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:
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@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:
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 >> >> _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________
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