ATA 186 and asterisk problem

Hi guys!

I have strange problem.

Cisco ATA 186 - 2 channel (extentions 6003, 6004) connected to Asterisk.

Goip4 gateway (goip) connected to Asterisk in trunk gateway mode (no auth, just by IP).

When make 2 calls at one time
call1:from 6003 to goip/number1
call2:from 6004 to goip/number2

6004 will connects to call1 and hear audio from this call.

Asterisk reports:
[2013-10-31 10:45:54] NOTICE[16238] chan_sip.c: Disconnecting call ‘SIP/6004-00000073’ for lack of RTP activity in 31 seconds

all delices are local 192.X.X.X

What is th eproblem?

Have read in Google to try
"canreinvite = no" for all trunks.
But im not cure it will help. Becuse i use “canreinvite = no” in whole SIP configuration. I think it applicable to all extentions and trunks?

also removed “_” in trunk name and set fixed outbount CID for call this trunk

HAHAHA it was wire problem! short curcuit!!! :slight_smile: