PJSUA Invites to Arbitrary URI/Address Pairs

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Hi Curt,

Yes, you'll need 30 accs for 30 calls, it is the simpler way. Another
way should be manually inserting route header into INVITE message in
tx_data (which you mentioned to be not working, well, it should), e.g:
--
pjsua_msg_data msg_data;
pjsip_route_hdr *h;
pjsip_sip_uri *url;

url = pjsip_sip_uri_create(pool, 1);
pj_strdup2(pool, &url->host, "1.2.3.4");
url->lr_param = 0;

h = pjsip_route_hdr_create(pool);
h->name_addr.uri = (pjsip_uri*)url;

pjsua_msg_data_init(&msg_data);
pj_list_push_front(&msg_data->hdr_list, h);

pjsua_call_make_call(..., &msg_data, ...)
---

Note that the route/proxy setting alone doesn't seem to solve the
certificate verification problem (see my first reply). You can trick
the TLS transport to verify subjectaltname in remote cert against
custom defined target name (as it is resolved using non-DNS mechanism)
by modifying destination name info in tx_data, e.g:
--
pj_strset2(&tx_data->dest_info.name, "p.com");
--
unfortunately, you'll need to modify pjsua_call_make_call() for this!

BR,
nanang


On Thu, Mar 1, 2012 at 10:35 AM, Curt Sampson <cjs at cynic.net> wrote:
> On 2012-02-29 17:09 +0700 (Wed), Nanang Izzuddin wrote:
>
>> No, single account can be used for multiple simultaneous calls.
>
> So how do I specify, when using a single account, that call A should go
> to "proxy" A', call B to proxy B', and so on?
>
> Remember, the SIP address to which I'm directing a call has no IP
> address or port information associated with it; I know the IP address
> and port currently being used by that SIP address through out-of-band
> means.
>
> cjs
> --
> Curt Sampson ? ? ? ? <cjs at cynic.net> ? ? ? ? +81 90 7737 2974
> ? ? ? ? ? ? http://www.starling-software.com/
> I have always wished for my computer to be as easy to use as my telephone;
> my wish has come true because I can no longer figure out how to use my
> telephone. ?--Bjarne Stroustrup
>
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