27 Oct
2015
27 Oct
'15
11:36 a.m.
Hello,
i've DID provider that require to register to it's server. I've been able
to configure this provider using kamailio and uac and to route all inbound
calls from DID provider to internal asterisk media server. All is working
great.
Now i need help to configure call routing from internal asterisk to the
external DID provider.
I've done something like:
route[LOCATION] {
if(is_method("INVITE")) {
route(FROMASTERISK);
route(TOASTERISK);
exit;
}
[...]
}
route[FROMASTERISK] {
if (ds_is_from_list()) {
xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server
Cluster\n");
rtpengine_manage("direction=internal direction=external replace-origin
replace-session-connection trust-address");
$ru = "sip:TESTING_NUMBER@EXTERNAL_PROVIDER_IP";
route(RELAY);
exit;
}
}
It seems that when i route the call outbound, i get a 407. So i would like
to react to this using uac_auth() as described here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
Is this the right way? Can someone help me, giving hints and some working
example?
Using kamailio 4.3.3 with rtpengine 4.1.0
Best regards,
Bruno
--
The life is short.. live at your best!
27 Oct
27 Oct
3:41 p.m.
New subject: Routing outgoing call from asterisk to registered SIP provider
On Tuesday 27 October 2015 07:36:15 Bruno Salzano wrote:
It seems that when i route the call outbound, i get a
407. So i would like
to react to this using uac_auth() as described here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
Is this the right way? Can someone help me, giving hints and some working
example?
Based on the info you have given I'd say this is correct. The working code is
in the URL you provide. Adding a specific failure route in the FROMASTERISK
when ds_is_from_list matches to the uac_auth() function is the way to go.
4:27 p.m.
New subject: Routing outgoing call from asterisk to registered SIP provider
Thank you. I'll try and let you know.
Bruno
2015-10-27 11:41 GMT+01:00 Daniel Tryba <d.tryba(a)pocos.nl>nl>:
On Tuesday 27 October 2015 07:36:15 Bruno Salzano
wrote:
--
The life is short.. live at your best!
It seems that when i route the call outbound, i
get a 407. So i would
like
to react to this using uac_auth() as described
here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
Is this the right way? Can someone help me, giving hints and some working
example?
Based on the info you have given I'd say this is correct. The working code
is
in the URL you provide. Adding a specific failure route in the FROMASTERISK
when ds_is_from_list matches to the uac_auth() function is the way to go.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
28 Oct
28 Oct
3:44 a.m.
New subject: Routing outgoing call from asterisk to registered SIP provider
Hello Daniel,
Il 27/10/15 11:41, Daniel Tryba ha scritto:
On Tuesday 27 October 2015 07:36:15 Bruno Salzano
wrote:
I got it
working. It required me to enable the track_cseq_updates in
dialog module and use uac_auth(). Actually my config is the following:
route[FROMASTERISK] {
if (ds_is_from_list()) {
xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server
Cluster\n");
rtpengine_manage("direction=internal direction=external
replace-origin replace-session-connection trust-address");
uac_replace_from("", "sip:NUMBER@HOST");
route(RELAY);
exit;
}
}
failure_route[TRUNKAUTH] {
xlog("L_INFO","TRUNKAUTH FAILURE ROUTE");
if (t_is_canceled()) {
exit;
}
if(t_check_status("401|407")) {
xlog("L_INFO", "Got 401/407 sending uac_auth() and
t_relay()");
$avp(auser) = "USERNAME";
$avp(apass) = "SECRET";
$avp(arealm) = "REALM";
uac_auth();
t_relay();
exit;
}
}
Now I've just last question on this topic: what's the best way to
replace credentials (USERNAME, SECRET, REALM) from db?
I've to manage also the uac_replace_from.
What I want is to avoid to change the configuration everytime a new
trunk is added.
Best Regards,
Bruno
It seems that when i route the call outbound, i
get a 407. So i would like
to react to this using uac_auth() as described here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
Is this the right way? Can someone help me, giving hints and some working
example?
Based on the info you have given I'd say this is correct. The working
code is
in the URL you provide. Adding a specific failure route in the FROMASTERISK
when ds_is_from_list matches to the uac_auth() function is the way to go.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
5:45 p.m.
New subject: Routing outgoing call from asterisk to registered SIP provider
On 27/10/15 23:44, Bruno Salzano wrote:
Hello Daniel,
Il 27/10/15 11:41, Daniel Tryba ha scritto:
You can use sqlops to load values from a database table. Or htable,
which has the benefit of storing it in memory at runtime, but you can
still issue reload commands when new values are set.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat
On Tuesday 27 October 2015 07:36:15 Bruno Salzano
wrote:
I got it working. It required me to enable the track_cseq_updates in
dialog module and use uac_auth(). Actually my config is the following:
route[FROMASTERISK] {
if (ds_is_from_list()) {
xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server
Cluster\n");
rtpengine_manage("direction=internal direction=external
replace-origin replace-session-connection trust-address");
uac_replace_from("", "sip:NUMBER@HOST");
route(RELAY);
exit;
}
}
failure_route[TRUNKAUTH] {
xlog("L_INFO","TRUNKAUTH FAILURE ROUTE");
if (t_is_canceled()) {
exit;
}
if(t_check_status("401|407")) {
xlog("L_INFO", "Got 401/407 sending uac_auth() and
t_relay()");
$avp(auser) = "USERNAME";
$avp(apass) = "SECRET";
$avp(arealm) = "REALM";
uac_auth();
t_relay();
exit;
}
}
Now I've just last question on this topic: what's the best way to
replace credentials (USERNAME, SECRET, REALM) from db?
I've to manage also the uac_replace_from.
What I want is to avoid to change the configuration everytime a new
trunk is added.
It seems that when i route the call outbound, i
get a 407. So i
would like
to react to this using uac_auth() as described here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
Is this the right way? Can someone help me, giving hints and some
working
example?
Based on the info you have given I'd say this is correct. The
working
code is
in the URL you provide. Adding a specific failure route in the
FROMASTERISK
when ds_is_from_list matches to the uac_auth() function is the way to
go.
29 Oct
29 Oct
10:08 a.m.
New subject: Routing outgoing call from asterisk to registered SIP provider
Il 28/10/15 13:45, Daniel-Constantin Mierla ha scritto:
On 27/10/15 23:44, Bruno Salzano wrote:
htable seems the best
solution. Thank you!
Regards,
Bruno
Hello Daniel,
Il 27/10/15 11:41, Daniel Tryba ha scritto:
You can use sqlops to load values from a database table. Or htable,
which has the benefit of storing it in memory at runtime, but you can
still issue reload commands when new values are set. On Tuesday 27 October 2015 07:36:15 Bruno Salzano
wrote:
I got it working. It required me to enable the track_cseq_updates in
dialog module and use uac_auth(). Actually my config is the following:
route[FROMASTERISK] {
if (ds_is_from_list()) {
xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server
Cluster\n");
rtpengine_manage("direction=internal direction=external
replace-origin replace-session-connection trust-address");
uac_replace_from("", "sip:NUMBER@HOST");
route(RELAY);
exit;
}
}
failure_route[TRUNKAUTH] {
xlog("L_INFO","TRUNKAUTH FAILURE ROUTE");
if (t_is_canceled()) {
exit;
}
if(t_check_status("401|407")) {
xlog("L_INFO", "Got 401/407 sending uac_auth() and
t_relay()");
$avp(auser) = "USERNAME";
$avp(apass) = "SECRET";
$avp(arealm) = "REALM";
uac_auth();
t_relay();
exit;
}
}
Now I've just last question on this topic: what's the best way to
replace credentials (USERNAME, SECRET, REALM) from db?
I've to manage also the uac_replace_from.
What I want is to avoid to change the configuration everytime a new
trunk is added.
It seems that when i route the call outbound, i
get a 407. So i
would like
to react to this using uac_auth() as described here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
Is this the right way? Can someone help me, giving hints and some
working
example?
Based on the info you have given I'd say this is correct. The
working
code is
in the URL you provide. Adding a specific failure route in the
FROMASTERISK
when ds_is_from_list matches to the uac_auth() function is the way to
go.
Cheers,
Daniel
27 Oct
27 Oct
7:30 p.m.
New subject: Routing outgoing call from asterisk to registered SIP provider
It seems that when i route the call outbound, i get a
407. So i would like
to react to this using uac_auth() as described here:
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
Is this the right way? Can someone help me, giving hints and some working
example?
Using uac_auth() is the perfect way to send calls to external
provider, if you want to authorise the user. If your provider provides
IP based auth, you don't need uac_auth() and you can simply terminate
using the PSTN gw.
I am not if this answers your question fully,. I myself a newbie here.
Tried and got a code working both for PSTN with/without uac_auth.
regards
Ganesh Kumar
3103
days inactive
3105
days old
6 comments
4 participants
participants (4)
-
Bruno Salzano -
Daniel Tryba -
Daniel-Constantin Mierla -
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