sr-users September 2020

sr-users@lists.kamailio.org

84 participants
108 discussions

Kamailio drop calls with Teams
by sip user 22 Sep '20

22 Sep '20

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

5 21

Kambdctl create command
by Pavithra Mohanraja 22 Sep '20

22 Sep '20

Hi, I need to use kambdctl create command but that has to create database in another vm where mysql is alone installed. How to do that? Thanks, Pavithra

2 1

Kamailio Slack Invite
by Peter Baines 22 Sep '20

22 Sep '20

Hi, Can someone send me an invite to the Kamailio Slack channel please. I am sure I used to be a member but I may have used a different email address. -- *Web:* http://pbaines.com *Email:* peter(a)pbaines.com

4 4

Info: recordings of Kamailio World 2020 sessions
by Daniel-Constantin Mierla 22 Sep '20

22 Sep '20

Hello, for those that couldn't attend the Kamailio World 2020 online event that happened a few weeks ago, or someone wants to view again a session, the recordings of the two days were split per session -- they are available on KamailioWorld youtube channel: * https://www.youtube.com/c/KamailioWorld/ For convenience, they were also grouped in a playlist for KW2020, see: * https://www.youtube.com/c/KamailioWorld/playlists No other editing than just cutting at start/stop of each session. Slides should be available at: * http://kamailio.org/events/2020-KamailioWorldOnline/ Enjoy! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Error on ACK/BYE After 200
by Neimar 21 Sep '20

21 Sep '20

Hello, I am trying to set up a Webrtc by using Kamailio, Asterisk and Rtpengine. So far, everything is working fine , I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension from Asterisk, Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. In console, I see: Sep 21 20:39:06 vsphone3-sbc /usr/sbin/kamailio[9756]: WARNING: {1 102 ACK 332ee81b4eb178ac156e8d3e53a98900@172.16.11.6:5060} <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:172.16.11.57:8443, to udp:187.20.56.83:27730) My scenario is: Internet -> Firewall -> Kamailio -> Asterisk Working: Softphone -> Kamailio -> Asterisk (TCP/UDP/TLS(SRTP) Asterisk -> Kamailio -> Softphone (TCP/UDP/TLS(SRTP)) Webrtc -> Kamailio -> Asterisk (WSS) Not Working: Asterisk -> Kamailio -> Webrtc After a answer from Webrtc, I can listen sound, so, RTP in theory is fine. Webrtc send 200 OK to Kamailio Kamailio Send 200 OK to Asterisk. Asterisk Send ACK to Kamailio ACK not forwarded to Webrtc. In attachment, my kamailio.cfg Every Help is welcome. Thanks Neimar.

1 0

Kamailio uses Python
by Pavithra Mohanraja 21 Sep '20

21 Sep '20

Hi, I am building kamailio as an docker image by installing kamailio using "apt-get install kamailio" in ubuntu 18.04, which installs python by default which makes my image size bigger. Does kamailio uses Python?. else what could be the alternative? Does anybody came across this situation. Kindly advise. Thanks, Pavithra

2 3

Kamailio IF Routes
by conradcordeiro＠gmail.com 21 Sep '20

21 Sep '20

Hello, Is it possible to set up Kamailio 5.4.1 to use IF to determine how calls are proxied depending on the source of the request? For example, *_if_* calls originate from /pstn.us1.twilio.com/ then _/$du = "sip01.example.com:5060"/_, wheres _*if*_ calls originate from /pstn.us2.twilio.com/ then _/$du = "sip01.example.com:5060"/_. Additionally, would this also be possible based on numbers? For example, _*if*_ calls are to number +1-800-777-8888 then /$du = "sip01.example.com:5060"/, or if calls are to number +1-800-777-9999 then /$du = "sip02.example.com:5060"/. To be clear, I don't expect both examples above to work in the same Kamailio instance but IF and AND clauses are possible, such as _*if*_ calls are to number +1-800-777-8888 AND from /pstn.us1.twilio.com/ then _/$du = "sip01.example.com:5060"/_, it'd be great to test this too. I'm not sure if I'm allowed to ask such specific questions on this mailing list and I hope I'm not asking too much. Many thanks, Conrad

3 4

SRTP/TLS BYE Problem with One of the Legs of the Call
by conradcordeiro＠gmail.com 20 Sep '20

20 Sep '20

Hello, Thank you for your reading this and for your help. I'm a Kamailio newbie and managed to set up an edge proxy, which works perfectly on UDP traffic. I'm now attempting to deploy TLS/SRTP and everything almost works perfectly. The single issue I'm having is that when either of the parties to an SRTP/TLS call disconnect, the other party's call remains active. With UDP, when one of the parties disconnects the call, the other leg of the call receives the BYE command and the call automatically disconnects. This is how I have our infrastructure set up: 1. Twilio SIP Trunk with Secure Media enabled. 2. Kamailio 5.4.1 set up with TLS module, set to listen on TLS port 5061, SSL certificates from Let's Encrypt, route set to our phone system. 3. Phone system is Asterisk. As per above, everything works almost perfectly with TLS/SRTP. The only issue is that calls will not disconnect when one of the sides hang up. If I disable TLS/SRTP and use UDP only, everything works. Audio is just fine with TLS/SRTP. Does anyone know why calls with SRTP/TLS will not disconnect automatically when one of the parties ends the call? Thank you, Conrad

2 2

Kamailio support for 5g
by Pavithra Mohanraja 18 Sep '20

18 Sep '20

Hi, Does kamailio IMS support for 5g Network functions since in 5g everything is handled via http requests. Could anyone please respond Thanks, Pavithra

3 4

IMS Diameter server module
by Pavithra Mohanraja 18 Sep '20

18 Sep '20

Hi, Can anybody reply what is the exact use of diameter server module. I am unable to understand from the kamailio doc provided. It is mentioned that it converts diameter messages to json messages. But without diameter.xml file, it is not working. Whats the exact use of it. Can i not be able to send the http request alone via Rx interface instead of using diameter.xml file. Kindly reply as i am blocked in this for few days. Thanks, Pavithra

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sr-users September 2020