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:
Hi
Have 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 0
Regards
On Tue, 1 Sep 2020 at 13:37, Pepelux pepeluxx@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@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 <+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 Listsr-users@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
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