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@213.239.197.41> From: <sip:02070604380@217.145.67.2>;tag=3609589914-358461 P-Asserted-Identity: <sip:+442070604380@193.113.183.202> Call-ID: 3560925-3609589914-358457@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@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@213.239.197.41>;tag=57cmagKg0yctN To: <sip:448700802332@89.202.128.200> Call-ID: 81dbf634-5ad9-1232-1f90-d43d7ee2fdec CSeq: 59964493 INVITE Contact: <sip:mod_sofia@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@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?