2010/8/29 Sujeev suppo...@meewadaya.com:
if ($avp(s:did)) {
if(is_present_hf(Remote-Party-ID))
{
remove_hf(Remote-Party-ID);
}
if(is_present_hf(Privacy))
{
remove_hf(Privacy);
}
uac_replace_from($avp(s:did),);
avp_delete($avp(s:did));
};
I hope you also add the P-Asserted-Identity
Hello Castillo,
I don't change caller ID to anonymous for inbound calls and calls to PSTN.
I've a problem with one of my international termination provider. They ask
me to set caller ID as Anonymous before I send them to. I don't know why's
that.(I also couldn't find any good reason)
Thank you!
Hi,
I have two Opensips servers, both with public IP's, I use them as an
outbound proxy.
both servers are running version 1.6.2, the difference is that they run
different OS and that one of them installed via SVN.
The problem is that when I register a client via one server the contact
field is
I saw a post today regarding setting callerid to Anonymous, I was
interested in testing the code; so I loaded the UAC module in and
restarted opensips;
Not much information why it crashed...
The only thing I added to my existing opensips.cfg is include uac.so;
Then restarted. (If I remove
Hi Bogdan,
Thanks for the answer. Basically, I know there is no magic and just matering
the routing techniques; sipout.com service are completely transparent to the
end users (sip UA) the way that an SIP UA can use it and connect to any VOIP
provider if the ISP is blockind SIP UDP ports. My
2010/8/30 S. Millard mga...@yahoo.fr:
My question is (and for sure if
I am not too much asking!) do you have any configuration files (XML) for
opensisps or freeswitch that I can use to achieve this function the same way
sipout.com are doing ?
The configuration file of OpenSIPS is not a XML
