One thing I could do with help on is how to make the sip_cid_type work for INBOUND calls? i.e. ignore what's in the P-Asserted header and just take the "from" address. I don't have a specific trunk set up for these calls, they just come in on IP-authorised acl. FWIW, this carrier is the only source of inbound traffic on the box, so setting a global default would be acceptable. Cheers!
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?
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