HI list,<br>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:<br>
<br>[Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission <a href="mailto:1348333597@127.0.0.1" target="_blank">1348333597@127.0.0.1</a> for seqno 94 (Critical Response) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br>
Packet timed out after 32000ms with no response<br>[Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3651 retrans_pkt: Hanging up call <a href="mailto:1348333597@127.0.0.1" target="_blank">1348333597@127.0.0.1</a> - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).<br>
<br>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!!!<br><br>Retransmitting #10 (no NAT) to <a href="http://127.0.0.1:5062/" target="_blank">127.0.0.1:5062</a>:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK53934;received=127.0.0.<div>1<br>From: IMSI208012601160193 <<a href="mailto:sip%3AIMSI208012601160193@127.0.0.1" target="_blank">sip:IMSIxxxxxx@127.0.0.1</a>>;tag=lkbdg<br>
To: <<a href="mailto:sip%3A2121211@127.0.0.1" target="_blank">sip:2121211@127.0.0.1</a>>;tag=as7c57c466<br>Call-ID: <a href="mailto:1348333597@127.0.0.1" target="_blank">1348333597@127.0.0.1</a><br>CSeq: 94 INVITE<br>
Server: Asterisk PBX 1.8.5.0-1digium1~lucid<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Contact: <<a href="http://sip:2121211@127.0.0.1:5060/" target="_blank">sip:2121211@127.0.0.1:5060</a>><br>
Content-Type: application/sdp<br>
Content-Length: 213<br><br>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!!!<br>
In wireshark trace, I see that OpenBTS is trying to ACK the OK from Asterisk, but Asterisk doesn't like it !!!<br><br>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?<br>
t1min=1000<br>timert1=5000<br>timerb=32000 <br><br>Any help will be appreciated.<br>A.H. Jos,</div>