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
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
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
You can send the info you need to Asterisk using X headers, for example:
Kamailio has the pseudo variable $si which contains the public IP originating the request and you can send that to Asterisk using append_hf.
append_hf("X-orig-IP: $si");
In Asterisk you can access the headers as ${SIP_HEADER(X-orig-IP)}
On Sat, Dec 27, 2014 at 7:00 PM, Cibin Paul paul_cibin@me.com wrote:
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
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 Luis. I will check today and inform you.
Regards Cibin
On 28-Dec-2014, at 9:06 am, Luis Jimenez ljjimenez@gmail.com wrote:
You can send the info you need to Asterisk using X headers, for example:
Kamailio has the pseudo variable $si which contains the public IP originating the request and you can send that to Asterisk using append_hf.
append_hf("X-orig-IP: $si");
In Asterisk you can access the headers as ${SIP_HEADER(X-orig-IP)}
On Sat, Dec 27, 2014 at 7:00 PM, Cibin Paul paul_cibin@me.com wrote: 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
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
Thanks a lot Luis. It worked like a charm.
Regards Cibin
On 28-Dec-2014, at 9:50 am, Cibin Paul paul_cibin@me.com wrote:
Thanks Luis. I will check today and inform you.
Regards Cibin
On 28-Dec-2014, at 9:06 am, Luis Jimenez <ljjimenez@gmail.com mailto:ljjimenez@gmail.com> wrote:
You can send the info you need to Asterisk using X headers, for example:
Kamailio has the pseudo variable $si which contains the public IP originating the request and you can send that to Asterisk using append_hf.
append_hf("X-orig-IP: $si");
In Asterisk you can access the headers as ${SIP_HEADER(X-orig-IP)}
On Sat, Dec 27, 2014 at 7:00 PM, Cibin Paul <paul_cibin@me.com mailto:paul_cibin@me.com> wrote: 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 mailto: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 mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda http://www.linkedin.com/in/miconda
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 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 mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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 mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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
-
Cibin Paul -
Daniel-Constantin Mierla -
Luis Jimenez