Just using your configuration file for minimal changes
1- Add this line in the defines
#!define WITH_PSTN
2- Change the lines where PSTN _ GW IP/PORT are defined
#!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 Asterisk SERVER Address"
pstn.gw_port = "7160"
#
# - by default is empty to avoid misrouting
#!endif
3- Change the condition accroding to your usage in [PSTN] route
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
return;
I think this should work ideally, if nothing has been changed from the original configuration file.
Regards,
Sammy.
On Thu, Nov 24, 2011 at 4:28 PM, arif tuhin
<etothepowerpi@hotmail.com> wrote:
kamailio is important because it also drives several mediaproxy. I will try that and post any update. Thank you.
Date: Thu, 24 Nov 2011 16:25:00 +0500
From:
govoiper@gmail.com
To:
sr-users@lists.sip-router.orgSubject: Re: [SR-Users] Problem to relay sip message recieved port 9090 to asterisk port 7160
You can take sip trace using sipgrep/ngrep or tcpdump/tshark. Also I was thinking if its just a matter of port forwarding then why deploy Kamailio why not just use two/three iptables lines and forward to asterisk IP:PORT !!?
Hhow about you follow some asterisk-kamailio integration post and change the ip/port of asterisk from there and you'll be ready to rock in just half an hour or so !!
Regards,
Sammy.
On Thu, Nov 24, 2011 at 3:08 PM, arif tuhin
<etothepowerpi@hotmail.com> wrote:
How can i get the Sip trace? if you can provide me any details it will be helpful. Also i have attached kamailio.cfg with the mail. Thank you.
Date: Tue, 22 Nov 2011 18:18:27 +0500
From:
govoiper@gmail.com
To:
sr-users@lists.sip-router.orgSubject: Re: [SR-Users] Problem to relay sip message recieved port 9090 to asterisk port 7160
Hi,
Can you capture a sip trace on Kamailio server for this, also paste your configuration file ! so someone can figure out.
Regards,
Sammy.
On Tue, Nov 22, 2011 at 5:14 PM, arif tuhin
<etothepowerpi@hotmail.com> wrote:
I'm Trying to relay sip messages from sip client to asterisk. Kamailio should accept the msg on 9090 udp port and it will forward the sip message to 7160 udp port of Asterisk.
I thought the default cfg file will be enough to accomplish this simple task. But kamailio is receiving the msg ok but its not forwarding the msg. Another interesting thing is if i enable tls and give a tls port of kamailio , it easily forwards and the client registers ok.
With Best Regards
_______________________________________________
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
_______________________________________________
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
_______________________________________________
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