending reply "E50

But if I comment out the param "Timeout_Socket" then there is no issue in 
bridging the media.

Re: [SR-Users] Options for SIP traffic duplicating to third party tcp port

2018-07-23 Thread Максим Баранов

Thank you, both siptrace and Homer looks like that is what I need.

2018-07-22 13:43 GMT+03:00 Gholamreza Sabery :

> Hi,
>
> I think it is better for you to also take a look at Homer. It provides a
> free and very nice web-interface to see captured SIP packets and also
> search for them. For more information refer to:
>
> https://github.com/sipcapture/homer
>
>
> On Sun, Jul 22, 2018 at 12:07 PM, Henning Westerholt 
> wrote:
>
>> Am Samstag, 21. Juli 2018, 15:32:44 CEST schrieb Максим Баранов:
>> > Yes, but I am confused by database part. I need just sip traffic
>> mirroring,
>> > database is just an option there? I don’t need database interaction.
>> > > 19 июля 2018 г., в 11:02, Henning Westerholt 
>> написал(а):
>>
>> Hello,
>>
>> ok - probably the siptrace module is then better:
>>
>> 1. Overview
>>
>>The SIPtrace module offer a possibility to store incoming and outgoing
>>SIP messages in a database and/or duplicate to the capturing server
>>(using HEP, the Homer encapsulation protocol, or plain SIP mode)
>>
>> Disable the DB with trace_to_database and then have a look to the
>> trace_to_ip
>> function.
>>
>> Best regards,
>>
>> Henning
>>
>> --
>> If you like my work in the Kamailio project, it would be great if you
>> could
>> consider supporting me on Patreon: https://www.patreon.com/henningw
>>
>> ___
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>>
>
>
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[SR-Users] Forwarding to backup server

2018-07-23 Thread Pali Rohár

Hello, can somebody help me how to configure Kamailio to route/forward
requests to different (backup) server when Kamailio itself is not able
to deliver them?

I have working SIP gateway (on different host/port) which deliver
received SIP messages to XMPP network and I want to achieve:

Kamailio should try to send SIP messages directly to foreign SIP domains
and when it fails (e.g. user not available or foreign SIP is not
running), then forward message to that my SIP <--> XMPP gateway which
will try to deliver it via XMPP.

I guess that Kamailio should be able to configure for such custom
routing/forwarding but I have not figured yet how to do it. I'm using
Kamailio 4.4 with slightly modified default kamailio.cfg config.

-- 
Pali Rohár
pali.ro...@gmail.com


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Re: [SR-Users] Module RTPPROXY param Timeout_Socket not Working

Re: [SR-Users] Module RTPPROXY param Timeout_Socket not Working

Re: [SR-Users] Options for SIP traffic duplicating to third party tcp port

[SR-Users] Forwarding to backup server

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