H323 conversion to SIP

We have a H323 phone that calls into Asterisk turns into sip. Then goes to our call manager, but gets denied. Mainly due to the header looks like x.x.x.x@x.x.x.x but we need to have it come in as 3000@x.x.x.x with the number in the header. We only need the number the ip is throwing off the sip server. Can we change the header into the file. Even on the sip phone it reaches and says from. x.x.x.x really can you change the header to have the H323 stream come in with a phone number and not ip or at least when it converts to sip to then carry the phone number and not the ip in the header. Thanks

Make and model of call manager?

What header looks like x.x.x.x@x.x.x.x (I’m not aware that any SIP header generated by Asterisk will look like this, even when you strip the header name)?

Please provide annotated logs, as it is not at all clear what you are trying to say.

It’s not a call manager but Wave a media server. What logs would be helpful. I have some tcpdumps.

Just to clarify my Cisco 7965 makes a call SIP call to another SIP phone. It has a header similar to From: 1000@ the extension and the ip of the phone.

My H323 phone make a call to a SIP phone. It is H323 format to asterisk, then asterisk will convert that call to sip. That header looks like with that latter being the up of asterisk. It still rings the SIP phone but I can see on the Cisco display there is text on bottom of screen that says “from a small the call is active. My goal would be to have that header not carry the H323 phone ip and just use the extension like the original sip call. It’s being rejected by the Wave system. I’m using the ooh323 file without gatekeepers it asterisk. I understand there is a H323 file which I’m curious about. Asterisk version is 13.19.2.
Thank you

Asterisk VERBOSE, and possibly DEBUG logs.