<div dir="ltr"><div><div><div>Hi Stas,<br><br></div><div>I assume:<br><br></div><div>UA => OpenSIPS => Freeswitch => VoIP provider<br></div><div><br></div><div><a href="http://172.31.0.169:5060">172.31.0.169:5060</a> - OpenSIPS<br></div><div><a href="http://172.31.22.124:5060">172.31.22.124:5060</a> - Freeswitch<br></div><div>178.*.*.12:5060 - VoIP provider<br><br></div><div><br></div>From log what you provided it looks like OpenSIPS is not configured properly...<br><br></div>If you check RFC:<br><br><a href="https://tools.ietf.org/html/rfc3665#section-3.1" target="_blank">https://tools.ietf.org/html/rfc3665#section-3.1</a><br><br></div>Successful call looks like this:<br><br><pre> Alice Bob
| |
| INVITE F1 |
|----------------------->|
| 180 Ringing F2 |
|<-----------------------|
| |
| 200 OK F3 |
|<-----------------------|
| ACK F4 |
|----------------------->|
| Both Way RTP Media |
|<======================>|
| |
| BYE F5 |
|<-----------------------|
| 200 OK F6 |
|----------------------->|
| |<br><br><br></pre><pre>In your case somehow Freeswitch never receive ACK from OpenSIPS.<br><br></pre><pre>As you can see following packet was sent several times by Freeswitch:<br><br></pre><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">U <a href="http://172.31.22.124:5060">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br></blockquote><pre>And on this OK Freeswitch should receive ACK, but never receive it from Opensips and because of this FreeSWITCH hang-up call:<br><br></pre><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">U <a href="http://172.31.22.124:5060">172.31.22.124:5060</a> -> 52.*.*.177:5060<br>BYE sip:8@85.*.*.4:53712;ob;transport=tls SIP/2.0.<br>Via: SIP/2.0/UDP 172.31.22.124;rport;branch=z9hG4bK7e89BXDX701yc.<br>Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>Max-Forwards: 70.<br>From: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>To: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 89844917 BYE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Reason: SIP;cause=408;text="ACK Timeout".<br>Content-Length: 0.<br></blockquote><pre>As you can see BYE is strange too... Bye should be sent back to OpenSIPS, but not to 52.*.*.177:5060.<br><br></pre><pre>On this point I think you have incorrect Record-Route records:<br><br></pre><pre>Good example you can find below, please check 16.12.1.1.<br><br><a href="https://www.ietf.org/rfc/rfc3261.txt">https://www.ietf.org/rfc/rfc3261.txt</a><br></pre><pre>Somehow you have following record-route record:<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr> but your opensips never use 5061 port, it is using 5060, so something is incorrect here...<br><br></pre><pre>On this point I think you have miss-configured OpenSIPS.<br></pre><pre>With kind regards,<br></pre></div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature"><div dir="ltr">Jurijs<br></div></div></div>
<br><div class="gmail_quote">On Mon, Apr 11, 2016 at 12:59 PM, Стас Тельнов <span dir="ltr"><<a href="mailto:stasan89@gmail.com" target="_blank">stasan89@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div><div><div><div>>>Artur Mega: Do you use sip-proxy? If you use sip proxy, maybe you forget to add "Record-Route" header to sip-query<br></div>"Record-Route" header already exists.<br><br>>>Regis M: 99% of the time 30 seconds hangup (or 32seconds) means a NAT problem... or a response not come back to source...<br></div>I understand, but I don`t know how fix it.<br><br>Jurijs Ivolga,<br></div>I analize packets in ngrep.<br></div>First BYE packet include reason: NORMAL_CLEARING. BYE packet for caller include reason: ACK Timeout.<br></div>About NORMAL_CLEARING in freeswitch documentation (<a href="https://wiki.freeswitch.org/wiki/Hangup_Causes" target="_blank">https://wiki.freeswitch.org/wiki/Hangup_Causes</a>):<br>This cause indicates that the call is being cleared because one of the users involved in the call has requested that the call be cleared. Under normal situations, the source of this cause is not the network. <br><br></div>But phone clients do not send this command.<br><br></div><span lang="en"><span>I suppose</span> <span>that this is</span> <span>because the server</span> <span>has not received</span> <span>confirmation</span> <span>from one of the</span> <span>customers</span> <span>that the call</span> <span>took place.<br><br></span></span></div><span lang="en"><span>All packages on call:<br></span></span></div><span lang="en"><span>#<br>U <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a> -> <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a><br>INVITE sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a> SIP/2.0.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Max-Forwards: 69.<br>Contact: <sip:8@85.*.*.4:53712;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>.<br>From: "8"<<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO.<br>Content-Type: application/sdp.<br>Supported: replaces, outbound, path.<br>User-Agent: PortSIP SDK for IOS.<br>Content-Length: 219.<br>.<br>v=0.<br>o=- 1460361915 1 IN IP4 85.*.*.4.<br>s=<a href="http://portsip.com" target="_blank">portsip.com</a>.<br>c=IN IP4 52.*.*.177.<br>t=0 0.<br>m=audio 40772 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=sendrecv.<br>a=nortpproxy:yes.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> 178.*.*.12:5060<br>INVITE sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a> SIP/2.0.<br>Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK5vNr86BpDFNSN.<br>Max-Forwards: 67.<br>From: "8" <sip:21***@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=52eDp9a81B1mg.<br>To: <sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>.<br>Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.<br>CSeq: 89844894 INVITE.<br>Contact: <sip:gw+freelycall.com@52.*.*.198:5060;transport=udp;gw=<a href="http://freelycall.com" target="_blank">freelycall.com</a>>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 244.<br>X-FS-Support: update_display,send_info.<br>Remote-Party-ID: "8" <<a href="mailto:sip%3A8@freelycall.com" target="_blank">sip:8@freelycall.com</a>>;party=calling;screen=yes;privacy=off.<br>.<br>v=0.<br>o=FreeSWITCH 1460338046 1460338047 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 23870 RTP/AVP 8 101 13.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=rtpmap:13 CN/8000.<br>a=ptime:20.<br><br>#<br>U 178.*.*.12:5060 -> <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a><br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.<br>From: "8" <sip:21***@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=52eDp9a81B1mg.<br>To: <sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>.<br>Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.<br>CSeq: 89844894 INVITE.<br>Server: Asterisk PBX 11.11.0.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.<br>Supported: replaces.<br>Contact: <sip:7906*******@178.*.*.12:5060>.<br>Content-Length: 0.<br>.<br><br>#<br>U 178.*.*.12:5060 -> <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a><br>SIP/2.0 183 Session Progress.<br>Via: SIP/2.0/UDP 52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.<br>From: "8" <sip:21***@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=52eDp9a81B1mg.<br>To: <sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=as75cc44fb.<br>Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.<br>CSeq: 89844894 INVITE.<br>Server: Asterisk PBX 11.11.0.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.<br>Supported: replaces.<br>Contact: <sip:7906*******@178.*.*.12:5060>.<br>Content-Type: application/sdp.<br>Content-Length: 240.<br>.<br>v=0.<br>o=root 1395806664 1395806664 IN IP4 178.*.*.12.<br>s=Asterisk PBX 11.11.0.<br>c=IN IP4 178.*.*.12.<br>t=0 0.<br>m=audio 11574 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 183 Session Progress.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Accept: application/sdp.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U 178.*.*.12:5060 -> <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.<br>From: "8" <sip:21***@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=52eDp9a81B1mg.<br>To: <sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=as75cc44fb.<br>Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.<br>CSeq: 89844894 INVITE.<br>Server: Asterisk PBX 11.11.0.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.<br>Supported: replaces.<br>Contact: <sip:7906*******@178.*.*.12:5060>.<br>Content-Type: application/sdp.<br>Content-Length: 240.<br>.<br>v=0.<br>o=root 1395806664 1395806664 IN IP4 178.*.*.12.<br>s=Asterisk PBX 11.11.0.<br>c=IN IP4 178.*.*.12.<br>t=0 0.<br>m=audio 11574 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> 178.*.*.12:5060<br>ACK sip:7906*******@178.*.*.12:5060 SIP/2.0.<br>Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK65eHa2vSarBcH.<br>Max-Forwards: 70.<br>From: "8" <sip:21***@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=52eDp9a81B1mg.<br>To: <sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=as75cc44fb.<br>Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.<br>CSeq: 89844894 ACK.<br>Contact: <sip:gw+freelycall.com@52.*.*.198:5060;transport=udp;gw=<a href="http://freelycall.com" target="_blank">freelycall.com</a>>.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.<br>Via: SIP/2.0/TLS 192.168.0.110:10977;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.<br>Record-Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>From: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>To: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 2 INVITE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 220.<br>Remote-Party-ID: "Outbound Call" <sip:7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;party=calling;privacy=off;screen=no.<br>.<br>v=0.<br>o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.<br>s=FreeSWITCH.<br>c=IN IP4 52.*.*.198.<br>t=0 0.<br>m=audio 27728 RTP/AVP 8 101.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> 178.*.*.12:5060<br>BYE sip:7906*******@178.*.*.12:5060 SIP/2.0.<br>Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK8Q12Dry049QHr.<br>Max-Forwards: 70.<br>From: "8" <sip:21***@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=52eDp9a81B1mg.<br>To: <sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=as75cc44fb.<br>Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.<br>CSeq: 89844895 BYE.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Reason: Q.850;cause=16;text="NORMAL_CLEARING".<br>Content-Length: 0.<br>.<br><br>#<br>U 178.*.*.12:5060 -> <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 52.*.*.198;branch=z9hG4bK8Q12Dry049QHr;received=52.*.*.198;rport=5060.<br>From: "8" <sip:21***@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=52eDp9a81B1mg.<br>To: <sip:7906*******@<a href="http://freelycall.com" target="_blank">freelycall.com</a>>;tag=as75cc44fb.<br>Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.<br>CSeq: 89844895 BYE.<br>Server: Asterisk PBX 11.11.0.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.<br>Supported: replaces.<br>Content-Length: 0.<br>.<br><br>#<br>U <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a> -> 52.*.*.177:5060<br>BYE sip:8@85.*.*.4:53712;ob;transport=tls SIP/2.0.<br>Via: SIP/2.0/UDP 172.31.22.124;rport;branch=z9hG4bK7e89BXDX701yc.<br>Route: <sip:<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>;r2=on;lr>.<br>Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.<br>Max-Forwards: 70.<br>From: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>To: "8" <<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 89844917 BYE.<br>Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.<br>User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.<br>Supported: timer, path, replaces.<br>Reason: SIP;cause=408;text="ACK Timeout".<br>Content-Length: 0.<br>.<br><br>#<br>U 52.*.*.177:5060 -> <a href="http://172.31.22.124:5060" target="_blank">172.31.22.124:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 172.31.22.124;received=52.*.*.198;rport=5060;branch=z9hG4bK7e89BXDX701yc.<br>Contact: <sip:8@192.168.0.110:10977;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".<br>To: "8"<<a href="mailto:sip%3A8@sip0.MY_SIP_DOMAIN.com" target="_blank">sip:8@sip0.MY_SIP_DOMAIN.com</a>>;tag=4c913b30.<br>From: <sip:*7906*******@<a href="http://sip0.MY_SIP_DOMAIN.com" target="_blank">sip0.MY_SIP_DOMAIN.com</a>>;tag=4SNmmet442a2m.<br>Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...<br>CSeq: 89844917 BYE.<br>User-Agent: PortSIP SDK for IOS.<br>Content-Length: 0.<br>.<br><br></span></span><div><div><div><div><div><div><div><br><div><div><div><div><br></div></div></div></div></div></div></div></div></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">2016-04-08 20:01 GMT+03:00 Artur Mega <span dir="ltr"><<a href="mailto:findmeinwland@gmail.com" target="_blank">findmeinwland@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div class="gmail_default" style="font-size:small">Do you use sip-proxy? If you use sip proxy, maybe you forget to add "Record-Route" header to sip-query</div></div><div class="gmail_extra"><div><div><br><div class="gmail_quote">2016-04-08 21:13 GMT+05:00 Regis M <span dir="ltr"><<a href="mailto:regis.freeswitch.org@tornad.net" target="_blank">regis.freeswitch.org@tornad.net</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">99% of the time 30 seconds hangup (or 32seconds) means a NAT problem... or a response not come back to source...</div><div><div><div class="gmail_extra"><br><div class="gmail_quote">2016-04-08 17:06 GMT+02:00 Jurijs Ivolga <span dir="ltr"><<a href="mailto:jurijs.ivolga@gmail.com" target="_blank">jurijs.ivolga@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Hi,<br><br></div>This is not what I need, please use ngrep:<br><br><a href="http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/" target="_blank">http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/</a><br><br></div>With kind regards,<br></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">Jurijs<br></div></div></div><div><div>
<br><div class="gmail_quote">On Fri, Apr 8, 2016 at 6:02 PM, Стас Тельнов <span dir="ltr"><<a href="mailto:stasan89@gmail.com" target="_blank">stasan89@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Yes, of cause. I hide some ip and real phone numbers.<br>178.*.*.12 - ip of provider.<br><br><b>On start call:</b><br>2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> [c618eafe-fd98-11e5-a353-831849fc41a3]<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_NEW<br>2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> receiving invite from <a href="http://172.31.0.169:5060" target="_blank">172.31.0.169:5060</a> version: 1.6.6 64bit<br>2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> entering state [received][100]<br>2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:<br>v=0<br>o=- 1460126829 1 IN IP4 85.*.*.4<br>s=<a href="http://portsip.com" target="_blank">portsip.com</a><br>c=IN IP4 52.*.*.177<br>t=0 0<br>m=audio 40082 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=nortpproxy:yes<br><br>2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State Change CS_NEW -> CS_INIT<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State NEW<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_INIT<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State INIT<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> SOFIA INIT<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard INIT<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State Change CS_INIT -> CS_ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State INIT going to sleep<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Callstate Change DOWN -> RINGING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> SOFIA ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard ROUTING<br>2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8 <8>->7906******* in context public<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> parsing [public->from_opensips] continue=false<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Regex (PASS) [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/ break=on-false<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Action transfer(${destination_number} XML default)<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State Change CS_ROUTING -> CS_EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State ROUTING going to sleep<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> SOFIA EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard EXECUTE<br>EXECUTE sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> transfer(7906******* XML default)<br>2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State Change CS_EXECUTE -> CS_ROUTING<br>2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> to XML[7906*******@default]<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State EXECUTE going to sleep<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> SOFIA ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard ROUTING<br>2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8 <8>->7906******* in context default<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> parsing [default->unloop] continue=false<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> parsing [default->tod_example] continue=true<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Date/Time Match (PASS) [tod_example] break=on-false<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Action set(open=true)<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> parsing [default->outbound_calls_to_freelycall] continue=false<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Regex (PASS) [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/ break=on-true<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Action set(hangup_after_bridge=true)<br>Dialplan: sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Action bridge(sofia/gateway/<a href="http://freelycall.com/7906*******" target="_blank">freelycall.com/7906*******</a>)<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State Change CS_ROUTING -> CS_EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State ROUTING going to sleep<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> SOFIA EXECUTE<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard EXECUTE<br>EXECUTE sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> set(open=true)<br>2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> [open]=[true]<br>EXECUTE sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> set(hangup_after_bridge=true)<br>2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> [hangup_after_bridge]=[true]<br>EXECUTE sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> bridge(sofia/gateway/<a href="http://freelycall.com/7906*******" target="_blank">freelycall.com/7906*******</a>)<br>2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables<br>2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776 (sofia/external/7906*******) State Change CS_NEW -> CS_INIT<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_INIT<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516 (sofia/external/7906*******) State INIT<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/7906******* SOFIA INIT<br>2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257 sofia/external/7906******* sending invite version: 1.6.6 64bit<br>Local SDP:<br>v=0<br>o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198<br>s=FreeSWITCH<br>c=IN IP4 52.*.*.198<br>t=0 0<br>m=audio 26402 RTP/AVP 8 101 13<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=rtpmap:13 CN/8000<br>a=ptime:20<br>a=sendrecv<br><br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40 sofia/external/7906******* Standard INIT<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48 (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516 (sofia/external/7906*******) State INIT going to sleep<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/7906******* entering state [calling][0]<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532 (sofia/external/7906*******) State ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/7906******* SOFIA ROUTING<br>2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67 (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532 (sofia/external/7906*******) State ROUTING going to sleep<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551 (sofia/external/7906*******) State CONSUME_MEDIA<br>2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551 (sofia/external/7906*******) State CONSUME_MEDIA going to sleep<br>2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/7906******* entering state [proceeding][183]<br>2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:<br>v=0<br>o=root 153112258 153112258 IN IP4 178.*.*.12<br>s=Asterisk PBX 11.11.0<br>c=IN IP4 178.*.*.12<br>t=0 0<br>m=audio 17362 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br><br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101@8000<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111 sofia/external/7906******* Original read codec set to PCMA:8<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101@8000<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485 sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload to 101<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port 17362 codec: 8 ms: 20<br>2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332 sofia/external/7906******* Set 2833 dtmf send payload to 101<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339 sofia/external/7906******* Set 2833 dtmf receive payload to 101<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362 sofia/external/7906******* Set rtp dtmf delay to 40<br>2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/7906*******!<br>2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468 (sofia/external/7906*******) Callstate Change DOWN -> EARLY<br>2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending early media<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101@8000<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> PCMA/8000 20 ms 160 samples 64000 bits 1 channels<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Original read codec set to PCMA:8<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101@8000<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Set 2833 dtmf send payload to 101 recv payload to 101<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>] 172.31.22.124 port 30630 -> 52.*.*.177 port 40082 codec: 8 ms: 20<br>2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Set 2833 dtmf send payload to 101<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Set 2833 dtmf receive payload to 101<br>2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Set rtp dtmf delay to 40<br>2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>!<br>2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Callstate Change RINGING -> EARLY<br>2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:<br>v=0<br>o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198<br>s=FreeSWITCH<br>c=IN IP4 52.*.*.198<br>t=0 0<br>m=audio 30630 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br>2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> entering state [early][183]<br>2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/external/7906*******]<br>2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591 (sofia/external/7906*******) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA<br>2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA<br>2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542 (sofia/external/7906*******) State EXCHANGE_MEDIA<br>2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA<br>2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.<br>2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.<br>2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/7906******* entering state [completing][200]<br>2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP<br>v=0<br>o=root 153112258 153112258 IN IP4 178.*.*.12<br>s=Asterisk PBX 11.11.0<br>c=IN IP4 178.*.*.12<br>t=0 0<br>m=audio 17362 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br><br>2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/7906******* entering state [ready][200]<br>2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel [sofia/external/7906*******] has been answered<br>2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767 (sofia/external/7906*******) Callstate Change EARLY -> ACTIVE<br>2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>:<br>v=0<br>o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198<br>s=FreeSWITCH<br>c=IN IP4 52.*.*.198<br>t=0 0<br>m=audio 30630 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br>2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> entering state [completed][200]<br>2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel [sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>] has been answered<br>2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Callstate Change EARLY -> ACTIVE<br>2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.<br>2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.<br><br><br><b>And after 30 seconds:</b><br>2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> entering state [terminating][0]<br>2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> [CS_EXECUTE] [NORMAL_UNSPECIFIED]<br>2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> ending bridge by request from write function<br>2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/external/7906*******]<br>2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542 (sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_HANGUP<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739 (sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741 (sofia/external/7906*******) State HANGUP<br>2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel sofia/external/7906******* hanging up, cause: NORMAL_CLEARING<br>2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/external/7906*******<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60 sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741 (sofia/external/7906*******) State HANGUP going to sleep<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508 (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_REPORTING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827 (sofia/external/7906*******) State REPORTING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104 sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827 (sofia/external/7906*******) State REPORTING going to sleep<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499 (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 2 (sofia/external/7906*******) Locked, Waiting on external entities<br>2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> ending bridge by request from read function<br>2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>]<br>2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> skip receive message [UNBRIDGE] (channel is hungup already)<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State EXECUTE going to sleep<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_HANGUP<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Callstate Change ACTIVE -> HANGUP<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State HANGUP<br>2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> hanging up, cause: NORMAL_UNSPECIFIED<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard HANGUP, cause: NORMAL_UNSPECIFIED<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State HANGUP going to sleep<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State Change CS_HANGUP -> CS_REPORTING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_REPORTING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State REPORTING<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard REPORTING, cause: NORMAL_UNSPECIFIED<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State REPORTING going to sleep<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State Change CS_REPORTING -> CS_DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 1 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Locked, Waiting on external entities<br>2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 1 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Ended<br>2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> [CS_DESTROY]<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) Running State Change CS_DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> SOFIA DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111 sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a> Standard DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640 (sofia/external/<a href="mailto:8@sip0.MY_DOMAIN.com" target="_blank">8@sip0.MY_DOMAIN.com</a>) State DESTROY going to sleep<br>2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 2 (sofia/external/7906*******) Ended<br>2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/7906******* [CS_DESTROY]<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630 (sofia/external/7906*******) Running State Change CS_DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640 (sofia/external/7906*******) State DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/7906******* SOFIA DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111 sofia/external/7906******* Standard DESTROY<br>2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640 (sofia/external/7906*******) State DESTROY going to sleep<div><div><br><div class="gmail_extra"><br><div class="gmail_quote">2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <span dir="ltr"><<a href="mailto:jurijs.ivolga@gmail.com" target="_blank">jurijs.ivolga@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div><div>Hi,<br><br></div>I would recommend you to capture SIP packets during call on Freeswitch server and send it here, I will take a look on it.<br><br></div>With kind regards,<br></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">Jurijs<br></div></div></div><div><div>
<br><div class="gmail_quote">On Fri, Apr 8, 2016 at 5:34 PM, Стас Тельнов <span dir="ltr"><<a href="mailto:stasan89@gmail.com" target="_blank">stasan89@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><span lang="en"><span>I already</span> <span>tried </span></span><span lang="en"><span>disabling timers</span><span>,</span> <span>does not work.</span></span></div><div class="gmail_extra"><br><div class="gmail_quote">2016-04-08 17:19 GMT+03:00 Oleg Stolyar <span dir="ltr"><<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Try disabling session timers in the sip profile. I think that line is commented out by default, so uncomment it.<div><br><div><param name="enable-timer" value="false"/><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Fri, Apr 8, 2016 at 6:59 AM, Стас Тельнов <span dir="ltr"><<a href="mailto:stasan89@gmail.com" target="_blank">stasan89@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><div><div dir="ltr">Hello.<br><br>When using a call or conference through sip — freeswitch with external provider there is a problem – the call is interrupted in 30 seconds. Though the sound goes all right.<br>I think that it caused by the NAT settings for freeswitch, but I don't understand how to adjust it correctly.<br>At start of freeswitch I see the following mistakes in the tracking data:<br><font size="2">2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT<br>2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5<br>2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5<br>2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5<br>2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5<br>2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5<br>2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP [general error]<br>2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP<br>2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT devices detected!</font><br><br>Despite of this mistake, conference communication between two internal users works normally. The problem arises at a call through external provider.<br><br>We have the following architecture:<br>In a cloud of Amazon EC2 there are 2 servers – opensips and freeswitch, both for NAT for external clients, but have an opportunity to work with each other directly.<br>opensips has the internal address 172.31.0.169 and external 52. *.*.177<br>freeswitch has the internal address 172.31.22.124 and external 52. *.*.198<br><br>In fact, freeswitch acts only for conferences, and is ready for use of a remote DB on opensips.<br>The auto-nat settings by default didn't work. The problem is in the external profile settings as far as I understand.<br><br>I have filled and created the following configuration:<br>vars.xml <br> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto”/><br> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198”/> <!— public freeswitch ip —><br> <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198”/> <!— public freeswitch ip —><br> <!-- External SIP Profile --><br> <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/><br> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/><br> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/><br> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/><br> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/tls"/><br><br>sip_profile/external.xml<br> <param name="rtp-ip" value="$${local_ip_v4}"/><br> <param name="sip-ip" value="$${local_ip_v4}"/><br><br> <param name="ext-rtp-ip" value=“52.*.*.198”/> <!— public freeswitch ip —><br> <param name="ext-sip-ip" value=“52.*.*.198”/> <!— public freeswitch ip —><br>In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server (that would be logical), but in that case conferences didn't work at all and errors below appeared:<br>[ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...<br>Also I tried to put such configuration:<br><span style="color:rgb(0,0,0)"> <param name="rtp-ip" value="auto"/><br> <param name="sip-ip" value="52.*.*.198”/></span><br>but it also hasn't helped to solve the problem.<br><br>autoload_configs/switch.conf.xml <br> <param name="rtp-start-port" value="16384"/><br> <param name="rtp-end-port" value="32768"/><br><br>"sofia status" looks as follows:<br> Name Type Data State<br>=================================================================================================<br> 172.31.22.124 alias internal ALIASED<br> external profile sip:mod_sofia@52.*.*.198:5060 RUNNING (0)<br> external profile sip:mod_sofia@52.*.*.198:5061 RUNNING (0) (TLS)<br> external::*********.com gateway sip:USER@*********.com REGED<br> internal profile sip:mod_sofia@52.*.*.198:5080 RUNNING (0)<br> internal profile sip:mod_sofia@52.*.*.198:5081 RUNNING (0) (TLS)<br>=================================================================================================<br>2 profiles 1 alias<br><br>"sofia status profile external" looks as follows:<br>=================================================================================================<br>Name external<br>Domain Name N/A<br>Auto-NAT false<br>DBName sofia_reg_external<br>Pres Hosts <br>Dialplan XML<br>Context public<br>Challenge Realm auto_to<br>RTP-IP 172.31.22.124<br>Ext-RTP-IP 52.*.*.198<br>SIP-IP 172.31.22.124<br>Ext-SIP-IP 52.*.*.198<br>URL sip:mod_sofia@52.*.*.198:5060<br>BIND-URL sip:mod_sofia@52.*.*.198:5060;maddr=172.31.22.124;transport=udp,tcp<br>TLS-URL sip:mod_sofia@52.*.*.198:5061<br>TLS-BIND-URL sips:mod_sofia@52.*.*.198:5061;maddr=172.31.22.124;transport=tls<br>HOLD-MUSIC local_stream://moh<br>OUTBOUND-PROXY N/A<br>CODECS IN PCMA<br>CODECS OUT PCMA<br>TEL-EVENT 101<br>DTMF-MODE rfc2833<br>CNG 13<br>SESSION-TO 0<br>MAX-DIALOG 0<br>NOMEDIA false<br>LATE-NEG true<br>PROXY-MEDIA false<br>ZRTP-PASSTHRU true<br>AGGRESSIVENAT false<br>CALLS-IN 0<br>FAILED-CALLS-IN 0<br>CALLS-OUT 0<br>FAILED-CALLS-OUT 0<br>REGISTRATIONS 0<br><br><br><br>What do I adjust wrong? Whether there is some opportunity, to tell freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?<br></div>
<br></div></div>_________________________________________________________________________<br>
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FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div></div></div></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div></div></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div>
</div></div><br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div></div></div><span><font color="#888888">-- <br><div><div dir="ltr"><div><div><br>Arthur</div></div></div></div>
</font></span></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div>