[Freeswitch-users] CallerID withheld problem

[Avi Marcus]

At a quick glance... sounds like you want to normalize everything to the
0XX format.
If you're getting +44 which should just be 0, then match ^\+44(\d*)$ and
set effective_caller_id_number to be 0$1.

-Avi


On Tue, May 20, 2014 at 7:26 PM, Alex Lake <alex at digitalmail.com> wrote:

> 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?
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
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