Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication. Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
- Call coming from FS on kamailio - Rewrite the from-uri (so the provider receives calls from the registered username) - modify the to-domain part to contain the IP address of the provider - set the $du to ip of the provider, and t_relay() the call. - Most likely the Provider would say Proxy-Auth required..that can be caught in failure_route[] - There you can call the uac_auth() function to have username.password attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth() - once this function is successful send the INVITE again to the provider.
Last three steps can be the following snippet of code(reference from here http://opensips.org/pipermail/users/2010-August/013947.html):
failure_route[2] { if (t_check_status("40[17]")) { xlog("got challenged \n"); if (uac_auth()) { xlog("auth was succesful \n"); t_relay("udp:ip_addr:5060"); //provider's IP_ADDR } }
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran alijibran@vividtech.io wrote:
Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ _______________________________________________ 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
Thanks for the awesome detailed explanation :)
I talked to Voipfone(trunk) and they only allow registered endpoints to make/receive calls. So I can't do IP Auth as of now.
I'll try the other method by rewriting $fu and $du. Hopefully that'll work.
Thanks for the help again.
AJ
On 30-Apr-2015, at 9:18 am, SamyGo govoiper@gmail.com wrote:
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication. Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
- Call coming from FS on kamailio
- Rewrite the from-uri (so the provider receives calls from the registered username)
- modify the to-domain part to contain the IP address of the provider
- set the $du to ip of the provider, and t_relay() the call.
- Most likely the Provider would say Proxy-Auth required..that can be caught in failure_route[]
- There you can call the uac_auth() function to have username.password attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
- once this function is successful send the INVITE again to the provider.
Last three steps can be the following snippet of code(reference from here):
failure_route[2] { if (t_check_status("40[17]")) { xlog("got challenged \n"); if (uac_auth()) { xlog("auth was succesful \n"); t_relay("udp:ip_addr:5060"); //provider's IP_ADDR } }
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran alijibran@vividtech.io wrote:
Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
#!ifdef WITH_FREESWITCH
if(is_method("INVITE") && route(FROMFREESWITCH))) {
xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call from FreeSWITCH needs to be sent TOVOIP \n");
route(TOVOIP);
t_on_failure("F_VOIP");
exit;
}
#!endif
route[TOVOIP] {
xlog("L_INFO","ALERT: $fu to $tu ");
$fU="XXXXXX";
$td="sip.voipfone.net";
$du="sip:XXXXXXX@sip.voipfone.net";
t_relay();
}
failure_route[F_VOIP] {
uac_auth();
xlog("L_INFO","ALERT: IN FAIL");
}
I tried this but it never makes it to the failure branch. Im a newbie to kamailio and still working around the scripting. Can you please help me out here to where I am making the mistake?
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of SamyGo Sent: Thursday, April 30, 2015 9:18 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] UAC Module
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication.
Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
- Call coming from FS on kamailio
- Rewrite the from-uri (so the provider receives calls from the registered username)
- modify the to-domain part to contain the IP address of the provider
- set the $du to ip of the provider, and t_relay() the call.
- Most likely the Provider would say Proxy-Auth required..that can be caught in failure_route[]
- There you can call the uac_auth() function to have username.password attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
- once this function is successful send the INVITE again to the provider.
Last three steps can be the following snippet of code(reference from here http://opensips.org/pipermail/users/2010-August/013947.html ):
failure_route[2] { if (t_check_status("40[17]")) { xlog("got challenged \n"); if (uac_auth()) { xlog("auth was succesful \n"); t_relay("udp:ip_addr:5060"); //provider's IP_ADDR } }
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran <alijibran@vividtech.io mailto:alijibran@vividtech.io > wrote:
Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com
t_on_failure("F_VOIP") to be used before t_relay(); That will arm the call to go to F_VOIP on failure responses.
On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran alijibran@vividtech.io wrote:
#!ifdef WITH_FREESWITCH
if(is_method("INVITE") && route(FROMFREESWITCH))) { xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call fromFreeSWITCH needs to be sent TOVOIP \n");
route(TOVOIP); t_on_failure("F_VOIP"); exit; }#!endif
route[TOVOIP] {
xlog("L_INFO","ALERT: $fu to $tu "); $fU="XXXXXX"; $td="sip.voipfone.net"; $du="sip:XXXXXXX@sip.voipfone.net"; t_relay();}
failure_route[F_VOIP] {
uac_auth(); xlog("L_INFO","ALERT: IN FAIL");}
I tried this but it never makes it to the failure branch. Im a newbie to kamailio and still working around the scripting. Can you please help me out here to where I am making the mistake?
*From:* sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *SamyGo *Sent:* Thursday, April 30, 2015 9:18 AM *To:* Kamailio (SER) - Users Mailing List *Subject:* Re: [SR-Users] UAC Module
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication.
Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
Call coming from FS on kamailio
Rewrite the from-uri (so the provider receives calls from the
registered username)
modify the to-domain part to contain the IP address of the provider
set the $du to ip of the provider, and t_relay() the call.
Most likely the Provider would say Proxy-Auth required..that can be
caught in failure_route[]
- There you can call the uac_auth() function to have username.password
attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
- once this function is successful send the INVITE again to the provider.
Last three steps can be the following snippet of code(reference from here http://opensips.org/pipermail/users/2010-August/013947.html):
failure_route[2] {
if (t_check_status("40[17]")) { xlog("got challenged \n"); if (uac_auth()) { xlog("auth was succesful \n"); t_relay("udp:ip_addr:5060"); //provider's IP_ADDR }}
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran alijibran@vividtech.io wrote:
Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ _______________________________________________ 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
This email is free from viruses and malware because avast! Antivirus http://www.avast.com/ protection is active.
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
Perfect. Yeah got the working.
Just one last issue. I don’t think this is rewriting the header. When I log the header again after the changes it still shows me the old values.
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of SamyGo Sent: Thursday, April 30, 2015 6:50 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] UAC Module
t_on_failure("F_VOIP") to be used before t_relay();
That will arm the call to go to F_VOIP on failure responses.
On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran <alijibran@vividtech.io mailto:alijibran@vividtech.io > wrote:
#!ifdef WITH_FREESWITCH
if(is_method("INVITE") && route(FROMFREESWITCH))) {
xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call from FreeSWITCH needs to be sent TOVOIP \n");
route(TOVOIP);
t_on_failure("F_VOIP");
exit;
}
#!endif
route[TOVOIP] {
xlog("L_INFO","ALERT: $fu to $tu ");
$fU="XXXXXX";
$td="sip.voipfone.net http://sip.voipfone.net ";
$du="sip:XXXXXXX@sip.voipfone.net mailto:sip%3AXXXXXXX@sip.voipfone.net ";
t_relay();
}
failure_route[F_VOIP] {
uac_auth();
xlog("L_INFO","ALERT: IN FAIL");
}
I tried this but it never makes it to the failure branch. Im a newbie to kamailio and still working around the scripting. Can you please help me out here to where I am making the mistake?
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org mailto:sr-users-bounces@lists.sip-router.org ] On Behalf Of SamyGo Sent: Thursday, April 30, 2015 9:18 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] UAC Module
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication.
Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
- Call coming from FS on kamailio
- Rewrite the from-uri (so the provider receives calls from the registered username)
- modify the to-domain part to contain the IP address of the provider
- set the $du to ip of the provider, and t_relay() the call.
- Most likely the Provider would say Proxy-Auth required..that can be caught in failure_route[]
- There you can call the uac_auth() function to have username.password attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
- once this function is successful send the INVITE again to the provider.
Last three steps can be the following snippet of code(reference from here http://opensips.org/pipermail/users/2010-August/013947.html ):
failure_route[2] { if (t_check_status("40[17]")) { xlog("got challenged \n"); if (uac_auth()) { xlog("auth was succesful \n"); t_relay("udp:ip_addr:5060"); //provider's IP_ADDR } }
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran <alijibran@vividtech.io mailto:alijibran@vividtech.io > wrote:
Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_____
This email is free from viruses and malware because avast! Antivirus http://www.avast.com/ protection is active.
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com
I'd like you to google around, there is a function available from another module which will apply the changes in SIP Message.
On Thu, Apr 30, 2015 at 9:51 AM, Ali Jibran alijibran@vividtech.io wrote:
Perfect. Yeah got the working.
Just one last issue. I don’t think this is rewriting the header. When I log the header again after the changes it still shows me the old values.
*From:* sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *SamyGo *Sent:* Thursday, April 30, 2015 6:50 PM
*To:* Kamailio (SER) - Users Mailing List *Subject:* Re: [SR-Users] UAC Module
t_on_failure("F_VOIP") to be used before t_relay();
That will arm the call to go to F_VOIP on failure responses.
On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran alijibran@vividtech.io wrote:
#!ifdef WITH_FREESWITCH
if(is_method("INVITE") && route(FROMFREESWITCH))) { xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call fromFreeSWITCH needs to be sent TOVOIP \n");
route(TOVOIP); t_on_failure("F_VOIP"); exit; }#!endif
route[TOVOIP] {
xlog("L_INFO","ALERT: $fu to $tu "); $fU="XXXXXX"; $td="sip.voipfone.net"; $du="sip:XXXXXXX@sip.voipfone.net"; t_relay();}
failure_route[F_VOIP] {
uac_auth(); xlog("L_INFO","ALERT: IN FAIL");}
I tried this but it never makes it to the failure branch. Im a newbie to kamailio and still working around the scripting. Can you please help me out here to where I am making the mistake?
*From:* sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *SamyGo *Sent:* Thursday, April 30, 2015 9:18 AM *To:* Kamailio (SER) - Users Mailing List *Subject:* Re: [SR-Users] UAC Module
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication.
Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
Call coming from FS on kamailio
Rewrite the from-uri (so the provider receives calls from the
registered username)
modify the to-domain part to contain the IP address of the provider
set the $du to ip of the provider, and t_relay() the call.
Most likely the Provider would say Proxy-Auth required..that can be
caught in failure_route[]
- There you can call the uac_auth() function to have username.password
attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
- once this function is successful send the INVITE again to the provider.
Last three steps can be the following snippet of code(reference from here http://opensips.org/pipermail/users/2010-August/013947.html):
failure_route[2] {
if (t_check_status("40[17]")) { xlog("got challenged \n"); if (uac_auth()) { xlog("auth was succesful \n"); t_relay("udp:ip_addr:5060"); //provider's IP_ADDR }}
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran alijibran@vividtech.io wrote:
Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ _______________________________________________ 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
[image: Image removed by sender.] http://www.avast.com/
This email is free from viruses and malware because avast! Antivirus http://www.avast.com/ protection is active.
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
This email is free from viruses and malware because avast! Antivirus http://www.avast.com/ protection is active.
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
One more thing may be useful for you. If you will get an error with cseq numder when provider send 401/407 message- usedialog module. It resole an issuevwith cseq( read documentation) 30.04.2015 18:23 пользователь "SamyGo" govoiper@gmail.com написал:
I'd like you to google around, there is a function available from another
module which will apply the changes in SIP Message.
On Thu, Apr 30, 2015 at 9:51 AM, Ali Jibran alijibran@vividtech.io
wrote:
Perfect. Yeah got the working.
Just one last issue. I don’t think this is rewriting the header. When I
log the header again after the changes it still shows me the old values.
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf
Of SamyGo
Sent: Thursday, April 30, 2015 6:50 PM
To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] UAC Module
t_on_failure("F_VOIP") to be used before t_relay();
That will arm the call to go to F_VOIP on failure responses.
On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran alijibran@vividtech.io
wrote:
#!ifdef WITH_FREESWITCH
if(is_method("INVITE") && route(FROMFREESWITCH))) { xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call from
FreeSWITCH needs to be sent TOVOIP \n");
route(TOVOIP); t_on_failure("F_VOIP"); exit; }#!endif
route[TOVOIP] {
xlog("L_INFO","ALERT: $fu to $tu "); $fU="XXXXXX"; $td="sip.voipfone.net"; $du="sip:XXXXXXX@sip.voipfone.net"; t_relay();}
failure_route[F_VOIP] {
uac_auth(); xlog("L_INFO","ALERT: IN FAIL");}
I tried this but it never makes it to the failure branch. Im a newbie
to kamailio and still working around the scripting. Can you please help me out here to where I am making the mistake?
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf
Of SamyGo
Sent: Thursday, April 30, 2015 9:18 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] UAC Module
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE
with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication.
Thats why its usually recommended to go with IP-Authentication only.
Send INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
Call coming from FS on kamailio
Rewrite the from-uri (so the provider receives calls from the
registered username)
modify the to-domain part to contain the IP address of the provider
set the $du to ip of the provider, and t_relay() the call.
Most likely the Provider would say Proxy-Auth required..that can be
caught in failure_route[]
- There you can call the uac_auth() function to have username.password
attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
- once this function is successful send the INVITE again to the
provider.
Last three steps can be the following snippet of code(reference from
here):
failure_route[2] {
if (t_check_status("40[17]")) { xlog("got challenged \n"); if (uac_auth()) { xlog("auth was succesful \n"); t_relay("udp:ip_addr:5060"); //provider's IP_ADDR }}
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran alijibran@vividtech.io
wrote:
Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with
the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from
FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks. AJ _______________________________________________ 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
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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
This email is free from viruses and malware because avast! Antivirus
protection is active.
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
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
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