Hi Joshua, Thanks very much. I presume this is the relevant part: "strictly monotonically increasing and contiguous CSeq sequence numbers (increasing-by-one) in each direction"
In that case I wonder what could be causing the 404 Not Found error. I've attached the relevant SIP packets from the Asterisk log. Can anyone see an issue that would cause the error? Thanks in advance. On Sat, 29 Oct 2022 at 12:03, Joshua C. Colp <[email protected]> wrote: > On Fri, Oct 28, 2022 at 6:28 PM David Cunningham < > [email protected]> wrote: > >> Hello, >> >> We have a problem where Asterisk is resetting the CSeq on a re-INVITE, >> and the phone receiving the re-INVITE is rejecting it, probably as a result >> of that. Would anyone be able to offer any insight please? >> >> The scenario is: >> >> Phone A makes call 1 to Asterisk which dials call 2 to phone B, which >> answers the call. >> >> Phone B puts call 1 on hold, makes call 3 to Asterisk which dials call 4 >> to phone C, which answers the call. >> >> Phone B does an attended REFER transfer of call 2 to call 3, taking >> itself out of the call. Asterisk bridges the remaining calls, so phones A >> and C are now talking to each other. >> >> Asterisk sends a re-INVITE to phone A with a P-Asserted-Identity, to tell >> phone A the updated details of phone C that it's talking to. However phone >> A rejects the re-INVITE with a "404 Not found" error. >> >> The only explanation I can see for the "404 Not found" error is that call >> 1 was set up with "CSeq: 954698786 INVITE", whereas the re-INVITE Asterisk >> sends with the P-Asserted-Identity has "CSeq: 102 INVITE". Why is Asterisk >> resetting the CSeq on the re-INVITE, and doesn't this appear to be >> incorrect? >> > > It's not incorrect. Each direction has its own CSeq[1]. From Phone A to > Asterisk can be 954698786 and from Asterisk to Phone A can be 102. > > [1] https://www.rfc-editor.org/rfc/rfc3261#section-12.2.1.1 > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782
ORIGINAL CALL PHONE A TO ASTERISK: [Oct 27 16:17:43] VERBOSE[2051] chan_sip.c: <--- SIP read from UDP:111.111.52.208:5060 ---> INVITE sip:[email protected]:5060;transport=udp SIP/2.0 Record-Route: <sip:111.111.52.208;lr=on> Via: SIP/2.0/UDP 111.111.52.208;branch=z9hG4bK2d9a.a2b48027e41283fbb0d79a59edd6d1ef.0 Via: SIP/2.0/UDP 222.222.127.193:5060;rport=5060;branch=z9hG4bK4ee82f350717c20f6 Proxy-Authorization: Digest username="111111",realm="example.com",nonce="Y1oVo2NaFHcQ7EwgMuEk3/LGyhpkbC8t",uri="sip:[email protected]:5060;user=phone",response="636c2526a687a35d2683951d0be695d4" Max-Forwards: 69 From: "111111" <sip:[email protected]:5060>;tag=8048438a0b To: <sip:[email protected]:5060;user=phone> Call-ID: 99eab43e2d6c6a20 CSeq: 954698786 INVITE ASTERISK SENDS CALL TO PHONE B: [Oct 27 16:17:43] VERBOSE[1205][C-002558c4] chan_sip.c: Reliably Transmitting (NAT) to 111.111.52.208:5060: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 111.111.52.201:5060;branch=z9hG4bK39357a26;rport Max-Forwards: 70 From: "111111" <sip:[email protected]>;tag=as3569276f To: <sip:[email protected]:5060> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 PHONE B CREATES NEW CALL TO PHONE C: [Oct 27 16:17:51] VERBOSE[2051] chan_sip.c: <--- SIP read from UDP:111.111.52.208:5060 ---> INVITE sip:[email protected]:5060;transport=udp SIP/2.0 Record-Route: <sip:111.111.52.208;lr=on;enat=yes> Via: SIP/2.0/UDP 111.111.52.208;branch=z9hG4bKa1b8.04fe0a3c5e61a15fa889b1d3c087e21d.0 Via: SIP/2.0/UDP 10.10.10.112;rport=60608;received=333.333.96.250;branch=z9hG4bK9bfc44c3c33f64f74 Proxy-Authorization: Digest username="222222",realm="example.com",nonce="Y1oVq2NaFH+v2ge8RzGSe6aI+Y1BIOQ/",uri="sip:[email protected]:5060;user=phone",response="571b389bf09d3bc6fb3d042720435923" Max-Forwards: 69 From: "User User" <sip:[email protected]:5060>;tag=ec8ebeab1f To: <sip:[email protected]:5060;user=phone> Call-ID: 84fb5d56a369394a CSeq: 1053485297 INVITE PHONE B REFERS FIRST CALL TO SECOND: [Oct 27 16:17:58] VERBOSE[2051] chan_sip.c: <--- SIP read from UDP:111.111.52.208:5060 ---> REFER sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 111.111.52.208;branch=z9hG4bK6c56.0b2da5a10fa953dd3d343d71848a6f81.0 Via: SIP/2.0/UDP 10.10.10.112;rport=60608;received=333.333.96.250;branch=z9hG4bK5d81e9ca437b68a91 Max-Forwards: 69 From: <sip:[email protected]:5060>;tag=1841802195 To: "111111" <sip:[email protected]>;tag=as3569276f Call-ID: [email protected]:5060 CSeq: 827299867 REFER Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH Allow-Events: aastra-xml, vdp-session, talk, hold, conference, LocalModeStatus Contact: <sip:[email protected];gr=urn:uuid:00000000-0000-1000-8000-08000FB1971C;alias=333.333.96.250~60608~1> Refer-To: <sip:[email protected]:5060;user=phone?Replaces=84fb5d56a369394a%3bto-tag%3das18a53842%3bfrom-tag%3dec8ebeab1f> Referred-By: <sip:[email protected]> [Oct 27 16:17:58] VERBOSE[2051][C-002558c4] chan_sip.c: <--- Transmitting (NAT) to 111.111.52.208:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 111.111.52.208;branch=z9hG4bK6c56.0b2da5a10fa953dd3d343d71848a6f81.0;received=111.111.52.208;rport=5060 Via: SIP/2.0/UDP 10.10.10.112;rport=60608;received=333.333.96.250;branch=z9hG4bK5d81e9ca437b68a91 From: <sip:[email protected]:5060>;tag=1841802195 To: "111111" <sip:[email protected]>;tag=as3569276f Call-ID: [email protected]:5060 CSeq: 827299867 REFER CALL TO PHONE B HANGS UP AFTER IT HAS TRANSFERRED CALL AWAY: [Oct 27 16:17:58] VERBOSE[2051] chan_sip.c: <--- SIP read from UDP:111.111.52.208:5060 ---> BYE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 111.111.52.208;branch=z9hG4bK4d56.b478685be88bb2085f2c0a828bda5245.0 Via: SIP/2.0/UDP 10.10.10.112;rport=60608;received=333.333.96.250;branch=z9hG4bK2f51e1580e918c15b Max-Forwards: 69 From: <sip:[email protected]:5060>;tag=1841802195 To: "111111" <sip:[email protected]>;tag=as3569276f Call-ID: [email protected]:5060 CSeq: 827299868 BYE ASTERISK SENDS RE-INVITE TO PHONE A WITH NEW REMOTE PARTY DETAILS IN P-Asserted-Identity: [Oct 27 16:17:58] VERBOSE[1194][C-002558c4] chan_sip.c: Reliably Transmitting (NAT) to 111.111.52.208:5060: INVITE sip:[email protected];gr=urn:uuid:00000000-0000-1000-8000-08000FD54867 SIP/2.0 Via: SIP/2.0/UDP 111.111.52.201:5060;branch=z9hG4bK0cc6df1e;rport Route: <sip:111.111.52.208;lr=on> Max-Forwards: 70 From: <sip:[email protected]:5060;user=phone>;tag=as6b77dd09 To: "111111" <sip:[email protected]:5060>;tag=8048438a0b Contact: <sip:[email protected]:5060> Call-ID: 99eab43e2d6c6a20 CSeq: 102 INVITE User-Agent: ConnectTel Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "User User" <sip:[email protected]> Content-Type: application/sdp Content-Length: 254 PHONE A RETURNS AN ERROR: [Oct 27 16:17:58] VERBOSE[2051] chan_sip.c: <--- SIP read from UDP:111.111.52.208:5060 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 111.111.52.201:5060;received=111.111.52.201;branch=z9hG4bK0cc6df1e;rport=5060 From: <sip:[email protected]:5060;user=phone>;tag=as6b77dd09 To: "111111" <sip:[email protected]:5060>;tag=8048438a0b Call-ID: 99eab43e2d6c6a20 CSeq: 102 INVITE Server: Mitel Border GW/4.10.0.169-01 Content-Length: 0
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
