[SR-Users] Kamailio drop calls with Teams
sip user
sipuser404 at gmail.com
Thu Sep 3 13:10:11 CEST 2020
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
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>> 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
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