I tried it(./res/res_pjsip.c :
if (!ast_strlen_zero(endpoint->contact_user)) {
pjsip_sip_uri *sip_uri;
sip_uri = pjsip_uri_get_uri(dlg->local.contact->uri);
file004r = fopen("/tmp/0/res_pjsip.log","ab");
fwrite("\n456=456\n",strlen("\n456=456\n"),1,file004r);
fwrite("\n===SIP_URI_USER_BEFORE=\n",strlen("\n===SIP_URI_USER_BEFORE=\n"),1,file004r);
fwrite(sip_uri->user.ptr,sip_uri->user.slen,1,file004r);
fwrite("\n===SIP_URI_HOST_BEFORE=\n",strlen("\n===SIP_URI_HOST_BEFORE=\n"),1,file004r);
fwrite(sip_uri->host.ptr,sip_uri->host.slen,1,file004r);
pj_strdup2(dlg->pool, &sip_uri->user, endpoint->contact_user);
fwrite("\n===SIP_URI_USER_AFTER=\n",strlen("\n===SIP_URI_USER_AFTER=\n"),1,file004r);
fwrite(sip_uri->user.ptr,sip_uri->user.slen,1,file004r);
fwrite("\n===SIP_URI_HOST_AFTER=\n",strlen("\n===SIP_URI_HOST_AFTER=\n"),1,file004r);
fwrite(sip_uri->host.ptr,sip_uri->host.slen,1,file004r);
fclose(file004r);
}
added at /etc/asterisk/pjsip.conf
contact_user = asterisk654
make && make install
systemctl restart asterisk
made a test call
and results:
cat /tmp/0/res_pjsip.log
456=456
===SIP_URI_USER_BEFORE=
asterisk
===SIP_URI_HOST_BEFORE=
ast.firma.org
===SIP_URI_USER_AFTER=
asterisk654
===SIP_URI_HOST_AFTER=
ast.firma.org
Apparently sip_uri->host has already had the value "ast.firma.org",
but in the INVITE in the "Contact" field - ip address = 11.22.33.44:
Contact: <sip:asterisk654@11.22.33.44:5061;transport=TLS>
Such INVITE sent asterisk:
<--- Transmitting SIP request (813 bytes) to TLS:52.114.76.76:5061 --->
INVITE sip:+380770...@sip.pstnhub.microsoft.com:5061 SIP/2.0
Via: SIP/2.0/TLS
11.22.33.44:5061;rport;branch=z9hG4bKPj70d53a0c-a0f3-479a-8917-00082632d069;alias
From: <sip:6...@ast.firma.org>;tag=afd3b600-886a-45fb-a120-5faac8220cec
To: <sip:+380770...@sip.pstnhub.microsoft.com>
Contact: <sip:asterisk654@11.22.33.44:5061;transport=TLS>
Call-ID: b0323db6-6e7f-4dc4-bfa9-309683031962
CSeq: 20795 INVITE
Serg
________________________________
From: Joshua C. Colp <jc...@digium.com>
Sent: Tuesday, October 30, 2018 15:19
To: Колтогян Сергей Рубенович U2; Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Asterisk 16. PJSIP. INVITE. "Contact" field and FQDN
On Tue, Oct 30, 2018, at 10:16 AM, KoltogyanU2 SergeyU2 wrote:
> I found a place in res_pjsip.c, where a new "contact_user" is added.
> This is file ./res/res_pjsip.c string 3514
> pj_strdup2(dlg->pool, &sip_uri->user, endpoint->contact_user);
>
> I Tried this:
> pj_strdup2(dlg->pool, &sip_uri->user, "XXYYZZ");
> So it works - it changes the ContactUserName ( on the left of @ ).
>
> But I do not find a place where I can enter FQDN, instead of the IP
> address.
> Someone can tell - in which place of the source text to make a change,
> what was right of @ was not an IP Address, but a FQDN?
The structure of a PJSIP SIP URI is documented in their doxygen[1], this
includes the part you are looking for (the hostname portion).
[1]
https://emea01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.pjsip.org%2Fpjsip%2Fdocs%2Fhtml%2Fstructpjsip__sip__uri.htm&data=02%7C01%7Cu2%40amintegrator.com%7C0016dac0a75c4036998908d63e6a4b76%7C806bcdf8cdfb4595995e80fbcae20097%7C0%7C0%7C636765023596895828&sdata=oMRG%2FzRlBEeTvbQudg2l7Y3UjNfp8iBgRls1OAxPSzo%3D&reserved=0
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Joshua C. Colp
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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&
https://emea01.safelinks.protection.outlook.com/?url=www.asterisk.org&data=02%7C01%7Cu2%40amintegrator.com%7C0016dac0a75c4036998908d63e6a4b76%7C806bcdf8cdfb4595995e80fbcae20097%7C0%7C0%7C636765023596895828&sdata=wRhVtosDsSsfYHrGe8YCKTRMia7EUQxTtRxBTaJpU%2Fg%3D&reserved=0
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