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
