<div dir="ltr">You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I would highly recommend you re-test with Master or at the very least 1.4.17 or 1.4.18 which should be out later today. <div><br></div><div>1.2 is not receiving patches, updates or support moving forward, our release branch is 1.4.x<br><div><br></div><div><br></div><div><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Mar 18, 2015 at 1:26 PM, Örn Arnarson <span dir="ltr"><<a href="mailto:orn@arnarson.net" target="_blank">orn@arnarson.net</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">Hello,<div><br></div><div>Not sure whether this belong in the users list or the dev list, but when in doubt; start with users :-)<br><div><br></div><div>I am using FreeSWITCH as an SBC, talking to Kamailio on one and and Asterisk on the other, and am seeing some strange behavior when calls are being forked on the Asterisk.</div><div><br></div><div>Call setup is like this:</div><div>1. FreeSWITCH receives INVITE from Kamailio </div><div>2. FreeSWITCH sends INVITE to Asterisk with new Call-ID</div><div>3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH (each with a unique call-id)</div><div>4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new Call-ID from step 2.</div><div><br></div><div>This is causing problems with one of the MGWs behind Kamailio, which is seeing multiple INVITEs to different destinations with the same Call-ID.</div><div><br></div><div>So, firstly, why is FreeSWITCH reusing call-ids?</div><div><br></div><div>Secondly, how is it matching up the calls? I can't find anything common in the INVITEs, other than the source number and obviously that the IP sent to and received from is the same.<br></div><div><br></div><div>I'm not sure if this is intended behavior or not, but is there a way to have FreeSWITCH not do that?</div><div><br></div><div>Regards,<br></div><div>Örn</div><div><br></div><div>P.S. Here is the sequence of INVITEs. I also have the console log (for a different call) if needed.</div><div><br></div><div><b>INVITE sent to FreeSWITCH by Kamailio:</b></div><div><div>INVITE <a href="http://sip:5344446@172.25.200.111:5080" target="_blank">sip:5344446@172.25.200.111:5080</a> SIP/2.0</div><div>Record-Route: <sip:172.25.200.101;lr=on></div><div>Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0</div><div>Via: SIP/2.0/UDP 172.25.200.121:5080;rport=5080;branch=z9hG4bK9aFD5m2KKerHN</div><div>Max-Forwards: 16</div><div>From: "4151502" <<a href="mailto:sip%3A4151502@172.25.200.121" target="_blank">sip:4151502@172.25.200.121</a>>;tag=33vB4BmmDtU0B</div><div>To: <<a href="mailto:sip%3A5344446@172.25.200.101" target="_blank">sip:5344446@172.25.200.101</a>></div><div>Call-ID: 84b63791-4839-1233-639f-00215e2db0e0</div><div>CSeq: 73014324 INVITE</div><div>Contact: <<a href="http://sip:mod_sofia@172.25.200.121:5080" target="_blank">sip:mod_sofia@172.25.200.121:5080</a>></div><div>User-Agent: FreeSWITCH-mod_sofia/1.2.7</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY</div><div>Supported: timer, precondition, path, replaces</div><div>Allow-Events: talk, hold, conference, refer</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session</div><div>Content-Length: 229</div><div>X-FS-Support: update_display,send_info</div><div>Remote-Party-ID: "4151502" <<a href="mailto:sip%3A4151502@172.25.200.121" target="_blank">sip:4151502@172.25.200.121</a>>;party=calling;screen=yes;privacy=off</div><div><br></div><div>v=0</div><div>o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121</div><div>s=FreeSWITCH</div><div>c=IN IP4 172.25.200.121</div><div>t=0 0</div><div>m=audio 19026 RTP/AVP 8 101</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20</div></div><div><br></div><div><br></div><div><b>INVITE sent to Asterisk by FreeSWITCH:</b></div><div><div>INVITE <a href="http://sip:5344446@172.26.0.62:5060" target="_blank">sip:5344446@172.26.0.62:5060</a> SIP/2.0<br></div><div>Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF</div><div>Max-Forwards: 15</div><div>From: "4151502" <<a href="mailto:sip%3A4151502@10.11.12.13" target="_blank">sip:4151502@10.11.12.13</a>>;tag=2BaZj0t076Q9B</div><div>To: <<a href="http://sip:5344446@172.26.0.62:5060" target="_blank">sip:5344446@172.26.0.62:5060</a>></div><div>Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90</div><div>CSeq: 73014353 INVITE</div><div>Contact: <<a href="http://sip:mod_sofia@10.11.12.13:5060" target="_blank">sip:mod_sofia@10.11.12.13:5060</a>></div><div>User-Agent: FreeSWITCH-mod_sofia/1.2.12</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE</div><div>Supported: timer, precondition, path, replaces</div><div>Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session</div><div>Content-Length: 223</div><div>X-FS-Support: update_display,send_info</div><div>Remote-Party-ID: "4151502" <<a href="mailto:sip%3A4151502@10.11.12.13" target="_blank">sip:4151502@10.11.12.13</a>>;party=calling;screen=yes;privacy=off</div><div><br></div><div>v=0</div><div>o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13</div><div>s=FreeSWITCH</div><div>c=IN IP4 10.11.12.13</div><div>t=0 0</div><div>m=audio 23230 RTP/AVP 8 101</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20 </div></div><div><br></div><div><b>First INVITE sent to FreeSWITCH by Asterisk (forked call):</b></div><div><div><div>INVITE <a href="mailto:sip%3A7712552@10.11.12.13" target="_blank">sip:7712552@10.11.12.13</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport</div><div>Max-Forwards: 70</div><div>From: "4151502" <<a href="mailto:sip%3A4151502@172.26.0.62" target="_blank">sip:4151502@172.26.0.62</a>>;tag=as24a51ba6</div><div>To: <<a href="mailto:sip%3A7712552@10.11.12.13" target="_blank">sip:7712552@10.11.12.13</a>></div><div>Contact: <<a href="http://sip:4151502@172.26.0.62:5060" target="_blank">sip:4151502@172.26.0.62:5060</a>></div><div>Call-ID: <a href="http://135674a534fad0fd5bfff55c2fdc3280@172.26.0.62:5060" target="_blank">135674a534fad0fd5bfff55c2fdc3280@172.26.0.62:5060</a></div><div>CSeq: 102 INVITE</div><div>User-Agent: Asterisk PBX 1.8.15-cert2</div><div>Date: Wed, 18 Mar 2015 17:47:11 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>Supported: replaces, timer</div><div>Diversion: <<a href="mailto:sip%3A5344446@172.26.0.62" target="_blank">sip:5344446@172.26.0.62</a>></div><div>Content-Type: application/sdp</div><div>Content-Length: 312</div><div><br></div><div>v=0</div><div>o=root 693576967 693576967 IN IP4 172.26.0.62</div><div>s=Asterisk PBX 1.8.15-cert2</div><div>c=IN IP4 172.26.0.62</div><div>t=0 0</div><div>m=audio 30440 RTP/AVP 8 0 9 101</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:9 G722/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div></div><div style="font-weight:bold"><br></div><div style="font-weight:bold">Second INVITE sent to FreeSWITCH by Asterisk (forked call):</div><div><div>INVITE <a href="mailto:sip%3A6595454@10.11.12.13" target="_blank">sip:6595454@10.11.12.13</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport</div><div>Max-Forwards: 70</div><div>From: "4151502" <<a href="mailto:sip%3A4151502@172.26.0.62" target="_blank">sip:4151502@172.26.0.62</a>>;tag=as22f810b0</div><div>To: <<a href="mailto:sip%3A6595454@10.11.12.13" target="_blank">sip:6595454@10.11.12.13</a>></div><div>Contact: <<a href="http://sip:4151502@172.26.0.62:5060" target="_blank">sip:4151502@172.26.0.62:5060</a>></div><div>Call-ID: <a href="http://6979c3dd69c5f8e557131e485466ad57@172.26.0.62:5060" target="_blank">6979c3dd69c5f8e557131e485466ad57@172.26.0.62:5060</a></div><div>CSeq: 102 INVITE</div><div>User-Agent: Asterisk PBX 1.8.15-cert2</div><div>Date: Wed, 18 Mar 2015 17:47:11 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>Supported: replaces, timer</div><div>Diversion: <<a href="mailto:sip%3A5344446@172.26.0.62" target="_blank">sip:5344446@172.26.0.62</a>></div><div>Content-Type: application/sdp</div><div>Content-Length: 310</div><div><br></div><div>v=0</div><div>o=root 89056081 89056081 IN IP4 172.26.0.62</div><div>s=Asterisk PBX 1.8.15-cert2</div><div>c=IN IP4 172.26.0.62</div><div>t=0 0</div><div>m=audio 30708 RTP/AVP 8 0 9 101</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:9 G722/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20</div><div>a=sendrecv</div><div style="font-weight:bold"><br></div></div><div><b>First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):</b></div></div><div><div>INVITE <a href="mailto:sip%3A7712552@172.25.200.101" target="_blank">sip:7712552@172.25.200.101</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H</div><div>Max-Forwards: 69</div><div>From: "4151502" <<a href="mailto:sip%3A4151502@172.25.200.111" target="_blank">sip:4151502@172.25.200.111</a>>;tag=Z6pSHe2eXSB2p</div><div>To: <<a href="mailto:sip%3A7712552@172.25.200.101" target="_blank">sip:7712552@172.25.200.101</a>></div><div>Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90</div><div>CSeq: 73014353 INVITE</div><div>Contact: <<a href="http://sip:mod_sofia@172.25.200.111:5080" target="_blank">sip:mod_sofia@172.25.200.111:5080</a>></div><div>User-Agent: FreeSWITCH-mod_sofia/1.2.12</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY</div><div>Supported: timer, precondition, path, replaces</div><div>Allow-Events: talk, hold, conference, refer</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session</div><div>Content-Length: 209</div><div>Diversion: <<a href="mailto:sip%3A5344446@172.25.200.101" target="_blank">sip:5344446@172.25.200.101</a>></div><div>X-FS-Support: update_display,send_info</div><div><br></div><div>v=0</div><div>o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111</div><div>s=FreeSWITCH</div><div>c=IN IP4 172.25.200.111</div><div>t=0 0</div><div>m=audio 19804 RTP/AVP 8 0 9 101 13</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div style="font-weight:bold"><br></div></div><div><b>Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):</b></div><div><div>INVITE <a href="mailto:sip%3A6595454@172.25.200.101" target="_blank">sip:6595454@172.25.200.101</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD</div><div>Max-Forwards: 69</div><div>From: "4151502" <<a href="mailto:sip%3A4151502@172.25.200.111" target="_blank">sip:4151502@172.25.200.111</a>>;tag=0FgjK9jjt21mj</div><div>To: <<a href="mailto:sip%3A6595454@172.25.200.101" target="_blank">sip:6595454@172.25.200.101</a>></div><div>Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90</div><div>CSeq: 73014353 INVITE</div><div>Contact: <<a href="http://sip:mod_sofia@172.25.200.111:5080" target="_blank">sip:mod_sofia@172.25.200.111:5080</a>></div><div>User-Agent: FreeSWITCH-mod_sofia/1.2.12</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY</div><div>Supported: timer, precondition, path, replaces</div><div>Allow-Events: talk, hold, conference, refer</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session</div><div>Content-Length: 209</div><div>Diversion: <<a href="mailto:sip%3A5344446@172.25.200.101" target="_blank">sip:5344446@172.25.200.101</a>></div><div>X-FS-Support: update_display,send_info</div><div><br></div><div>v=0</div><div>o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111</div><div>s=FreeSWITCH</div><div>c=IN IP4 172.25.200.111</div><div>t=0 0</div><div>m=audio 31376 RTP/AVP 8 0 9 101 13</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div style="font-weight:bold"><br></div></div></div></div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
<p><font size="1" face="courier new, monospace"><img src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br></font></p><p><font face="courier new, monospace"><b><i>Twitter: @FreeSWITCH , @briankwest</i></b><br><a href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br><a href="http://www.freeswitchcookbook.com" target="_blank">http://www.freeswitchcookbook.com</a></font></p>
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