[SR-Users] Kamailio drop calls with Teams

sip user sipuser404 at gmail.com
Thu Sep 3 13:35:08 CEST 2020


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
>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>
>>>> _______________________________________________
>>>> 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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