Goip + asterisk

Hi everyone.
Tot much people who work with Goip know that goip is work not corect. From manual:
SIP 183

The device sends back a SIP 183 Session In Progress message to the calling SIP device. The calling SIP device then goes into early media mode to receive audio packets. Since it may take a few to over 10 seconds for a ringback tone is returned from the GSM network, the caller may hear a long silent period.

It is mean that goip send two progress. First after invite and second when progress come from gsm operator. It asterisk it look like:
18:04:29] – Called SIP/t101/474305+375297878483
18:04:29] – SIP/t101-00001771 is making progress passing it to IAX2/2goip-2400
18:04:29] – Probation passed - setting RTP source address to
18:04:40] – SIP/t101-00001771 is making progress passing it to IAX2/2goip-2400
18:04:51] – SIP/t101-00001771 answered IAX2/2goip-2400

In dump before asterisk it look the same (it is another call, so time is different):
|84.132637| 100 Trying| |SIP Status
| |(5060) <------------------ (5087) |
|84.133042| 183 Ringing SDP (g72 |SIP Status
| |(5060) <------------------ (5087) |
|99.061594| 183 Ringing SDP (g72 |SIP Status

Who now how we can ignore quick firs 183. I have only one idea - male pathc by compiling chan_sip.so with such case “if 183 come during 1 second after invite - drop packet”. But i am 0 in C. So, who know does it possible to make in dialplan or sip.conf something to ingore this first 183 or to whom i have to ask about make pathc for asterisk :smile: