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

[Skyler]

Hi,

 My mistake. I must have had 2.2.12 proxy.cfg loaded (might be a good
idea to add #NGCP-Proxy-v2.2.xx in the file(s) for reference).

 I have manually ensured that 2.2.13 is loaded, retested the reminder
and it is in fact working perfectly without any need for callerID-name
patching. 

 Please find attached the updated ngcp-proxy-2.2.13_callerID-name.patch
diffed from 2.2.13

 With regards to reasoning the PAI removal; here is the call flow:

 PAP2T --> SPCE --> Aastra-6731

 I suspect that the Aastra device recognizes this PAI and replaces the
FROM at the device level. I may be completely wrong but when this
specific PAI is removed in the script, my troubles ceased.

--Skyler

On Fri, 2011-09-09 at 11:40 +0200, Andrew Pogrebennyk wrote:
> 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.
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