Re: [Freeswitch-users] Two calls with same call-id
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 -- Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>
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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html> _________________________________________________________________________
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Join our online community to chat in real time https://signalwire.community
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Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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What do you mean exactly by “managed”? They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question. So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID. You treat them as different calls altogether. On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
Show me a pcap? On Tue, Sep 10, 2019 at 9:06 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What do you mean exactly by “managed”?
They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question.
So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID.
You treat them as different calls altogether.
On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
The trace is redacted a lot but you can see not just the call-id is the same but also the branch and the tags etc. Also sip call-id is not the same thing as UUID and only the outbound SIP call gets to choose the CALL-ID not the inbound call. On Thu, Sep 12, 2019 at 1:12 PM Brian West <brian@freeswitch.com> wrote:
Show me a pcap?
On Tue, Sep 10, 2019 at 9:06 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What do you mean exactly by “managed”?
They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question.
So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID.
You treat them as different calls altogether.
On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
On Fri, Sep 13, 2019, 00:41 Anthony Minessale <anthony.minessale@gmail.com> wrote:
The trace is redacted a lot but you can see not just the call-id is the same but also the branch and the tags etc. Also sip call-id is not the same thing as UUID and only the outbound SIP call gets to choose the CALL-ID not the inbound call.
On Thu, Sep 12, 2019 at 1:12 PM Brian West <brian@freeswitch.com> wrote:
Show me a pcap?
On Tue, Sep 10, 2019 at 9:06 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What do you mean exactly by “managed”?
They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question.
So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID.
You treat them as different calls altogether.
On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com
https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
Possibly the SIP dialog-transparency is enabled at SBC?? On Thu, Sep 12, 2019 at 1:12 PM Anthony Minessale < anthony.minessale@gmail.com> wrote:
The trace is redacted a lot but you can see not just the call-id is the same but also the branch and the tags etc. Also sip call-id is not the same thing as UUID and only the outbound SIP call gets to choose the CALL-ID not the inbound call.
On Thu, Sep 12, 2019 at 1:12 PM Brian West <brian@freeswitch.com> wrote:
Show me a pcap?
On Tue, Sep 10, 2019 at 9:06 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What do you mean exactly by “managed”?
They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question.
So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID.
You treat them as different calls altogether.
On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com
https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
Thanks, Brian, i think i have resolved my issue (a stupid FW ports issue ...), but still interested to understand if this call flow can lead to issues on Freeswitch (doesn't seem to be the case, as I really see freeswitch managing two independant calls). How can i send you a pcap ? Le jeu. 12 sept. 2019 à 19:27, Brian West <brian@freeswitch.com> a écrit :
Show me a pcap?
On Tue, Sep 10, 2019 at 9:06 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What do you mean exactly by “managed”?
They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question.
So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID.
You treat them as different calls altogether.
On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>
The SIP dialog is not identified by the Call-ID alone. From and To are also part of the tuple. So although they share a Call-ID the From and To headers are different, therefore they are different SIP dialogs (calls). On Fri, 13 Sep 2019 at 11:17, Dominique Jeannerod < dominique.jeannerod@interactiv-group.com> wrote:
Thanks, Brian,
i think i have resolved my issue (a stupid FW ports issue ...), but still interested to understand if this call flow can lead to issues on Freeswitch (doesn't seem to be the case, as I really see freeswitch managing two independant calls). How can i send you a pcap ?
Le jeu. 12 sept. 2019 à 19:27, Brian West <brian@freeswitch.com> a écrit :
Show me a pcap?
On Tue, Sep 10, 2019 at 9:06 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What do you mean exactly by “managed”?
They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question.
So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID.
You treat them as different calls altogether.
On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
Thank you all for your comments and explanations Best regards D. Jeannerod Le lun. 16 sept. 2019 à 18:52, Steven Ayre <steveayre@gmail.com> a écrit :
The SIP dialog is not identified by the Call-ID alone. From and To are also part of the tuple.
So although they share a Call-ID the From and To headers are different, therefore they are different SIP dialogs (calls).
On Fri, 13 Sep 2019 at 11:17, Dominique Jeannerod < dominique.jeannerod@interactiv-group.com> wrote:
Thanks, Brian,
i think i have resolved my issue (a stupid FW ports issue ...), but still interested to understand if this call flow can lead to issues on Freeswitch (doesn't seem to be the case, as I really see freeswitch managing two independant calls). How can i send you a pcap ?
Le jeu. 12 sept. 2019 à 19:27, Brian West <brian@freeswitch.com> a écrit :
Show me a pcap?
On Tue, Sep 10, 2019 at 9:06 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
What do you mean exactly by “managed”?
They are not the same call. Every time a call is created in FS, FS created a unique UUID, and that’s what is used by Fe to manage the call I question.
So when you create an outbound call. It has an associated UUID, and when the same call is sent to you by the proxy, it created another UUID.
You treat them as different calls altogether.
On Tue, 10 Sep 2019 at 10:35, 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
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>
participants (7)
-
Anthony Minessale
-
Brian West
-
David Villasmil
-
Dominique Jeannerod
-
raj singh
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Ramesh Kandasamy
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Steven Ayre