[Freeswitch-users] CallerID withheld problem

[Alex Lake]

Hello there....

One of our carriers has changed their signalling today and its resulted 
in some strange behaviour through our Freeswitch box.

The change was that the P-Asserted ID has changed format to have a "+44" 
at the beginning - it used to be in the national format (eg. "02070601234").

The problem manifests itself further down the chain (the FS box forwards 
to an Asterisk box which forwards to a legacy system using E1's and 
Aculab cards). For some reason the E1 interprets the signalling as 
withheld caller id.

Message Header is:

Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:448700802332 at 213.239.197.41>
From: <sip:02070604380 at 217.145.67.2>;tag=3609589914-358461
P-Asserted-Identity: <sip:+442070604380 at 193.113.183.202>
Call-ID: 3560925-3609589914-358457 at MSX1.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, 
SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 
217.145.67.2:5060;branch=z9hG4bK630bccab557da527f8875b902c88fb6b
Contact: <sip:02070604380 at 217.145.67.2:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 373

I forward this on as:

Via: SIP/2.0/UDP 213.239.197.41;rport;branch=z9hG4bKm3jHajt7g1gaa
Max-Forwards: 68
From: "02070604380" <sip:02070604380 at 213.239.197.41>;tag=57cmagKg0yctN
To: <sip:448700802332 at 89.202.128.200>
Call-ID: 81dbf634-5ad9-1232-1f90-d43d7ee2fdec
CSeq: 59964493 INVITE
Contact: <sip:mod_sofia at 213.239.197.41:5060>
User-Agent: 
FreeSWITCH-mod_sofia/1.2.23+git~20140319T002132Z~b96946822d~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, 
call-info, sla, include-session-description, presence.winfo, 
message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 207
X-FS-Support: update_display,send_info
P-Asserted-Identity: "02070604380" <sip:02070604380 at 213.239.197.41>

I don't know if the SIPerati can discern anything from this. I could 
pastebin the pcap file if that would be useful.

But the angle might be for me to tell FS to give calls a sip_cid_type of 
"none". But I don't quite undersrand from the wiki 
(https://wiki.freeswitch.org/wiki/Variable_sip_cid_type) where I should 
put this.

The current dialplan is:

<extension name="ajl-test2">
   <condition field="destination_number" expression="^44">
     <action application="privacy" data="no"/>
     <action application="log" data="INFO caller_id_number 
0${caller_id_number:3}"/>
     <action application="export" 
data="origination_caller_id_number=0${caller_id_number:3}"/>
     <action application="bridge" 
data="sofia/internal/${destination_number}@89.202.128.200"/>
   </condition>
</extension>

Note the hack to change callerid from +44xxx... format to 0xxx... has 
been replaced by a Lua script that handles it a bit more cleanly.

But that "privacy" setting doesn't seem to be enough.

Any ideas or suggestions for more diagnostics?