[Spce-user] Outgoing callerID on SIP Subscriber Diversion

[Andrew Pogrebennyk]

Hi Kalen,

On 11/13/2014 05:19 PM, Kalen Krueger wrote:
> 
> Here is my observations from the Asterisk Systems perspective
> (subscriber pbx)
> 
> CallerID of original incoming call to subscriber: 1112223333
> 
> Subscriber alias dialed: 4445556666
> 
> Cell Phone Number the call is forwarded to be the asterisk system 7778889999
> 
>  
> 
> INVITE sip:7778889999 at myspcehost.com SIP/2.0
> 
> From: <sip:1112223333 at myspcehost.com>;tag=as46614923
> 
> To: sip:7778889999 at myspcehost.com
> 
> Contact sip:1112223333 at subscriber_pbx.com:5060
> 
> Diversion: <tel:4445556666>;reason=no-answer;screen=no;privacy=off
> 
>  
> 
> In SPCE here are my settings (that clearly appear relevant)
>  [...]
> 
> Actual Results are that the call forward works, however the subscriber
> number is shown for CallerID rather than the original callerid as desired.

The INVITE snippet above shows that the Diversion header is delivered
with URI "tel:4445556666", however this is not SPCE that's adding this
Diversion header. So I believe this is coming from the PSTN:
<PSTN> -- <SPCE> -- <Asterisk SIP>

I'm not sure why the PSTN gw puts the called number 4445556666 in the
Diversion header.. it shouldn't do that. And it seems asterisk is
showing 4445556666 as CallerID, isn't it? So probably the problem there
is that the SPCE is not removing the incoming Diversion header. We have
fixed this issue a few weeks ago. Could you please specify what version
of templates you are running (dpkg -l ngcp-templates-ce-kamailio) if you
have a customtt. Or if I misunderstood the problem, please also provide
the kamailio-proxy.log and the packet dump that belong together.

Regards,
Andrew