[SR-Users] Kamailio drop calls with Teams

sip user sipuser404 at gmail.com
Fri Sep 11 13:25:56 CEST 2020


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