Recompiling chan_sip.so

Hi All,

So I have been trying to make a change to my chan_sip.so I can get my MWI working with all my Avaya phones. I have it working with two of them and the other two not so much. Anyways here is the rundown:

FreePBX 12.0.76.2
Asterisk 13.5.0

I need to perform the following as per this blog:

change this:

if (!sip_standard_port(p->socket.type, ourport)) {
if (p->socket.type == SIP_TRANSPORT_UDP) {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d\r\n", exten, domain, ourport);
} else {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d;transport=%s\r\n", exten, domain, ourport, sip_get_transport(p->socket.type));
}
} else {
if (p->socket.type == SIP_TRANSPORT_UDP) {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n", exten, domain);
} else {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d;transport=%s\r\n", exten, domain, ourport, sip_get_transport(p->socket.type));
}
}

To This:

if (!sip_standard_port(p->socket.type, ourport)) {
if (p->socket.type == SIP_TRANSPORT_UDP) {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d\r\n", exten, domain, ourport);
} else {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d\r\n", exten, domain, ourport);
}
} else {
if (p->socket.type == SIP_TRANSPORT_UDP) {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n", exten, domain);
} else {
ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n", exten, domain);
}
}

Now here is what I have tried. I followed the guide to manually installing freepbx for cent OS etc. I have downloaded the source files for my version of asterisk 13.0.5 and made the changes to the chan_sip file and was able to recompile the code.

The big problem is that when I look at the chan_sip.so compiled file sizes they are way off. So the existing one that comes with the distro I have is only 941Kb very small. The complied one that I end up with is 4.5Mb which I figured was weird. I SSH'd into freepbx copied the newly compiled file over and then performed an amportal restart.

Well that didn't work because although I could still get access to the admin webpage I could see that the phone link was down hence asterisk was not starting.

Can someone provide some help on how I can achieve this, I think it's the key to getting these Avaya phones (at least the newer firmwares) to be able to speak nicely with SIP notify tag.

Thanks very much!