Re: [SR-Users] Translate calling/called party based on username to telephone number base id.

2014-04-28 Thread Daniel-Constantin Mierla

Hello,

On 22/04/14 17:33, Helena Garcia-Nieto wrote:


Hi all.

I want to thank you all for the great info and help that you provide 
in this mailing list. I am just starting to work with kamailio and 
still learning.


I would like some help to point me on the right direction on our new 
challenge:


We have a kamailio server acting basically as location server and call 
proxy. We are connected to a PSTN gw to reach the outside world. It is 
working fine but now the client requested some changes that may imply 
we work with username base subscribers (user1. Characters based) at 
kamailio side but we still need to convert these usernames to numbers 
when sending the call to the PSTN gw. Username-number should be fix 
and can be stored in a table on the db.


We would need to lookup userpart of all the header (from, to, 
contact…) and substitute incoming username with the peer number for 
calls from the app to pstn. And translate numbers to usernames on the 
calls from pstn to the apps.


I am reading uac module  and the sqlops module to search in the db 
before the substitution but I am not sure if that is the best approach 
to the problem.


Could anyone point me in the right direction or suggest a  better way 
to do that? Have anyone already implemented this type of 
modifications? I am a bit worried for the dialog consistency although 
I’ve read the uac module should take care of it


yes, uac module can take care of replacing from/to headers. You don't 
need to touch contact header, that's irrelevant for the gateway in terms 
of origin/destination IDs.


sqlops can be used for loading the values from database, so you are 
looking in the right direction.


Cheers,
Daniel


Thanks in advanced

Helena

*From:*Helena Garcia-Nieto [mailto:helena.gni...@morodo.co.uk]
*Sent:* martes, 10 de diciembre de 2013 16:19
*To:* 'Kamailio (SER) - Users Mailing List'
*Cc:* helena.gni...@morodo.co.uk
*Subject:* 500 I'm terribly sorry error

Hello,

Thanks in advanced for the help. I am almost new with kamailio and 
still struggling through silly problems so please forgive me if the 
solution is so obvious.


I have a network like

Xlitle -- Kamailio -- GW

The GW is more or less out of my reach for changing the behaivour.

As devices I have xlitle

Kamailio is on version 4.0.2

I’ve changed only few things from the default config file. Add mysql 
support, auth, userlocdb, pstngw.


For this part, gw routing , I’ve defined gw ip and port inside the 
PSTN definition like:


#!ifdef WITH_PSTN

# PSTN GW Routing

#

# - pstn.gw_ip: valid IP or hostname as string value, example:

# pstn.gw_ip = 10.0.0.101 desc My PSTN GW Address

#

# - by default is empty to avoid misrouting

pstn.gw_ip =  desc PSTN GW Address

pstn.gw_port =  desc PSTN GW Port

iskratel.gw_ip = 10.XX.XX.XX

iskratel.gw_port = 5060

#!endif

I route the calls with:

   route(ISKRATEL);

And defined a routing function

route[ISKRATEL] {

#!ifdef WITH_PSTN

# check if ISKRATEL GW IP is defined

if (strempty($sel(cfg_get.iskratel.gw_ip))) {

xlog(SCRIPT: PSTN rotuing enabled but iskratel.gw_ip 
not defined\n);


return;

}

# only local users allowed to call

if(from_uri!=myself) {

sl_send_reply(403, Not Allowed);

exit;

}

if (strempty($sel(cfg_get.iskratel.gw_port))) {

$ru = sip: + $rU + @ + $sel(cfg_get.iskratel.gw_ip);

   } else {

$ru = sip: + $rU + @ + 
$sel(cfg_get.iskratel.gw_ip) + :


+ $sel(cfg_get.iskratel.gw_port);

}

# Add profix to ISKRATEL: A99901

subst_uri('/^sip:(.*)/sip:A99901\1/i'); # add A99901

route(RELAY);

exit;

#!endif

return;

}

I can see that the code goes all the way to the route(relay) but all I 
can see is the 500 I’m terribly sorry, server error occurred (7/SL) 
and a second response 500 I’m terribly sorry, server error occurred (7/TM)


I’ve tried defining the IP as the given pstn.gw_ip and route(pstn) 
without changing anything on the pst default routing but the response 
of the server is the same.


I cannot see any special error on the logs.

I have the exact same config for this part, in another test server and 
the calls go to the gw without reporting any error.


I’ll appreciate any help from you! Thanks in advanced for your time

Helena



___
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://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org

[SR-Users] Translate calling/called party based on username to telephone number base id.

2014-04-22 Thread Helena Garcia-Nieto
Hi all.

 

I want to thank you all for the great info and help that you provide in this
mailing list. I am just starting to work with kamailio and still learning.

 

I would like some help to point me on the right direction on our new
challenge:

 

We have a kamailio server acting basically as location server and call
proxy. We are connected to a PSTN gw to reach the outside world. It is
working fine but now the client requested some changes that may imply we
work with username base subscribers (user1. Characters based) at kamailio
side but we still need to convert these usernames to numbers when sending
the call to the PSTN gw. Username-number should be fix and can be stored in
a table on the db.

 

We would need to lookup userpart of all the header (from, to, contact.) and
substitute incoming username with the peer number for calls from the app to
pstn. And translate numbers to usernames on the calls from pstn to the apps.


 

I am reading uac module  and the sqlops module to search in the db before
the substitution but I am not sure if that is the best approach to the
problem. 

 

Could anyone point me in the right direction or suggest a  better way to do
that? Have anyone already implemented this type of modifications? I am a bit
worried for the dialog consistency although I've read the uac module should
take care of it

 

Thanks in advanced

 

Helena

From: Helena Garcia-Nieto [mailto:helena.gni...@morodo.co.uk] 
Sent: martes, 10 de diciembre de 2013 16:19
To: 'Kamailio (SER) - Users Mailing List'
Cc: helena.gni...@morodo.co.uk
Subject: 500 I'm terribly sorry error

 

 

Hello,

 

Thanks in advanced for the help. I am almost new with kamailio and still
struggling through silly problems so please forgive me if the solution is so
obvious.

 

I have a network like

 

Xlitle -- Kamailio -- GW

 

The GW is more or less out of my reach for changing the behaivour.

As devices I have xlitle

 

Kamailio is on version 4.0.2

 

I've changed only few things from the default config file. Add mysql
support, auth, userlocdb, pstngw.

 

For this part, gw routing , I've defined gw ip and port inside the PSTN
definition like:

#!ifdef WITH_PSTN

# PSTN GW Routing

#

# - pstn.gw_ip: valid IP or hostname as string value, example:

# pstn.gw_ip = 10.0.0.101 desc My PSTN GW Address

#

# - by default is empty to avoid misrouting

pstn.gw_ip =  desc PSTN GW Address

pstn.gw_port =  desc PSTN GW Port

 

 

iskratel.gw_ip = 10.XX.XX.XX

iskratel.gw_port = 5060

 

#!endif

 

I route the calls with:

   route(ISKRATEL);

 

And defined a routing function

route[ISKRATEL] {

#!ifdef WITH_PSTN

# check if ISKRATEL GW IP is defined

if (strempty($sel(cfg_get.iskratel.gw_ip))) {

xlog(SCRIPT: PSTN rotuing enabled but iskratel.gw_ip not
defined\n);

return;

}

 

 

# only local users allowed to call

if(from_uri!=myself) {

sl_send_reply(403, Not Allowed);

exit;

}

 

if (strempty($sel(cfg_get.iskratel.gw_port))) {

$ru = sip: + $rU + @ + $sel(cfg_get.iskratel.gw_ip);

   } else {

$ru = sip: + $rU + @ + $sel(cfg_get.iskratel.gw_ip) +
:

+ $sel(cfg_get.iskratel.gw_port);

}

 

 

 

# Add profix to ISKRATEL: A99901

subst_uri('/^sip:(.*)/sip:A99901\1/i'); # add A99901

 

route(RELAY);

exit;

#!endif

 

return;

}

 

 

I can see that the code goes all the way to the route(relay) but all I can
see is the 500 I'm terribly sorry, server error occurred (7/SL) and a second
response 500 I'm terribly sorry, server error occurred (7/TM)

 

I've tried defining the IP as the given pstn.gw_ip and route(pstn) without
changing anything on the  pst default routing but the response of the server
is the same.

 

I cannot see any special error on the logs.

 

I have the exact same config for this part, in another test server and the
calls go to the gw without reporting any error.

 

I'll appreciate any help from you! Thanks in advanced for your time

 

Helena

___
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