Hey, thanks for looking into this. Phones register with kamailio and I have configured kamailio to load balance the asterisk servers using round robin algorithm. Phones registered in kamailio will directly send the call to the asterisk servers. From there I will send the call out using other routes.
In the contact field of the sip header, I get the username@private ip instead of the public ip. I need it in the format username@public ip
On 27-Dec-2014, at 3:26 pm, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
I am not that familiar with asterisk to know all the insides SIPURI and SIPCHANNEL(recvip), you have to explain which fields you need from the SIP packet in order to be able to assist further, unless someone else more familiar with asterisk can jump in.
Also, it is not clear how the flow is there, you say you have kamailio behind asterisk servers, does that mean the phone is sending to asterisk first, which then forwards to kamailio?
Cheers, Daniel
On 26/12/14 18:37, Cibin Paul wrote: Hi,
I have a kamailio 4.1 as a gateway and registrar behind asterisk servers. On asterisk, I save the IP address of the originating call parsing the SIPURI or SIPCHANNEL(recvip). In both cases, I am receiving the private ip of the user agent registered with kamailio. Do I need to change anything in kamailio to receive the actual IP address. Please suggest.
Regards Cibin
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://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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users