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<P>HI</P>
<P> </P>
<P>I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)</P>
<P> </P>
<P>Out going calls from asterisk to the ata works fine</P>
<P>Incoming calls from the ata to asterisk cuts off with the error msg</P>
<P> </P>
<P>Maximum retries exceeded on transmission <A href="mailto:70854efe-4157e3a8@10.168.7.103" rel=nofollow target=_blank ymailto="mailto:70854efe-4157e3a8@10.168.7.103">70854efe-4157e3a8@10.168.7.103</A> for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.<BR>[Dec 6 13:52:43] WARNING[3921]: chan_sip.c:3858 retrans_pkt:</P>
<P> Hanging up call <A href="mailto:70854efe-4157e3a8@10.168.7.103" rel=nofollow target=_blank ymailto="mailto:70854efe-4157e3a8@10.168.7.103">70854efe-4157e3a8@10.168.7.103</A> - no reply to our critical packet (see doc/sip-retransmit.txt).<BR></P>
<P>I been googling this error and it was mentioned to use</P>
<P>t1min= 500 however its only delaying the problem.</P>
<P> </P>
<P>any ideas on what is the cause of this problem.</P>
<P>Only 2-3 atas are having this problem the rest are fine.</P>
<P> </P>
<P>Here is the sip debug</P>
<P>the sip invites are not being received</P>
<P>and in one of the message a busy response was sent back.</P>
<P> </P><SPAN lang="">
<P><FONT color=#ff0000 size=1>Retransmitting #4 (no NAT) to 10.168.7.103:5060:</FONT></P>
<P><FONT color=#ff0000 size=1>SIP/2.0 200 OK</FONT></P>
<P><FONT color=#ff0000 size=1>Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103</FONT></P>
<P><FONT color=#ff0000 size=1>From: Ridge <sip:287@10.10.0.1>;tag=f314fd35733eba9bo0</FONT></P>
<P><FONT color=#ff0000 size=1>To: <sip:204@10.10.0.1>;tag=as4593172b</FONT></P>
<P><FONT color=#ff0000 size=1>Call-ID: f54a1cbd-891ce0b3@10.168.7.103</FONT></P>
<P><FONT color=#ff0000 size=1>CSeq: 102 INVITE</FONT></P>
<P><FONT color=#ff0000 size=1>Server: Asterisk PBX 1.6.2.13</FONT></P>
<P><FONT color=#ff0000 size=1>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</FONT></P>
<P><FONT color=#ff0000 size=1>Supported: replaces, timer</FONT></P>
<P><FONT color=#ff0000 size=1>Contact: <sip:204@41.146.208.131></FONT></P>
<P><FONT color=#ff0000 size=1>Content-Type: application/sdp</FONT></P>
<P><FONT color=#ff0000 size=1>Content-Length: 337</FONT></P>
<P><FONT color=#ff0000 size=1>v=0</FONT></P>
<P><FONT color=#ff0000 size=1>o=root 777980638 777980638 IN IP4 41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>s=Asterisk PBX 1.6.2.13</FONT></P>
<P><FONT color=#ff0000 size=1>c=IN IP4 41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>t=0 0</FONT></P>
<P><FONT color=#ff0000 size=1>m=audio 19726 RTP/AVP 8 0 18 101</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:8 PCMA/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:0 PCMU/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:18 G729/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=fmtp:18 annexb=no</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:101 telephone-event/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=fmtp:101 0-16</FONT></P>
<P><FONT color=#ff0000 size=1>a=silenceSupp:off - - - -</FONT></P>
<P><FONT color=#ff0000 size=1>a=ptime:20</FONT></P>
<P><FONT color=#ff0000 size=1>a=sendrecv</FONT></P>
<P><FONT color=#ff0000 size=1>---</FONT></P>
<P><FONT color=#ff0000 size=1></FONT> </P>
<P><FONT color=#ff0000 size=1>Retransmitting #5 (no NAT) to 10.168.7.103:5060:</FONT></P>
<P><FONT color=#ff0000 size=1>SIP/2.0 200 OK</FONT></P>
<P><FONT color=#ff0000 size=1>Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103</FONT></P>
<P><FONT color=#ff0000 size=1>From: Ridge <sip:287@10.10.0.1>;tag=f314fd35733eba9bo0</FONT></P>
<P><FONT color=#ff0000 size=1>To: <sip:204@10.10.0.1>;tag=as4593172b</FONT></P>
<P><FONT color=#ff0000 size=1>Call-ID: f54a1cbd-891ce0b3@10.168.7.103</FONT></P>
<P><FONT color=#ff0000 size=1>CSeq: 102 INVITE</FONT></P>
<P><FONT color=#ff0000 size=1>Server: Asterisk PBX 1.6.2.13</FONT></P>
<P><FONT color=#ff0000 size=1>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</FONT></P>
<P><FONT color=#ff0000 size=1>Supported: replaces, timer</FONT></P>
<P><FONT color=#ff0000 size=1>Contact: <sip:204@41.146.208.131></FONT></P>
<P><FONT color=#ff0000 size=1>Content-Type: application/sdp</FONT></P>
<P><FONT color=#ff0000 size=1>Content-Length: 337</FONT></P>
<P><FONT color=#ff0000 size=1>v=0</FONT></P>
<P><FONT color=#ff0000 size=1>o=root 777980638 777980638 IN IP4 41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>s=Asterisk PBX 1.6.2.13</FONT></P>
<P><FONT color=#ff0000 size=1>c=IN IP4 41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>t=0 0</FONT></P>
<P><FONT color=#ff0000 size=1>m=audio 19726 RTP/AVP 8 0 18 101</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:8 PCMA/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:0 PCMU/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:18 G729/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=fmtp:18 annexb=no</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:101 telephone-event/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=fmtp:101 0-16</FONT></P>
<P><FONT color=#ff0000 size=1>a=silenceSupp:off - - - -</FONT></P>
<P><FONT color=#ff0000 size=1>a=ptime:20</FONT></P>
<P><FONT color=#ff0000 size=1>a=sendrecv</FONT></P>
<P><FONT color=#ff0000 size=1>---</FONT></P>
<P><FONT color=#ff0000 size=1>Reliably Transmitting (no NAT) to 10.168.7.103:5060:</FONT></P>
<P><FONT color=#ff0000 size=1>OPTIONS sip:287@10.168.7.103:5060 SIP/2.0</FONT></P>
<P><FONT color=#ff0000 size=1>Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89;rport</FONT></P>
<P><FONT color=#ff0000 size=1>Max-Forwards: 70</FONT></P>
<P><FONT color=#ff0000 size=1>From: "asterisk" <sip:asterisk@41.146.208.131>;tag=as21bdce7e</FONT></P>
<P><FONT color=#ff0000 size=1>To: <sip:287@10.168.7.103:5060></FONT></P>
<P><FONT color=#ff0000 size=1>Contact: <sip:asterisk@41.146.208.131></FONT></P>
<P><FONT color=#ff0000 size=1>Call-ID: 062f8f6c4e7f5929487f3db12a93f7c2@41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>CSeq: 102 OPTIONS</FONT></P>
<P><FONT color=#ff0000 size=1>User-Agent: Asterisk PBX 1.6.2.13</FONT></P>
<P><FONT color=#ff0000 size=1>Date: Mon, 06 Dec 2010 12:47:41 GMT</FONT></P>
<P><FONT color=#ff0000 size=1>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</FONT></P>
<P><FONT color=#ff0000 size=1>Supported: replaces, timer</FONT></P>
<P><FONT color=#ff0000 size=1>Content-Length: 0</FONT></P>
<P><FONT color=#ff0000 size=1> </FONT></P>
<P><FONT color=#ff0000 size=1>---</FONT></P>
<P><FONT color=#ff0000 size=1><--- SIP read from UDP:10.168.7.103:5060 ---></FONT></P>
<P><FONT color=#000000 size=1><STRONG>SIP/2.0 486 Busy Here</STRONG></FONT></P>
<P><FONT color=#ff0000 size=1>To: <sip:287@10.168.7.103:5060>;tag=18c8b9ab85ca5068i0</FONT></P>
<P><FONT color=#ff0000 size=1>From: "asterisk" <sip:asterisk@41.146.208.131>;tag=as21bdce7e</FONT></P>
<P><FONT color=#ff0000 size=1>Call-ID: 062f8f6c4e7f5929487f3db12a93f7c2@41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>CSeq: 102 OPTIONS</FONT></P>
<P><FONT color=#ff0000 size=1>Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89</FONT></P>
<P><FONT color=#ff0000 size=1>Server: Linksys/SPA3102-3.3.6(GW)</FONT></P>
<P><FONT color=#ff0000 size=1>Content-Length: 0</FONT></P>
<P><FONT color=#ff0000 size=1>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER</FONT></P>
<P><FONT color=#ff0000 size=1>Supported: x-sipura</FONT></P>
<P><FONT color=#ff0000 size=1> </FONT></P>
<P><FONT color=#ff0000 size=1><-------------></FONT></P>
<P><FONT color=#ff0000 size=1>--- (10 headers 0 lines) ---</FONT></P>
<P><FONT color=#ff0000 size=1>Really destroying SIP dialog '062f8f6c4e7f5929487f3db12a93f7c2@41.146.208.131' Method: OPTIONS</FONT></P>
<P><FONT color=#ff0000 size=1>Retransmitting #6 (no NAT) to 10.168.7.103:5060:</FONT></P>
<P><FONT color=#ff0000 size=1>SIP/2.0 200 OK</FONT></P>
<P><FONT color=#ff0000 size=1>Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103</FONT></P>
<P><FONT color=#ff0000 size=1>From: Ridge <sip:287@10.10.0.1>;tag=f314fd35733eba9bo0</FONT></P>
<P><FONT color=#ff0000 size=1>To: <sip:204@10.10.0.1>;tag=as4593172b</FONT></P>
<P><FONT color=#ff0000 size=1>Call-ID: f54a1cbd-891ce0b3@10.168.7.103</FONT></P>
<P><FONT color=#ff0000 size=1>CSeq: 102 INVITE</FONT></P>
<P><FONT color=#ff0000 size=1>Server: Asterisk PBX 1.6.2.13</FONT></P>
<P><FONT color=#ff0000 size=1>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</FONT></P>
<P><FONT color=#ff0000 size=1>Supported: replaces, timer</FONT></P>
<P><FONT color=#ff0000 size=1>Contact: <sip:204@41.146.208.131></FONT></P>
<P><FONT color=#ff0000 size=1>Content-Type: application/sdp</FONT></P>
<P><FONT color=#ff0000 size=1>Content-Length: 337</FONT></P>
<P><FONT color=#ff0000 size=1>v=0</FONT></P>
<P><FONT color=#ff0000 size=1>o=root 777980638 777980638 IN IP4 41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>s=Asterisk PBX 1.6.2.13</FONT></P>
<P><FONT color=#ff0000 size=1>c=IN IP4 41.146.208.131</FONT></P>
<P><FONT color=#ff0000 size=1>t=0 0</FONT></P>
<P><FONT color=#ff0000 size=1>m=audio 19726 RTP/AVP 8 0 18 101</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:8 PCMA/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:0 PCMU/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:18 G729/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=fmtp:18 annexb=no</FONT></P>
<P><FONT color=#ff0000 size=1>a=rtpmap:101 telephone-event/8000</FONT></P>
<P><FONT color=#ff0000 size=1>a=fmtp:101 0-16</FONT></P>
<P><FONT color=#ff0000 size=1>a=silenceSupp:off - - - -</FONT></P>
<P><FONT color=#ff0000 size=1>a=ptime:20</FONT></P>
<P><FONT color=#ff0000 size=1>a=sendrecv</FONT></P></SPAN>
<P> </P>
<P> </P>
<P>zakir</P>
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