HI list, I have succeeded to establish calls using OpenBTS/USRP1/Asterisk. but the problem is that my cell phone rings, I get 2 way audio but after a few seconds the call is dropped. In my asterisk log I see this: [Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 1348333597 at 127.0.0.1 for seqno 94 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3651 retrans_pkt: Hanging up call 1348333597 at 127.0.0.1 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). In the SIP Debug, I see always 10 Retransmissions of the same "SIP/2.0 200 Ok" message!!! after that the above "Retransmission timeout" message is viewed!!! Retransmitting #10 (no NAT) to 127.0.0.1:5062: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK53934;received=127.0.0. 1 From: IMSI208012601160193 <sip:IMSIxxxxxx at 127.0.0.1<sip%3AIMSI208012601160193 at 127.0.0.1>>;tag=lkbdgTo: <sip:2121211 at 127.0.0.1>;tag=as7c57c466 Call-ID: 1348333597 at 127.0.0.1 CSeq: 94 INVITE Server: Asterisk PBX 1.8.5.0-1digium1~lucid Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:2121211 at 127.0.0.1:5060> Content-Type: application/sdp Content-Length: 213 I have established only a one call from my hardphone (connected to OpenBTS) to my twinkle softphone. but after the call is dropped (T == 32 secondes) by my softphone and after hanging up my hardphone (T == 60 seconds) I have received automatically a call from my twinkle softphone!!! In wireshark trace, I see that OpenBTS is trying to ACK the OK from Asterisk, but Asterisk doesn't like it !!! I have tried to modify the value of the SIP timers, that works only from a hardphone to a softphone but not from hard to hard. can some one tell us what's the definition of t1min and timert1? t1min=1000 timert1=5000 timerb=32000 Any help will be appreciated. A.H. Jos, -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110721/f310f4ff/attachment.htm>