[SR-Users] Ynt: Kamailio drop calls with Teams
sip user
sipuser404 at gmail.com
Tue Sep 22 16:37:20 CEST 2020
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 (<ulus_egemen at hotmail.com>)
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 <sipuser404 at gmail.com> adına sr-users <
> sr-users-bounces at lists.kamailio.org>
> *Gönderildi:* 11 Eylül 2020 Cuma 14:25
> *Kime:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *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 <sipuser404 at gmail.com> 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 at 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:+1099 at sip.pstnhub.microsoft.com:5061
> ;user=phone>;tag=295acf4c5acf4a3c8ae8f64dce4a9a05
> TO: <sip:+34590 at 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:+1099 at sip.pstnhub.microsoft.com:5061
> ;user=phone>;tag=295acf4c5acf4a3c8ae8f64dce4a9a05
> To: <sip:+34590 at 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 at 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 (<pepeluxx at gmail.com>)
> 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 <sipuser404 at gmail.com> 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 (<pepeluxx at gmail.com>)
> 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 <sipuser404 at gmail.com> 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 (<pepeluxx at gmail.com>)
> 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:+34XXXXXXXXX at sip.pstnhub.microsoft.com:5061
> ;user=phone>;tag=3a6ca98c0a9a46c98ad781c82f389c4d
> To: <sip:+34YYYYYYYYY at 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 at 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 at SBC-IP-ADDR:5080 SIP/2.0
> FROM: Pepelux <sip:+34XXXXXXXXX at sip.pstnhub.microsoft.com:5061
> ;user=phone>;tag=3a6ca98c0a9a46c98ad781c82f389c4d
> TO: <sip:+34YYYYYYYYY at 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 at 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 at 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 at 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 at gmail.com>)
> escribió:
>
> Hi
>
> Have you loaded the module?
>
> loadmodule "sipdump.so"
>
> On Tue, 1 Sep 2020 at 13:56, sip user <sipuser404 at 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 at 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 at 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 at 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 at 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 at 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 at FQND:5061;user=phone]
> kamailio[1096]: 9(1109) DEBUG: <core> [core/parser/msg_parser.c:174]:
> get_hdr_field(): to body [<sip:+34590 at 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 at 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 at 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 at sip.pstnhub.microsoft.com:5061;user=phone>
> ;tag=c17bb1eb7f8649d4a89d8d4a876ac32b.
> TO: <sip:+34560 at 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 at CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940>.
> To: <sip:+34560 at FQND:5061;user=phone>;tag=de4e6b45.
> From: "Javier Gonz..lez Mu..oz"
> <sip:+324 at 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 at CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940>.
> To: <sip:+34560 at FQND:5061;user=phone>;tag=de4e6b45.
> From: "Javier Gonz..lez Mu..oz"
> <sip:+324 at 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 at 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 at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
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> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
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> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
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> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
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> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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> _______________________________________________
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> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
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> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> _______________________________________________
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> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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