It looks good but in the capture file I saw FQNDIP in RR and not FQNDDNS
This post by Henning may help you: https://skalatan.de/en/blog/kamailio-sbc-teams
And also you can read that: http://sip-router.1086192.n5.nabble.com/Kamailio-as-SBC-for-Microsoft-Teams-...
This is a response from my Kamailio to Teams. Maybe it can be useful for you:
tag: snd pid: 1394 process: 1 time: 1599126436.582012 date: Thu Sep 3 11:47:16 2020 proto: tls ipv4 srcip: SBC-IP-ADDR srcport: 5061 dstip: 52.114.75.24 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK8ac0b4eb Record-Route: sip:SBC-DNS-DOMAIN:5060;r2=on;lr Record-Route: sip:SBC-DNS-DOMAIN:5061;transport=tls;r2=on;lr Record-Route: sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr From: Pepelux sip:+34XXXXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone;tag=3a6ca98c0a9a46c98ad781c82f389c4d To: sip:+34YYYYYYYYY@SBC-DNS-DOMAIN:5061;user=phone;tag=as524dd8d6 Call-ID: 99ac64b5ad3455be9fd8c838cbdd4c6c CSeq: 1 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1 Allow:sINVITE, ACK, CANCEL, OPTIONS, BYE, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: sip:+34YYYYYYYYY@SBC-IP-ADDR:5080 Content-Type: application/sdp Content-Length: 532
v=0 o=root 11212956 11212956 IN IP4 SBC-IP-ADDR s=Asterisk PBX 16.2.1~dfsg-1+deb10u1 c=IN IP4 SBC-IP-ADDR t=0 0 m=audio 30444 RTP/SAVP 8 a=maxptime:150 a=mid:1 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtcp:30445 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:13aivX1dzg2Pbon6bNcvyzNenxMbtvCEh65mnL0t a=ptime:20 a=ice-ufrag:oysP7oty a=ice-pwd:Vsv8LeF21onN8NFIKhOvpF53zL a=candidate:MLkUNexgYqowLxtY 1 UDP 2130706431 SBC-IP-ADDR 30444 typ host a=candidate:MLkUNexgYqowLxtY 2 UDP 2130706430 SBC-IP-ADDR 30445 typ host ~~~~~~~~~~~~~~~~~~~~ tag: rcv pid: 1412 process: 19 time: 1599126436.612972 date: Thu Sep 3 11:47:16 2020 proto: tls ipv4 srcip: 52.114.75.24 srcport: 6209 dstip: SBC-IP-ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ ACK sip:+34YYYYYYYYY@SBC-IP-ADDR:5080 SIP/2.0 FROM: Pepelux sip:+34XXXXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone;tag=3a6ca98c0a9a46c98ad781c82f389c4d TO: sip:+34YYYYYYYYY@SBC-DNS-DOMAIN:5061;user=phone;tag=as524dd8d6 CSEQ: 1 ACK CALL-ID: 99ac64b5ad3455be9fd8c838cbdd4c6c MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK693bb042 ROUTE: sip:SBC-DNS-DOMAIN:5061;transport=tls;r2=on;lr,sip:SBC-DNS-DOMAIN:5060;r2=on;lr CONTACT: sip:api-du-c-euwe.pstnhub.microsoft.com:443 ;x-i=21167d9c-8cf9-4253-b059-2d987fadae5a;x-c=99ac64b5ad3455be9fd8c838cbdd4c6c/d/8/90df5d0ec92048de868e268fa57ac0f1 CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.9.1.3 i.EUWE.7 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
Regards
On Thu, 3 Sep 2020 at 12:34, sip user sipuser404@gmail.com wrote:
Hi Pepelux,
I have this one:
remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { if($src_ip != "IP ASTERISK"){ record_route(); xlog("L_INFO", "***********ROUTE PSTN***********"); $rU="1005"; } else { xlog("L_INFO","LLamada desde $si con puerto $sp"); record_route_preset("FQNDDNS:5061;transport=tls", "FQNDIP:5060"); add_rr_param(";r2=on"); route(DISPATCH); route(RELAY); } }
When the call is from Teams (src_ip != "IP ASTERISK"), incoming calls, I send the call to 1005 extension. Is here where I have to make the change? Or where?
Thanks
El jue., 3 sept. 2020 a las 12:14, Pepelux (pepeluxx@gmail.com) escribió:
Hi
Kamailio doesn't receive any ACK from Teams. I think the problem is the '200 Ok' that you send to Teams is not what he expected. Maybe this is wrong: Record-Route: sip:FQNDIP;r2=on;lr Record-Route: sip:FQNDIP:5061;transport=tls;r2=on;lr
Try to put the registered domain (FQNDDNS) and not de IP address
Regards
On Thu, 3 Sep 2020 at 10:56, sip user sipuser404@gmail.com wrote:
Sorry.. Yes, I need to load sipdump.so module..
I attach the result..
Thanks
El mar., 1 sept. 2020 a las 14:03, Pepelux (pepeluxx@gmail.com) escribió:
Hi
Have you loaded the module?
loadmodule "sipdump.so"
On Tue, 1 Sep 2020 at 13:56, sip user sipuser404@gmail.com wrote:
Hi pepelux.. When I set:
modparam("sipdump", "enable", 1)
Error, Kamailio not start, error bad config..
Thanks
El mar., 1 sept. 2020 a las 13:45, Pepelux (pepeluxx@gmail.com) escribió:
Sorry, I've sent last mail without finishing :)
https://kamailio.org/docs/modules/5.5.x/modules/sipdump.html
You only have to load the module and set:
modparam("sipdump", "enable", 1)
Also you can enable or disable using RPC commands:
kamcmd sipdump.enable kamcmd sipdump.enable 1 kamcmd sipdump.enable 0
Regards
On Tue, 1 Sep 2020 at 13:37, Pepelux pepeluxx@gmail.com wrote:
> Hi > > https://kamailio.org/docs/modules/5.5.x/modules/sipdump.html > > You only have to load the module and set: > > modparam("sipdump", "enable", 1) > > kamcmd sipdump.enable 1 > kamcmd sipdump.enable 0 > > modparam("sipdump", "enable", 1) > > > On Tue, 1 Sep 2020 at 13:23, sip user sipuser404@gmail.com wrote: > >> Hi Daniel.. >> >> And how load sipdump? >> I'm using kamailio 5.2.1-1 and I think sipdump module is not >> available, right? >> >> Thanks >> >> El mar., 1 sept. 2020 a las 12:27, Daniel-Constantin Mierla (< >> miconda@gmail.com>) escribió: >> >>> Hello, >>> >>> it seems that the ACK comes in, but my guess is that the R-URI is >>> not properly set. From the logs it looks like same value as for To header >>> URI, while it should be the address in Contact header of 200ok for INVITE. >>> >>> Load the sipdump module and that will save all the sip traffic in >>> a text file, making it easier to see what comes/goes on both directions, no >>> matter is over tls or not. If you use kamailio devel version (master >>> branch), then sipdump module can also store traffic in pcap file (tls >>> traffic saved as udp for simplicity, but it is easy to spot from headers or >>> meta data extra header). >>> >>> You can send the sipdump file here for investigation, so we can >>> see if some headers or r-uri are not correct. >>> >>> Cheers, >>> Daniel >>> On 01.09.20 11:15, sip user wrote: >>> >>> Hi Daniel, thanks for answered to me... >>> >>> With debug=3 I see that: >>> >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:610]: parse_msg(): SIP Request: >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:612]: parse_msg(): method: <ACK> >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:614]: parse_msg(): uri: >>> sip:+34590@FQND:5061;user=phone;transport=tls >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: >>> tag=92e2fd8688a9d17b927d9be2f84faa55-8079 >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header >>> reached, state=29 >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:171]: get_hdr_field(): <TO> [94]; uri=[ >>> sip:+34590@FQND:5061;user=phone] >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:174]: get_hdr_field(): to body [ >>> sip:+34590@FQND:5061;user=phone], to tag >>> [92e2fd8688a9d17b927d9be2f84faa55-8079] >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSEQ>: <1> <ACK> >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, >>> <branch> = <z9hG4bKf4784e39>; state=16 >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via >>> kamailio[1096]: 9(1109) DEBUG: <core> [core/receive.c:240]: >>> receive_msg(): --- received sip message - request - call-id: >>> [d3649f52dc0057768ec6c18733de8206] - cseq: [1 ACK] >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0 >>> kamailio[1096]: 9(1109) DEBUG: <core> >>> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header >>> kamailio[1096]: 9(1109) DEBUG: {1 1 ACK >>> d3649f52dc0057768ec6c18733de8206} <core> [core/receive.c:295]: >>> receive_msg(): preparing to run routing scripts... >>> kamailio[1096]: 9(1109) DEBUG: {1 1 ACK >>> d3649f52dc0057768ec6c18733de8206} sl [sl_funcs.c:397]: sl_filter_ACK(): too >>> late to be a local ACK! >>> >>> So, I understand that ACK comes from Teams, right? So kamailio >>> routing problem? >>> >>> Thanks >>> >>> El mar., 25 ago. 2020 a las 15:32, Daniel-Constantin Mierla (< >>> miconda@gmail.com>) escribió: >>> >>>> Hello, >>>> >>>> run with debug=3 in kamailio.cfg and see if the ACK comes to >>>> Kamailio, if yes, then some routing issue in kamailio.cfg. If does not >>>> come, you will have to check the headers to see if MS Teams expects >>>> something else there, typically is about Record-Route domains... >>>> >>>> Cheers, >>>> Daniel >>>> On 20.08.20 12:25, sip user wrote: >>>> >>>> Hi, I'm connecting Teams with kamailio server. From Kamailio to >>>> teams I have no problems, but from teams to Kamailio yes. Drop the call.. >>>> >>>> With ngrep I see that: >>>> >>>> INVITE >>>> sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940 >>>> SIP/2.0. >>>> Record-Route: sip:FQND_IP;r2=on;lr. >>>> Record-Route: sip:FQND_IP:5061;transport=tls;r2=on;lr. >>>> FROM: "Javier Gonz..lez Mu..oz" >>>> sip:+324@sip.pstnhub.microsoft.com:5061;user=phone >>>> ;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. >>>> TO: sip:+34560@FQND:5061;user=phone. >>>> CSEQ: 1 INVITE. >>>> CALL-ID: c1364913e582553a9a9c2544c3583b0a. >>>> MAX-FORWARDS: 69. >>>> Via: SIP/2.0/UDP >>>> 92.222.217.64;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. >>>> VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. >>>> RECORD-ROUTE: >>>> sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr. >>>> CONTACT: >>>> sip:api-du-a-euno.pstnhub.microsoft.com:443;x-i=b0b53fc5-76ef-4619-9a68-13e0a4eea92d;x-c=c1364913e582553a9a9c2544c3583b0a/d/8/6c25eb3789a14bf188ce6b05b5e27891 >>>> . >>>> CONTENT-LENGTH: 1091. >>>> MIN-SE: 300. >>>> SUPPORTED: timer. >>>> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.7.31.1 i.EUNO.0. >>>> CONTENT-TYPE: application/sdp. >>>> ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY. >>>> P-ASSERTED-IDENTITY: tel:+324 <+324>,sip:EMAIL. >>>> PRIVACY: id. >>>> SESSION-EXPIRES: 3600. >>>> . >>>> v=0. >>>> o=- 165103 0 IN IP4 127.0.0.1. >>>> s=session. >>>> c=IN IP4 52.113.44.8. >>>> b=CT:10000000. >>>> t=0 0. >>>> m=audio 50452 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. >>>> c=IN IP4 52.113.44.8. >>>> a=rtcp:50453. >>>> a=ice-ufrag:FZTb. >>>> a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. >>>> a=rtcp-mux. >>>> a=candidate:1 1 UDP 2130706431 52.113.44.8 50452 typ srflx raddr >>>> 10.0.33.240 rport 50 >>>> >>>> U CLIENT_IP:55766 -> FQND_IP:5060 #2 >>>> SIP/2.0 180 Ringing. >>>> Via: SIP/2.0/UDP >>>> FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. >>>> Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. >>>> Record-Route: sip:FQND_IP;lr;r2=on. >>>> Record-Route: sip:FQND_IP:5061;transport=tls;r2=on;lr. >>>> Record-Route: >>>> sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr. >>>> Contact: >>>> sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940 >>>> . >>>> To: sip:+34560@FQND:5061;user=phone;tag=de4e6b45. >>>> From: "Javier Gonz..lez Mu..oz" >>>> sip:+324@sip.pstnhub.microsoft.com:5061;user=phone >>>> ;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. >>>> Call-ID: c1364913e582553a9a9c2544c3583b0a. >>>> CSeq: 1 INVITE. >>>> User-Agent: 3CXPhone 6.0.26523.0. >>>> Content-Length: 0. >>>> >>>> U CLIENT_IP:55766 -> FQND_IP:5060 #3 >>>> SIP/2.0 200 OK. >>>> Via: SIP/2.0/UDP >>>> FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. >>>> Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. >>>> Record-Route: sip:FQND_IP;lr;r2=on. >>>> Record-Route: sip:FQND_IP:5061;transport=tls;r2=on;lr. >>>> Record-Route: >>>> sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr. >>>> Contact: >>>> sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940 >>>> . >>>> To: sip:+34560@FQND:5061;user=phone;tag=de4e6b45. >>>> From: "Javier Gonz..lez Mu..oz" >>>> sip:+324@sip.pstnhub.microsoft.com:5061;user=phone >>>> ;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. >>>> Call-ID: c1364913e582553a9a9c2544c3583b0a. >>>> CSeq: 1 INVITE. >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, >>>> NOTIFY, REFER, INFO, MESSAGE. >>>> Content-Type: application/sdp. >>>> Supported: replaces. >>>> User-Agent: 3CXPhone 6.0.26523.0. >>>> Content-Length: 1067. >>>> . >>>> v=0. >>>> o=3cxVCE 324945090 117647850 IN IP4 . >>>> s=3cxVCE Audio Call. >>>> t=0 0. >>>> m=audio 0 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. >>>> c=IN IP4 52.113.44.8. >>>> a=rtpmap:104 SILK/16000. >>>> a=rtpmap:9 G722/8000. >>>> a=rtpmap:103 SILK/8000. >>>> a=rtpmap:111 SIREN/16000. >>>> a=fmtp:111 bitrate=16000. >>>> a=rtpmap:18 G729/8000. >>>> a=fmtp:18 annexb=no. >>>> a=rtpmap:0 PCMU/8000. >>>> a=rtpmap:8 PCMA/8000. >>>> a=rtpmap:97 RED/8000. >>>> a=rtpmap:101 telephone-event/8000. >>>> a=fmtp:101 0-16. >>>> a=rtpmap:13 CN/8000. >>>> a=rtpmap:118 CN/16000. >>>> a=rtcp:50453. >>>> a=ice-ufrag:FZTb. >>>> a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. >>>> a=rtcp-mux. >>>> a=candidate:1 1 UDP 213 >>>> >>>> I never received ACK.. >>>> >>>> In my configuration: >>>> >>>> Kamailio.cfg: >>>> >>>> #!KAMAILIO >>>> #!define WITH_TLS >>>> >>>> event_route[tm:local-request] { >>>> >>>> if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") >>>> { >>>> append_hf("Contact: sip:FQND:5061;transport=tls >>>> \r\n"); >>>> } >>>> xlog("L_INFO", "Sent out tm request: $mb\n"); >>>> } >>>> >>>> request_route{ >>>> >>>> remove_hf("Route"); >>>> if (is_method("INVITE|SUBSCRIBE")) { >>>> xlog("L_INFO","$fU is trying to call to $rU con >>>> valores $tu\n"); >>>> $rU="1005"; >>>> } >>>> } >>>> >>>> What I'm doing wrong? >>>> >>>> I don't understand why not received ACK.. >>>> >>>> Could anyone help me? >>>> >>>> Thanks >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Funding: https://www.paypal.me/dcmierla >>>> >>>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users@lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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