It would appear you're calling yourself. /b On Tue, Sep 10, 2019 at 3:27 AM Dominique Jeannerod < dominique.jeannerod@interactiv-group.com> wrote:
Hi all,
currently working to understand some calls without RTP (one-way audio, or no audio at all), I noticed a strange case in our traces : we send an outgoing call on a trunk. The destination number is translated to another number, and the call comes back to our SBC (the new dest is on our network). What i noticed is that the incoming call has the same call-id as our outgoing call (I suppose that the other operator uses proxies rather than SBCs), and it comes to the *same *freeswitch (has 2 IP addresses)
So, wondering what could be the impact of that on our Freeswitch SBC. Can those 2 calls be managed as one and only call by freeswitch ?
Thanks with anticipation Best regards,
Dominique Jeannerod
*Calls traces : * *First call (outgoing) :* Our SBC IP1 -> External IP 1 INVITE sip:+<Dest number 1>@<External IP 1> SIP/2.0 Via: SIP/2.0/UDP <Our SBC IP 1>;rport;branch=z9hG4bKg6y1BXQ5ZvreF Max-Forwards: 68 From: "Source num" <sip:+<Source Num>@< Our SBC IP 1>;tag=r8p71798metHN To: <sip:+ <Dest number 1>@<External IP 1>
*Call-ID: d3b11cd5-4a87-1238-068f-5254000db631*CSeq: 9292108 INVITE Contact: <sip:gw+gwname@ <Our SBC IP 1>:5060;transport=udp;gw=gwname> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 14400;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 248 X-FS-Support: update_display,send_info P-Asserted-Identity: "<PAI Display>" <sip:+<PAI Number>@ <Our SBC IP 1> > v=0 o=FreeSWITCH 1567668560 1567668561 IN IP4 <Our SBC IP 1> s=FreeSWITCH c=IN IP4 <Our SBC IP 1> t=0 0 m=audio 23240 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
SIP/2.0 200 OK Via: SIP/2.0/UDP <Our SBC IP 1> ;received= <Our SBC IP 1> ;rport=5060;branch=z9hG4bKg6y1BXQ5ZvreF From: "Source num" <sip:+<Source Num>@< Our SBC IP 1>>;tag=r8p71798metHN To: <sip:+3 Dest number 1>@<External IP 1>;tag=t25KQD4evZ2DF Call-ID: d3b11cd5-4a87-1238-068f-5254000db631 CSeq: 9292108 INVITE Contact: <sip:+<Dest number 1>@ <External IP 1>;did=4a5.72bd7e82> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip: <Dest number 1>@<domain> X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1567660096 1567660097 IN IP4 <Our SBC IP 2> s=FreeSWITCH c=IN IP4 <External RTP IP 1> t=0 0 m=audio 58152 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
*Second call (Incoming) :* External IP2 -> Our SBC IP 2 INVITE sip:+<Dest number 2>@<Our SBC IP 2>:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP <External IP 2>:5060;branch=z9hG4bK446.1e47ee86.0 Max-Forwards: 62 From: "Source num" <sip:+<Source Num>@<domain>;user=phone>;tag=r8p71798metHN To: <sip:+<Dest number 2>@<domain>;user=phone>
*Call-ID: d3b11cd5-4a87-1238-068f-5254000db631*CSeq: 9292108 INVITE Contact: <sip:gw+<gwname>@<External IP 2>;did=4a5.e4c51b72> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 14400;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Support: update_display,send_info P-Asserted-Identity: "<PAI Display>" <sip:+<PAI Number>@<domain>;user=phone> v=0 o=FreeSWITCH 1567668560 1567668561 IN IP4 <Our SBC IP 1> s=FreeSWITCH c=IN IP4 <External RTP IP 2> t=0 0 m=audio 56204 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
SIP/2.0 200 OK Via: SIP/2.0/UDP <External IP 2>:5060;branch=z9hG4bK446.1e47ee86.0 From: "Source num" <sip:+<Source Num>@<domain>;user=phone>;tag=r8p71798metHN To: <sip:+<Dest number 2>@<domain>;user=phone>;tag=t25KQD4evZ2DF Call-ID: d3b11cd5-4a87-1238-068f-5254000db631 CSeq: 9292108 INVITE Contact: <sip:+<Dest num 2>@>Our SBC IP 2>:5060;transport=udp> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 14400;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 248 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:<Dest number 2>@keyyo.net X-FS-Support: update_display,send_info P-Asserted-Identity: "Outbound Call" <sip:<Dest number 2>@<domain>> v=0 o=FreeSWITCH 1567660096 1567660097 IN IP4 <Our SBC IP 2> s=FreeSWITCH c=IN IP4 <Our SBC IP 2> t=0 0 m=audio 31704 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
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