[OpenSIPS-Users] ACK and NAT problem
Nawfel Oujdi
noujdi at egtelecom.es
Fri Nov 26 10:14:02 CET 2010
Hello Bogdan,
Thanks for reply my email and just say i fixed the problem.It was exactly
like you explain in your email so i did 2 changes in my opensips.cfg: first
i preset the record route with my public ip ""record_route_preset("Public
IP")" and second i put aliases= "Public IP" and then everything seems
work well.
Best Regards and great job!!
Nawfel
2010/11/25 Bogdan-Andrei Iancu <bogdan at voice-system.ro>
> Hi Nawfel,
>
> The problem is that opensips, as running on a private address, will
> advertise in Route hdr that private IP (that will be useless when GW tries
> to route the ACK).
>
> So, currently, what is the way your opensips is reachable from outside ?
>
> Best regards,
> Bogdan
>
> Nawfel Oujdi wrote:
>
>> Hi!!
>>
>> I load balancing between 2 asterisk,the load_balancer it s done
>> correctly but the calls hangs up after 10 second approximately. My setup is
>> (just with 1 asterisk):
>> Cisco GW ----(NAT)----> Opensips --------> Asterisk
>>
>> My problem is that opensips never receive the ACK from Cisco GW even if
>> this last one sent it.
>>
>> CISCO logs:
>> ACK sip:911126667@*public ip of asterisk* :5060 SIP/2.0 Route:
>> <sip:*internal ip of opensips*;lr=on;ftag=F40B44FC-18A6;did=1cd.369d0f26>
>> I think cisco reply ACK to Asterisk through the internal ip of opensips
>> cause opensips did not provide Cisco with his public IP (NAT).
>>
>> Can somebody help me with this??
>>
>> I heard about fix_nated_contact() but i don't know how and where in my
>> script¿Somebody send me examples ?
>>
>>
>> Thank you very much!!
>>
>> my opensips.cfg:
>> route{
>>
>> if (!mf_process_maxfwd_header("10")) {
>> sl_send_reply("483","looping");
>> exit;
>> }
>> if ($rU==NULL) {
>> # request with no Username in RURI
>> sl_send_reply("484","Address Incomplete");
>> exit;
>> }
>> if (!has_totag()) {
>> # initial request
>> record_route();
>> } else {
>> # sequential request -> obey Route indication
>> loose_route();
>> t_relay();
>> exit;
>> }
>> if ( is_method("CANCEL") ) {
>> if ( t_check_trans() )
>> t_relay();
>> exit;
>> }
>>
>>
>> # from now on we have only the initial requests
>> if (!is_method("INVITE")) {
>> send_reply("405","Method Not Allowed");
>> exit;
>> }
>>
>> # detect resources and do balancing
>> if (method=="INVITE") {
>> load_balance("1","calls"); }
>>
>> # LB function returns negative if no suitable destination (for
>> requested resources) is found,
>> # or if all destinations are full
>> if ($retcode<0) {
>> sl_send_reply("500","Service full");
>> exit;
>> }
>>
>> xlog("Selected destination is: $du\n");
>>
>> # send it out
>> if (!t_relay()) {
>> sl_reply_error();
>> }
>> }
>>
>>
>>
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
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>>
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>>
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>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Nawfel Oujdi
*Ingeniero VoIP*
noujdi at egtelecom.es
EG telecom S.A | www.egtelecom.es
Oficina: *902 050 080*
Agustín de Foxá, 25 - 9B | 28036 Madrid
------------------------------
Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s)
dirección(es) indicadas anteriormente; el carácter confidencial, personal e
intransferible del mismo está protegido legalmente. Cualquier revelación,
uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha
recibido este mensaje por equivocación, notifíquelo inmediatamente a la
persona que lo ha enviado y borre el mensaje original junto con sus ficheros
anexos sin leerlo ni grabarlo, total o parcialmente.
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