Thanks at all to answer.. but I cannot get it going.. In the INVITE always I have two record-route:
Record-Route: <sip:FQDN_DNS:5061;transport=tls;lr> Record-Route: <sip:FQDN_IP:5060;lr> And in the 200 I have three: Record-Route: <sip:FQDN_DNS:5061;transport=tls;lr> Record-Route: <sip:FQDN_IP:5060;lr> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> Could be it the problem? I'm going crazy, and I donnot know how fix it.. Any ideas? Thanks!! El vie., 11 sept. 2020 a las 14:32, egemen ulus (<[email protected]>) escribió: > Hi, > > I am not sure what you exactly try to achieve, but for the record-route > parameter, I can provide two options for you. > > If you are not satisfied with the second record-route, you might remove > (remove_hf();) "Record-Route" header before adding a new one via > 'record_route_preset'. > But I think it is like a workaround solution, for better way you can check > whether you used "record_route();" or not before/after using > ''record_route_preset'' > > Regards > Egemen U. > ------------------------------ > *Gönderen:* sip user <[email protected]> adına sr-users < > [email protected]> > *Gönderildi:* 11 Eylül 2020 Cuma 14:25 > *Kime:* Kamailio (SER) - Users Mailing List <[email protected]> > *Konu:* Re: [SR-Users] Kamailio drop calls with Teams > > Any idea? Can i change that second récord router? > > Thanks > > El lun., 7 sept. 2020 8:42, sip user <[email protected]> escribió: > > Hi.... I've tried to add record_route_preset( > "yourdomain.com:5061;transport=tls", > "your_ip:5060" ) in incoming calls, call, from Teams to Asterisk, and with > sipdump I see that: > > INVITE: > > tag: snd > pid: 15506 > process: 10 > time: 1599460531.198988 > date: Mon Sep 7 06:35:31 2020 > proto: udp ipv4 > srcip: FQDN IP > srcport: 5060 > dstip: IP ASTERISK > dstport: 18060 > ~~~~~~~~~~~~~~~~~~~~ > INVITE sip:s@IP ASTERISK:18060 SIP/2.0 > Record-Route: <sip:FQDN DNS:5061;transport=tls;lr> > Record-Route: <sip:FQDN IP:5060;lr> > FROM: AdminTeams<sip:[email protected]:5061 > ;user=phone>;tag=295acf4c5acf4a3c8ae8f64dce4a9a05 > TO: <sip:+34590@FQDN DNS:5061;user=phone> > CSEQ: 1 INVITE > CALL-ID: 901952e5fbc15d8ca107fd3c6e8f2edc > MAX-FORWARDS: 69 > Via: SIP/2.0/UDP FQDN > IP;branch=z9hG4bK4108.41910a703c892d309f8aaa6eee303e1c.0;i=1 > VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKab9565a8 > RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061 > ;transport=tls;lr> > CONTACT: <sip:api-du-a-usea.pstnhub.microsoft.com:443 > ;x-i=8ce77537-20fd-43b0-9a49-7c5b7fb7e198;x-c=901952e5fbc15d8ca107fd3c6e8f2edc/d/8/31abc1996a874f5a8133d653d07239f4> > CONTENT-LENGTH: 1102 > MIN-SE: 300 > SUPPORTED: timer > USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.9.1.3 i.EUWE.0 > CONTENT-TYPE: application/sdp > ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY > P-ASSERTED-IDENTITY: <tel:+1099>,<sip:mail> > PRIVACY: id > SESSION-EXPIRES: 3600 > > 200 OK > > tag: rcv > pid: 15498 > process: 2 > time: 1599460531.207751 > date: Mon Sep 7 06:35:31 2020 > proto: udp ipv4 > srcip: IP ASTERISK > srcport: 18060 > dstip: FQDN IP > dstport: 5060 > ~~~~~~~~~~~~~~~~~~~~ > SIP/2.0 200 OK > Via: SIP/2.0/UDP > FQDNIP;branch=z9hG4bK4108.41910a703c892d309f8aaa6eee303e1c.0;i=1;received=92.222.217.64 > Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKab9565a8 > Record-Route: <sip:FQDN DNS:5061;transport=tls;lr> > Record-Route: <sip:FQDN IP:5060;lr> > Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061 > ;transport=tls;lr> > From: AdminTeams<sip:[email protected]:5061 > ;user=phone>;tag=295acf4c5acf4a3c8ae8f64dce4a9a05 > To: <sip:+34590@FQDN DNS:5061;user=phone>;tag=as5e107437 > Call-ID: 901952e5fbc15d8ca107fd3c6e8f2edc > CSeq: 1 INVITE > Server: Asterisk PBX 11.25.3 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Session-Expires: 1800;refresher=uas > Contact: <sip:s@IP ASTERISK:18060> > Content-Type: application/sdp > Require: timer > Content-Length: 345 > > I rewrite the first record-route in both, INVITE and 200 OK, but the > second record-route, is the FQDN IP again.. > Could be it the problem? > > How can I rewrite that record-route? > > Thanks > > El jue., 3 sept. 2020 a las 13:53, Pepelux (<[email protected]>) > escribió: > > I don't know. Try to write the domain directly and not an alias: > > record_route_preset("yourdomain.com:5061;transport=tls", "your_ip:5060"); > > On Thu, 3 Sep 2020 at 13:38, sip user <[email protected]> wrote: > > Yes, this is I do: > > record_route(); > xlog("L_INFO", "***********ROUTE PSTN***********"); > $rU="1005"; > > Have I do any more? Why mu record-route is different yours? > > Thanks > > El jue., 3 sept. 2020 a las 13:27, Pepelux (<[email protected]>) > escribió: > > You have to use record_route_preset when the message is sent from Kamailio > to Teams > > if (from_uri =~ ".*microsoft.com") { > record_route(); > } else { > record_route_preset("SBC-DNS-DOMAIN:5061;transport=tls", > "SBC-IP-ADDR:5060"); > } > > On Thu, 3 Sep 2020 at 13:13, sip user <[email protected]> wrote: > > Thanks Pepelux.. > > Yes, I follow that post to configure it. But I don´t know where could be > the problem and change Record-Route, because, in the post say, only I have > to change it when I call from kamailio to Teams, so outgoing calls, right? > With record-route-preset... I'm wrong? > > Thanks > > El jue., 3 sept. 2020 a las 13:07, Pepelux (<[email protected]>) > escribió: > > 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-td181493.html > > 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:[email protected]: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:[email protected]: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 <[email protected]> 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 (<[email protected]>) > 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 <[email protected]> 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 (<[email protected]>) > escribió: > > Hi > > Have you loaded the module? > > loadmodule "sipdump.so" > > On Tue, 1 Sep 2020 at 13:56, sip user <[email protected]> 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 (<[email protected]>) > 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 <[email protected]> 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 <[email protected]> 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 (< > [email protected]>) 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 (< > [email protected]>) 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:[email protected]: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:[email protected]: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:[email protected]: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 > [email protected]https://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 > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >
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