Greetings ! I am able to call from extention 1001 to 1002 and vise versa but its hanging its self within 6 seconds and there is no voice transmissiond

Please provide logs as plain text and with sip set debug on enabled. Please include the whole call.

From this and your previous posting, you seem to be using FreePBX. FreePBX is supported at htttps://community.freepbx.org/ Do not expect answers in terms of the FreePBX GUI here.

Because of the huge amount of dialplan involved in a FreePBX call, you will need to use a pastebin type service for the logs. FreePBX provide their own one for use with their forum. Your log starts after the decision to end the call has been made so tells us nothing except that the call is ending.

Note that chan_sip is deprecated and has been scheduled for removal. You should use chan_pjsip unless there is a specific reason to do otherwise. ITSPs saying they do not support it is generally not a good reason.

This is the error I am getting within those 6 seconds of the call

86351[2021-12-28 12:09:33] WARNING[13517] chan_sip.c: Retransmission timeout reached on transmission Rj-4zOhcgvO848Sc-hu3Xg… for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

[2021-12-28 12:09:33] WARNING[13517] chan_sip.c: Hanging up call Rj-4zOhcgvO848Sc-hu3Xg… - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

That’s rather fast, but it suggests that Contact header is wrong in the response, which, in turn, would be most likely because you hadn’t set your public address correctly, or less likely, you had set your local network to include the ITSP’s public address.