On Thu, 3 Sep 2020 at 12:34, sip user <sipuser404@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@gmail.com>) escribió:HiKamailio 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 addressRegards_______________________________________________On Thu, 3 Sep 2020 at 10:56, sip user <sipuser404@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@gmail.com>) escribió:HiHave you loaded the module?loadmodule "sipdump.so"_______________________________________________On Tue, 1 Sep 2020 at 13:56, sip user <sipuser404@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@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 0RegardsOn Tue, 1 Sep 2020 at 13:37, Pepelux <pepeluxx@gmail.com> wrote:HiYou only have to load the module and set:modparam("sipdump", "enable", 1)kamcmd sipdump.enable 1 kamcmd sipdump.enable 0modparam("sipdump", "enable", 1)On Tue, 1 Sep 2020 at 13:23, sip user <sipuser404@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@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@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@FQND:5061;user=phone]
kamailio[1096]: 9(1109) DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body [<sip:+34590@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@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@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@sip.pstnhub.microsoft.com:5061;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b.
TO: <sip:+34560@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>,<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@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940>.
To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45.
From: "Javier Gonz..lez Mu..oz"<sip:+324@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@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940>.
To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45.
From: "Javier Gonz..lez Mu..oz"<sip:+324@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 List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla
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