28 Aug
2014
28 Aug
'14
2:33 p.m.
I'm assuming with the 5080 that this call goes
through the Asterisk box before hitting the registered user on Kamailio... if that's
correct, have you also forced a CALLERID(name) on the call?
A grep of the sip traffic would show if you have
something perhaps removing this information before sending to the client.
Im using realtime and setting the caller id in the db i.e. I have an entry like "Test
User 6" <50006> in the caller ID column.
This explanation may clarify a bit more:
1) Test User 1 (50001) dials 50006
2) Asterisk (Server resides in the voice DMZ with an IP of 172.16.52.80 listening on port
5080) sends and invite to the kamailio box's voice DMZ ip (172.16.52.70:5060)
3) This invite contains a From and Contact header looking like: From: "Test User
1" <50001@172.16.52.80:5080> & Contact:
<sip:50001@172.16.52.80:5080>
4) Kamailio sends the invite onto the registered client
5) The registered client displays "Test User 1 50001@172.16.52.80:5080"
What I would like to be displayed when the registered client rings is something like
"Test User 1 50001(a)sip.domain.tld" or "Test User 1 50001@publicip" or
even just "Test User 1 50001"
If I enable SIP inspection on the ASA that sits infront of the kamailio box, I will get a
public IP of the gateway displayed but not the one that points to the DMZ interface of the
kamailio box, and it still displays the internal port that the asterisk box is on, i.e.
"Test User 1 50001@123.321.123.321:5080"
Im getting the feeling that I am not grasping something really basic, or that I have
misconfigured asterisk somewhere along the line, as far as I can tell kamailio is working
exactly as advertised, and the problem is originating upstream, or should I use kamailio
to normalise the traffic emitting from it?