Thank you for your answer.
The problem I have is with internet router doing to PAT to SIP port.
I am already advertising public IP but unfortunately I cant know the public
port I am using.
2015-12-28 18:17 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
> AFAIK bye is usually sent to the address stored in record_route. Try
> setting changing record_route() to
> record_route_preset("PUBLICIP:5060;nat=yes:)
>
> 2015-12-23 16:28 GMT+02:00 Nelson Migliaro <eng.migli...@gmail.com>:
>
>>
>> Hello,
>>
>> I am running Kamailio behind NAT.
>>
>> Kanailio has a private IP and I am relaying NAT to internet router.
>>
>> I am using:
>>
>> - #!define WITH_NAT
>> - listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
>>
>> - Patched RTP proxy including the advertise option
>>
>> And everything goes fine. I can make calls and have two way audio.
>>
>> The problem begins when the callee ends the call. BYE is not received in
>> Kamailio (caller)
>>
>> I included the public IP using "add_contact_alias" because
>> "set_contact_alias" was not adding the public IP. I included this in in
>> NATDETECT (pre loaded router)
>>
>> if(is_first_hop()) {
>> xlog("L_NOTICE","Metodo: $rm \n");
>> xlog("L_NOTICE","is first hop\n");
>> #set_contact_alias();
>> if (!add_contact_alias("PUBLIC-IP", "$Rp", "udp")) {
>> xlog("L_ERR", "Error in aliasing contact $ct\n");
>> send_reply("400", "Bad request");
>> exit;
>> }
>> }
>>
>> I think the problem is related to destination that BYE is sent by the
>> vendor. From what I see IP and port is taken from advertised in contact
>> (PUBLIC-IP and 5060).
>> The problem is that internet router changes the source port.
>>
>> Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
>>
>> --- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp", "udp") in
>> order to received new transactions or should I follow a different
>> procedure???
>>
>> Thank you
>>
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>
> _______________________________________________
> 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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