Yes, I am. The only header line that seems to resemble is the following:
P-Called-Party-ID:<sip:ABCDEFGHIJ@192.168.11.250>
where ABCDEFGHIJ is my DID number. Oddly, I do not see the same line in the INVITES from the same gateway to Express Talk (software SIP phone) and AseteriskWin32.

Does this help at all?

Yasuro

P.S. Here's an INVITE in its entirety:
 INVITE sip:gw+192.168.11.250@192.168.11.100:5080 SIP/2.0
 f:<sip:anonymous@anonymous.invalid>;tag=2D690025DC15F5C9001DAD62
 t:<tel:0787920215;phone-context=192.168.11.250>
 m:<sip:09909290030970274201@192.168.11.250>
 Allow:INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,REFER,NOTIFY
 k:replaces,100rel,timer
 x:300
 Max-Forwards:68
 P-Called-Party-ID:<sip:ABCDEFGHIJ@192.168.11.250>
 Record-Route:<sip:192.168.11.250;lr>
 c:application/sdp
 Call-ID:78E99320A3B0D174D7C754C031078615@192.168.11.250
 CSeq:1 INVITE
 Via:SIP/2.0/UDP 192.168.11.250:5060;branch=z9hG4bK8C863FC5D5
 Content-Length:141

 v=0
 o=- 395453 395453 IN IP4 192.168.11.250
 s=-
 c=IN IP4 192.168.11.250
 t=0 0
 m=audio 5004 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 a=ptime:20

David Ponzone wrote (11/17/2010 7:39 PM):
Hmm, are you sure your ITSP is not sending a RPID in the INVITE ?

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.




Le 17/11/2010 à 11:25, Yasuro a écrit :

Hello!

Would you tell me how you can stop FreeSWITCH from including the Remote-Party-ID header in its SIP messages? I am suspecting my SIP service provider does not like to see it in the OK messages FS sends in response to their INVITEs.

I know there is sip_cid_variable channel variable, but setting it to none by
 <action application="set" data="sip_cid_type=none" />
in the 00_inbound_did.xml dialplan before transferring the incoming call to an extension does not seem to have any effect.

Please advise. Thanks!

Yasuro


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