Are you executing the route block for NAT handling in request_route and
failure_route blocks? It should be done in branch route or only once in
request_route. Look for add_rr_param() and see how many times it is
executed for a branch (if executed in request_route, then it is for all
branches at least one time).
Cheers,
Daniel
On 6/4/13 10:57 PM, hiro wrote:
actually, I now see my last message is wrong.
I've compared the 200 OKs that gets sent from freeswitch to my phone
after the busy error and after direct voicemail routing from LOCATION
when user is offline. Both 200 OKs look the same with one exception:
The one that doesn't work had nat=yes two times here:
Record-Route: <sip:77.13.20.156;r2=on;lr=on;nat=yes;nat=yes>
Record-Route:
<sip:77.13.20.156;transport=tcp;r2=on;lr=on;nat=yes;nat=yes>
On 6/4/13, hiro <23hiro@gmail.com> wrote:
ok. right now from tcpdump I can see the session progress and OK
messages are sent to the correct ip:port of my phone, but either the
phone doesn't receive it or it doesn't process it.
I assume the problem to be the headers sent by freeswitch, and perhaps
not changed appropriately by kamailio, but I'm not sure:
Call id and Cseq are the same as in RINGING, but contact header has
freeswitch's IP (on the same server as kamailio)
Contact: <sip:3@127.0.0.1:5070;transport=udp>
Could that ever work this way?
On 6/4/13, Daniel Tryba <daniel@pocos.nl> wrote:
On Tuesday 04 June 2013 12:07:35 hiro wrote:
Sometimes it also seemed that kamailio was sending the INVITE to the
phone instead of to freeswitch, or when i played around between
changing $du or $ru the INVITE gets sends to freeswitch but with the
wrong URI pointing to the phone instead of 127.0.0.1:5070 which is
where freeswitch is listening.
I guess it would be easier to reproduce if that random factor wasn't
there, but at least it's failing most of the time, only in different
ways.
I had hoped I could get at least confirmation that it "works here" to
keep me going :P
I will test with xlog when I can test at home again which would at
least exclude NAT issues.
It works here :)
I know your pain. I spend days figuring out the magic trick was to set
$du
to
null (which I stumbled upon by accident). Without $du=$null traffic was
being
routed (seemingly random) to either the registered phone or the actual
voicemail server.
# route to voicemail server
route[TOVOICEMAIL] {
if(!is_method("INVITE"))
return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail routing enabled but IP not
defined\n");
return;
}
if($avp(dst_voicemail))
{
$du=$null;
$ru = "sip:tovm-" + $avp(dst_voicemail) + "@" +
$sel(cfg_get.voicemail.srv_ip) + ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
return;
}
failure_route[MANAGE_FAILURE] {
....
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
route(CALLREDIRECT);
route(TOVOICEMAIL);
exit;
}
....
}
--
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Telefoon: 040 293 8661 - Fax: 040 293 8658
http://www.pocos.nl/ - http://www.sipo.nl/
K.v.K. Eindhoven 17097024
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Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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