[Spce-user] NGCP-2.2.13_Proxy.cfg.tt2_Caller-ID_Name_Patch

[Andrew Pogrebennyk]

I am keeping mailing list in cc so that the others can correct me if I'm 
wrong.

On 09/08/2011 03:03 PM, Skyler wrote:
>   Sure, from 2.2.13 proxy.cfg you can see that there is no subscriber
> name display when set on voip devices or soft phones. The name is
> replaced by script and verified by ngrep traces.
>
>   what I found is by removing the 2 uac_replace_from("$var(caller_cli)",
> "$var(caller_cli_uri)"); that now the display name is working.
> [...]

I would to see the trace and the log of the reminder call made with 
unmodified config. From my POV everything is correct, I've just retested
with default display-name "reminder":
> INVITE sip:123100 at 10.15.20.67 SIP/2.0
> Max-Forwards: 10
> Record-Route: <sip:192.168.31.108;r2=on;lr=on;ftag=27E8B7F2-4E69DA05000BD40F-D3B57700>
> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=27E8B7F2-4E69DA05000BD40F-D3B57700>
> Via: SIP/2.0/UDP 192.168.31.108;branch=z9hG4bK3e54.0599dfa2.0
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKebSerahK;rport=5080
> From: "reminder" <sip:reminder at voicebox.sipwise.local>;tag=27E8B7F2-4E69DA05000BD40F-D3B57700
> To: <sip:123100__AT__192.168.31.108 at 127.0.0.1:5062>
> CSeq: 10 INVITE
> Call-ID: 116c1d8e1461c0976aedaa37109c9b87 at voicebox.sipwise.local_b2b-1
> P-Asserted-Identity: <sip:reminder at voicebox.sipwise.local>
> Supported: timer
> Session-Expires: 300
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 286
> Contact: <sip:ngcp-lb at 192.168.31.108:5060>

with e164 number:
> INVITE sip:123100 at 10.15.20.67 SIP/2.0
> Max-Forwards: 10
> Record-Route: <sip:192.168.31.108;r2=on;lr=on;ftag=0E87D1E9-4E69DC5D0008BCE0-D3B57700>
> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=0E87D1E9-4E69DC5D0008BCE0-D3B57700>
> Via: SIP/2.0/UDP 192.168.31.108;branch=z9hG4bK0ac4.eb5116c3.0
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKHVFhvaI.;rport=5080
> From: "1234567890" <sip:1234567890 at voicebox.sipwise.local>;tag=0E87D1E9-4E69DC5D0008BCE0-D3B57700
> To: <sip:123100__AT__192.168.31.108 at 127.0.0.1:5062>
> CSeq: 10 INVITE
> Call-ID: 49d99e1f31bcb7873e7b2d54719fa865 at voicebox.sipwise.local_b2b-1
> P-Asserted-Identity: <sip:1234567890 at voicebox.sipwise.local>
> Supported: timer
> Session-Expires: 300
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 286
> Contact: <sip:ngcp-lb at 192.168.31.108:5060>

so the From is correct, isn't it?

>   Further, I found with certain voip phones (Aastra/Cisco) the incoming
> caller name display is not working from ATA's (Linksys/grandstream) on
> inbound calls. Once append_hf("P-Asserted-Identity:<
> $var(caller_cli_uri)>\r\n"); was removed, this became normal. So the PAI
> removal was to fix this. Now, all 5 devices here have no issue with
> caller name display.
>

My concern with PAI removal in such way is that it will affect all calls 
so it is not something we would like to do taking into account that PAI 
is a common source of cli/username. If it is needed for some 
Aastra/Cisco/ATAs - it should be made a configurable option turned off 
by default, but that's a different topic.