[Freeswitch-users] Altering From Header in SIP Invite

[Rob Moore]

Hi All,

I'm having a little trouble with 'presentation numbers' with a new provider I'm in IOT with this week.

I'm trying to recreate the following Invite as the calls pass through our dialplan to this provider but there are issues with trying to get a different CLI into the From and P-Asserted-Identity headers.

When presenting a Freephone number (for example) we need to still present the regular number that would be used by this extension in the P-Asserted-Identity whilst passing the number we wish to have presented in the From header.

Currently we are not using Gateways so we cannot resort to using  <param name="caller-id-in-from" value="true"/> (although I expect this won't do what we need in this case) so I've looked at altering the channel variables  sip_from_user,sip_full_from and sip_full_uri using set and export dial plan apps but none of these seem to have any effect so I guess these variables must be read only.

I'm sure this must be simple, but can't for the life of me work out what I need to do.

Below is an extract from an example header from the provider I am trying to recreate, I've also added a copy of the Dialplan extension I am using to test.
If someone can tell me what I'm getting wrong I would really appreciate it.

Thanks

Rob


INVITE sip:+445600005262 at primarysip.barfoo.com;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: +445600000262 <sip:+445600005262 at primarysip.barfoo.com;user=phone>
From: <sip:+448000655262 at uk.foo.bar.2.net;user=phone>;tag=12544
P-Asserted-Identity: <sip:+445600655262 at uk.sdin.bt.net>
Call-ID: 1347372978-13100 at mgc-uk-998.n2<mailto:1347372978-13100 at mgc-uk-998.n2>
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE


DIAL Plan: (attempting +445600655262 in P Asserted-Id and +448000655262 in from) have commented out some things that I have tried.)
If you are worried about [sip_h_X-Gateway=4003:4] this is added to help our SBC forward calls to our different providers.

   <extension name="TC 25">
     <condition field="destination_number" expression="^10125$">
                                <action application="set" data="effective_caller_id_name=_undef_"/>
                                <action application="set" data="effective_caller_id_number=+445600005262 "/>
                                <action application="set" data="dtmf_type=rfc2833"/>
                                <action application="set" data="sip_cid_type=pid"/>
                                <!--<action application="export" data=" sip_from_user =+448000655262"/> -->
                                <!-- <action application="export" data="sip_h_from=<sip:+448000655262 at primarysip.barfoo.com>;tag=${sip_from_tag} "/> -->
                                <!--  <action application="export" data="sip_h_P-Asserted-Identity= +445600005262 <sip:+445600655262 at 172.17.7.4> "/> -->
                                <action application="info"/>
                  <action application="bridge" data="[sip_h_X-Gateway=4003:4]sofia/internal/+442920001199 at pstn.tel" />
     </condition>
   </extension>


Rob Moore
Telephony Systems Infrastructure Manager
Aeriandi
Aeriandi Ltd, Prama House, Banbury Road, Oxford, OX27HT
E: rob.moore at aeriandi.com<mailto:rob.moore at aeriandi.com>
W: www.aeriandi.com<http://www.aeriandi.com/>
M: +44 (0)7766 838040
T: +44 (0) 845 108 0308

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