[SR-Users] Kamailio drop calls with Teams
sip user
sipuser404 at gmail.com
Mon Sep 7 08:42:30 CEST 2020
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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