[Kamailio-Users] 30x Redirect / Forward, even questions
Daniel-Constantin Mierla
miconda at gmail.com
Thu Jan 14 11:31:01 CET 2010
Hello,
On 1/13/10 10:59 PM, Samuel Muller wrote:
> Hello Daniel,
>
> thanks for your answer.
>
> I'm using route(3) to find the new callee, using a radius request (no
> DB for this kind of request).
> Using t_relay(), I just forward the request without try to know if the
> new callee is one of my own user, isn't it ?
> In an operator way, I've to control the "who's who" ...
>
> I took this part of config in the sipwise tool :
>
> after the get_redirect in the failure_route() :
>
> $avp(s:fwdtmpds) = $ds;
> avp_subst("$avp(s:fwdtmpds)", "/.*(sip:.+@[^:;>]+).*$/\1/");
> avp_pushto("$ru", "$avp(s:fwdtmpds)");
> # flag call fwd (for acc, permissions, rtp proxying, ...)
> setflag(17);
> append_branch();
> t_on_branch("1");
> # route to find the new callee
> route(3);
> exit;
> }
>
> It works, but it's not a clean way, and really difficult to replace
> good avps, determinates rtp proxying (one way audio only when the UAC
> who forwarded the call is behind nat), delete few double insertion of
> P-Hints. Accouting is really hard to clean too (who pays the first
> call pstn -> phone, and who's paying call to redirected number, then
> consider all of them as only one call for billing system).
Can you grab a sip trace and send it of an entire scenario (from initial
invite). Would help to understand what is happening and what is not a
clean way.
Cheers,
Daniel
>
> With SIP over TCP, it doesn't work (it creates two pinholes, and I've
> errors on the debug logs). I'm afraid to test all of this stuff with
> TLS ...
>
> Do you think that a dedicated redirect server is a better way to take
> in charge call forwards ? so I could use the t_relay() function.
> Sure the best practice is to use SBC, IPBX, media servers and/or real
> class5 softswitch to do all the advanced voice services, but in some
> case ...
>
> Does someone developped a sip-i module eventually ?
> it could be a nice feature for operators who are blocked by the call
> forwards systems, few parts of ISUP must be taken in charge sometimes,
> like when you have to forward a call to a PSTN gateway interconnected
> in SS7 with a tier1-operator.
>
> Cheers,
>
> .Sam.
>
>
> On Wed, Jan 13, 2010 at 12:44 PM, Daniel-Constantin Mierla
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
> Hello,
>
>
> On 1/11/10 1:56 PM, Samuel Muller wrote:
>
> Hello all,
>
> I've questions 'bout the redirect/forward with 30x requests.
>
> Actually, it does not work, it seems the uac redirect module
> doesn' t use the new contact for the new invite.
>
> simple schema : pstn gw -> kamailio -> linksys -> call forward
> unconditionnal to another pstn number -> kamailio -> gw pstn
>
> the 302 as seen with ngrep tool :
>
> #SIP/2.0 302 Moved Temporarily.
> #To: <sip:B at sip_proxy;user=phone>;tag=91515579fff6b8d1i0.
> #From:
> <sip:A at pstn_gw1;user=phone>;tag=16627-UI-00020cf3-54eea92c4.
> #Call-ID: 16627-RO-00020cf2-4186e9762 at pstn_gw1.
> #CSeq: 81990 INVITE.
> #Via: SIP/2.0/UDP sip_proxy;branch=z9hG4bK0f0c.22989b87.1.
> #Via: SIP/2.0/UDP
> pstn_gw1:5060;rport=5060;received=ip_pstn_gw1;branch=z9hG4bK-25EA-243F3.
> #Record-Route:
> <sip:sip_proxy;lr=on;ftag=16627-UI-00020cf3-54eea92c4;did=f65.bcc44f06>.
> #Contact: <sip:C at sip_proxy>.
> #Diversion: "B" <sip:B at sip_proxy>;reason=unconditional.
> #Server: Linksys/SPA942-6.1.3(a).
>
> DBG:uac_redirect:shmcontact2dset: adding contact <sip:C at sip_proxy>
> dest set : Contact: sip:B at ip_B, <sip:C at sip_proxy>;q=0.01
> contact $ct: <sip:pstn_gw1:5060>
>
> the second pstn gateway received a request to B, and not C (so
> it creates a loop through pstn ...).
>
> in my config :
>
> loadmodule "uac.so"
> modparam("uac", "rr_store_param", "vsf")
> modparam("uac", "from_restore_mode", "auto")
>
> loadmodule "uac_redirect.so"
> modparam("uac_redirect", "default_filter", "accept")
> modparam("uac_redirect","bflags", 1)
>
> loadmodule "diversion.so"
>
> failure_route[2]
> {
> setdebug(4);
>
> if (t_check_status("301|302"))
> {
> route(26) # acc fwd
> $avp(s:acc_status) = "cfu";
>
> if (!get_redirects("2:1"))
> {
> xlog("L_NOTICE", "failed to extract
> contact info from 30x header");
> route(10); # stop rtp
> t_reply("480", "Temporarily Unavailable");
> exit;
> }
>
> # flag 3xx status, open a new branch and back
> to find the new callee
> setflag(28);
> append_branch();
> t_on_branch("1");
> route(3); # search the new callee
> setdebug();
> exit;
> }
>
> t_on_branch("1");
> setdebug();
> route(3); # find callee
> route(12); # local outbound
> route(10); # stop rtp
> }
>
> I'm using lcr module to route calls, kamailio v. 1.5.3.
>
> I've another question, is the 30x always considered as a
> failure_route ?
>
> Any help and/or explanations about the tricks would be really
> appreciated ... thanks,
>
> as I can see from you comments, in failure_route you get the
> contacts from 30x and append branch then call route(3) which is
> find callee -- what is that, is it appending another branch?
> Practically after append branch you should just do relay.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> * http://www.asipto.com/
>
>
--
Daniel-Constantin Mierla
* http://www.asipto.com/
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