[SR-Users] Asterisk cluster behind kamailio natted to pubic IP, presenting internal ip addresses in From tag
miconda at gmail.com
Thu Sep 4 09:27:12 CEST 2014
On 28/08/14 20:33, Tim Chubb wrote:
>> 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 at 172.16.52.80:5080> & Contact: <sip:50001 at 172.16.52.80:5080>
> 4) Kamailio sends the invite onto the registered client
> 5) The registered client displays "Test User 1 50001 at 172.16.52.80:5080"
> What I would like to be displayed when the registered client rings is something like "Test User 1 50001 at sip.domain.tld" or "Test User 1 50001 at 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 at 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?
As Fred said, there are some dialplan functions for asterisk that should
help setting caller id. IIRC, there are also some specific fields in
database (e.g., domain part of the user).
Anyhow, what I really wanted to point here is, if you don't get it fixed
in asterisk (where is better), look at uac_replace_from() from uac
module of kamailio.
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
More information about the sr-users