- sr-users - kamailio.org

rtpproxy - no media connection
by Steve 05 Feb '18

05 Feb '18

Dear sr-users, I am trying to set up a simple Kamailio server with rtpproxy on the same host computer, behind a NAT router. “Run your own Skype-like service” is my set-up guide, but I’m using Kamailio version 5.1 so I modified the config files that came with the download. My problem is that SIP clients can register over the internet via TLS encryption with the server and establish a call session connection, but the ZRTP media connection is not being made. Once the call session connection is made, both clients send out UDP packets to each other, but neither receives them—the media traffic is one way and being dropped somewhere. I’ve tried different rtpproxy start settings. If I start the rtpproxy service with “sudo /etc/init.d/rtpproxy start” then it uses the default “unix:/var/run/rtpproxy/rtpproxy.sock”. I’ve also tried starting with “sudo rtpproxy –l /my_public_ip/ –s udp:localhost:7722” which then asks for either –F (superuser) or –u (user). I’ve tried it both with options “–F” and “-u kamailio kamailio”. All the variations tried so far have the same result as noted above. The server behind the router is in a DMZ zone that bypasses the router firewall. Is there anything in the user-list archives that addresses this issue? Or some likely place to start troubleshooting the problem? Any suggestions would be greatly appreciated. Thanks in advance. Steve --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

1 0

IMS (ng-voice vs github)
by Amar Tinawi 04 Feb '18

04 Feb '18

Hello Dears, can you please tell me what are the differences between the github version of ims modules and the ng-voice modules introduced by Mr Carsten repo ? Thanks in advance

2 1

Fosdem dinner
by Daniel-Constantin Mierla 03 Feb '18

03 Feb '18

Hello, in the past 10 years or so, many of us met at Fosdem conference in Brussels (the edition this year happens during February 3-4) and had a dinner on Saturday evening. I know already couple of people from community that will go there, so the question is if there is any interest to organize again such event. If yes, it will be again on Saturday, on the evening of the 3rd of February. At the past editions, it turned into an all-RTC dinner, but we can consider to have it alone again (eventually with few other invited friends), if there are too many or people want a smaller meeting. Reply this week if you want to join, so we can estimate how many seats we need and have enough time to find a place that can accommodate us. More about Fosdem can be found at: - https://fosdem.org There is a devroom on Sunday with topics covering real time communications. I will have a presentation there. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

10 13

SDP Rewrite issues
by Steven Meyer 02 Feb '18

02 Feb '18

Our topology is: Client <> Client Router <> Internet <> Firewall (NAT) <> Kamailio <> FS Unfortunately, when enabling SDP Rewrite: fix_nated_sdp, 0x08, it rewrites packets to the firewall external IP, not the client router external IP. Right now we have to do a local NATed IP on each phone.

1 0

Routing based on A and B numbers
by Grant Bagdasarian 02 Feb '18

02 Feb '18

Hi, Which module would be best suited for routing calls based on A and B numbers? We're currently using the LCR module, with some additional htables to add this functionality, but I was wondering if there is a module which does exactly this? Or perhaps a combination of modules? Regards, Grant Bagdasarian Senior Developer +31765727054<tel:+31765727054> cm.com<https://cm.com> [cid:image001.png@01D39C36.945A76C0]

2 2

Distributed Registrations.
by Володимир Іванець 02 Feb '18

02 Feb '18

Hello! Yesterday I read email conversation "Distributed Presence between Multiple Kamailios" where it was suggested to use Presence module with new dmq parameter. So was wondering if someone was thinking about approach to share registrations between multiple Kamailio servers without using database? Thanks!

2 1

Contact Header need update?
by Duarte Rocha 02 Feb '18

02 Feb '18

Greetings, My proxy does some number normalization in order to guarantee E164 numeration. It also verifies for valid Calling Numbers and replaces them if needed. In order to this, i'm updating the $fU pseudo-variable. Should i update the "Contact" Header too? I've made some testing and i can get calls to work with keeping the Contact intact and changing it to the new number. What is the most correct approach? Cheers, Duarte Rocha

1 0

Distributed Presence between Multiple Kamailios
by SamyGo 01 Feb '18

01 Feb '18

Hi, I'm working on multiple Kamailios that have users distributed evenly among them, I now need to enable presence in those Kamailio boxes. The scenario is a mesh where any user may have subscribed to BLF of other user registered & calling on any other proxy. What modules can help me achieve this? can this be done without creating a new Presence-Node for the whole environment? Hope to get some suggestions on how experts here have done this. Looking forward to positive response. Regards, Sammy

3 5

Case insensitive regexp in textops::search() function
by Andrey Deykunov 01 Feb '18

01 Feb '18

Hi, I have an htable containing some black list words (i.e. user IDs or phone models). To find matches of the words in every message I use the following script: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { if ( search($shtitval(i1)) ) { // do something } } sht_iterator_end("i1"); How can I modify search($shtitval(i1)) to get case insensitive matching? The example below does not work: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { $var(re) = $shtitval(i1) + "/i"; if ( search($var(re)) ) { // do something } } sht_iterator_end("i1"); Other possible modifications of $var(re) aren't working too. Thanks, Andrey

2 2

Partial/regex match on hash table key?
by Anthony Joseph Messina 31 Jan '18

31 Jan '18

Is is possible (without iterating through all entries in a hash table) to return the value based on a partial or regex match to the key? Something like... $var(one)=$sht(a=>*$var(two)) -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 2

question about force_send_socket
by Fabian Borot 31 Jan '18

31 Jan '18

our kamailo has 2 IPs, a public (87.x.x.x) and a private (192.168.1.1). It is listening on all of them. For all calls it listens on the public IP but for some calls I need to forward the incoming INVITE to a server in the private network so I do a " force_send_socket(privateip:port)" before the route that forwards the INVITE to the server in the private network and that works. But then when the server in the private network replies with 302 I have to contact the servers inside the Contact header using the Public IP now so I do this hoping that it will switch back to the public IP: if (t_check_status("302")) { force_send_socket(PUBLIC_IP:5060); get_redirects("*"); t_relay(); } But it stays in the private, so the INVITEs to the servers inside the Contact header are sent from the Private IP and they are blocked. This would be the flow: customer INVITE -> kamailio(publicIP) kamailio (publicIP) 100 trying -> customer kamailio(privateIP) INVITE -> Server(privateIP) Server(privateIP) 302 redirect -> kamailio(privateIP) kamailio(privateIP) ACK -> Server(privateIP) after I would like this to happen kamailio(publicIP) INVITE -> Server1(publicIP) but instead this happens: kamailio(privateIP) INVITE -> Server1(publicIP) how can I switch back to the public IP to continue with this call flow? thank you

2 2

[4.2] how to send out BYE successfully to close a TCP/TLS connection which misses the connection ID?
by Jack Wang 31 Jan '18

31 Jan '18

For some reasons, My settings will cause kamailio to be restarted even if there were connections connected to it. And when one SIP client in a connection wanted to hangup the call with a WebRTC Client (SIP over websocket) after the proxy-restarting, local client was closed, however , the other side was not. >From the log below it seems that the connection missed. Jan 30 18:19:31 ./kamailio[27857]: WARNING: <core> [msg_translator.c:2777]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message So, Is it equivalent to connection ID missing? And is there any way to make Websocket found the connection? Btw, after proxy-restarting for seconds , I typed `sh kamctl stats` : websocket:ws_current_connections = 1 It showed that the WebRTC Client will re-connect automatically, So I think this connection could be found by proxy, Any suggestion? Thanks :)

1 2

Normal behaviour with loose_routing
by Loic Chabert 30 Jan '18

30 Jan '18

Hi list, I have a setup configurered has diagram below: UAC -> Proxy LB (proxy lb) -> Proxy Router/registrar (proxy B) -> Asterisk SBC My UAC is SIPP and sending a classic call scenario: INVITE - ACK - BYE. But i have some trouble to understand which queries should be loose routed and which queries should be dispatched by dispatcher module. My first invite request has been dispatched to proxy B by proxy LB. After that, ACK and BYE request should be loose_routed or should be dispatched as other request ? My proxy B is a statefull proxy, logging cdrs and other stuff ... >From my point of view, subsequent request should be loose_routed to keep dialog informations consistent. My subsequent requests have record-route headers. Should i use theses headers to route my calls ? Loose routing features only work with "Route" header ? What is the difference beetwen Route headers and Record-route headers ? Thanks in advance, Regards.

3 5

rtjson + uac + event - OK, ACK and BYE display name not updated
by Diego Nadares 30 Jan '18

30 Jan '18

Hi Guys, We are modifying "from" and "to" headers with rtjson module via a json struct. It works fine in INVITE, TRYING, RINGING and OK but not in the following messages. Kamailio It's changing ONLY "from" and "to" URI BUT NOT the DISPLAY NAME. I think it's the tm module but I can't figure out how to confirm this. Maybe I need to configure something else in my cfg or could it be a bug? My json struct that is handled by rtjson module response.routing = "serial"; //response.routing = "parallel"; response.routes = []; response.routes[0] = {}; response.routes[0].uri = "sip:1111111111@172.16.213.31:5060"; response.routes[0].headers = { // Headers to be modified "from": { "display": "1111111111", "uri": "sip:111111111@172.16.213.38" }, "to": { "display": "2222222222", "uri": "sip:2222222222@172.16.213.38" }, }; response.routes[0].headers.extra = {};//"X-Hdr-A: abc\r\nX-Hdr-B: bcd\r\n"; This is part of my cfg .... modparam("dialog","dlg_flag", FLD_START) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) .... # account only INVITEs if (is_method("INVITE") && !has_totag()) { setflag(FLT_ACC); # do accounting * setflag(FLD_START); # do accounting* setflag(FLT_ACCMISSED); setflag(FLT_ACCFAILED); } ..... >From source/Aleg: *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* To dest/Bleg: *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Incorrect display name in BYE and ACK: *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* interface: any filter: ( port 5060 ) and (ip or ip6) # U 2016/12/06 15:52:13.638731 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:5555555555@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: "SIPP" <sip:sipp@172.16.213.21:5060>. Max-Forwards: 70. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Remote-Party-ID: <sip:sipp@172.16.213.21 >:5060;party=calling;id-type=subscriber;privacy=off;screen=no>. Content-Length: 549. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.734159 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. . ................ # U 2016/12/06 15:52:13.878600 172.16.213.38:5060 -> 172.16.213.31:5060 INVITE sip:1111111111@172.16.213.31:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Max-Forwards: 69. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Content-Length: 788. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40236 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=sendrecv. a=rtcp:40237. a=ice-ufrag:QNCzXhUr. a=ice-pwd:Ci90BRpdZH51Hx5BaPgGMrGCeh. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40236 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40237 typ host. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.879441 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. ................ # U 2016/12/06 15:52:13.936565 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 180 Ringing. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 316. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:20.882165 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 213. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ................ # U 2016/12/06 15:52:21.173798 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 380. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:21.174390 172.16.213.21:5060 -> 172.16.213.38:5060 ACK sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-4. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:21.248172 172.16.213.38:5060 -> 172.16.213.31:5060 ACK sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.98116b5795d4c30f52156187a72059c8.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.178826 172.16.213.21:5060 -> 172.16.213.38:5060 BYE sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.468639 172.16.213.38:5060 -> 172.16.213.31:5060 BYE sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.469153 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.519423 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Content-Length: 0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . ................ exit 13 received, 0 dropped Sorry for my english. Thanks in advance! Diego.

2 7

Session Timers
by Duarte Rocha 30 Jan '18

30 Jan '18

Hey guys, I have two issues about Session Timers that need some clarification. 1) I have my Kamailio between my clients CPE's and my GW to the PSTN. I want to configure it in a way that i always try to make the CPE do the refreshing so I can save resources on my kamailio and in the GW. If the CPE doesn't support it, the refresh should be done by the GW. Is it possible to do this test and configuration in every call ? 2) SIP has the MIN-SE mechanic and is well documented, but is it possible to implement a MAX-SE in any way ? Thanks for the help. If you need more details about the issue, please let me know. Cheers

2 1

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Victor Seva 30 Jan '18

30 Jan '18

2018-01-26 17:46 GMT+01:00 Alain Bonnefoy - XOOL <a.bonnefoy(a)xool.fr>: > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > Wiki updated. Thanks

1 0

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Bastian Triller 30 Jan '18

30 Jan '18

There another command on http://deb.kamailio.org for people behind firewalls which block port 11371: % wget -O- http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - On Jan 26, 2018 5:46 PM, "Alain Bonnefoy - XOOL" <a.bonnefoy(a)xool.fr> wrote: > Hello, > > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > > > > But when I execute the command to add your GPG key, I just get replied > that the timeout occured. > > > > What to do ? > > > > Alain > > Provenance : Courrier <https://go.microsoft.com/fwlink/?LinkId=550986> > pour Windows 10 > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

kamailio tls passthrough
by pavel＠eremina.net 30 Jan '18

30 Jan '18

Hi, All i am trying to load balancing Kamailio with TLS in passthrough mode. is it possible to do it in stateless mode?

2 1

whats happened to iptel.org
by voipspace voipspace 29 Jan '18

29 Jan '18

Anyone know whats happened to iptel.org The webiste is now a wix page iptel.org the sip uri music(a)iptel.org was a useful resource. Now the song has changed and the service is down alot Has it it changed hands or been sold?

1 0

i can not receive BYE after websocket has been closed and reconnected during the call
by Kenan Kocaerkek 29 Jan '18

29 Jan '18

Hi, I have a WebRTC softphone based on JsSIP and kamailio 4.4.7. Generally it works well but sometimes websocket connections interrupt and reconnect immediately (especially running behind proxies). If the interruption occurs during an active call, kamailio tries to send BYE message to softphone over closed websocket. gruu_enabled flag is active on registrar module and JsSIP sends proper sip.instance value. So after ws reconnects, registered contact details do not change on kamailio except Expires value. Is it possible to send BYE message via contact's last active websocket after reconnect? Jan 29 14:35:20 registrar-kenan /usr/local/sbin/kamailio[30357]: WARNING: <script>: WebSocket connection from 195.142.112.66:49086 has closed Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: <core> [msg_translator.c:1979]: build_req_buf_from_sip_req(): could not create Via header Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (2/SL) Regards, Kenan

1 0

Ann: Registration to Kamailio World 2018 is open!
by Daniel-Constantin Mierla 29 Jan '18

29 Jan '18

Hello, a quick note to the community forums to announce that the registration to next Kamailio World Conference (May 14-17, 2018, in Berlin, Germany) is now open! More details are available at: * https://www.kamailioworld.com/k06/registration/ In a matter of several days, we expect to publish the details about the first group of accepted speakers. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

New module – embedded MSRP relay
by kossivi HOLONOU 28 Jan '18

28 Jan '18

Hello,I would like to develop a project that will allow me to: - voice, video, screen sharing, etc. sessions with content communication via RTP - end to end presence – this is purely SIP routing - SIMPLE-based presence (aka, presence server or presence agent model) via presence* and pua* modules — user presence, dialog states notification (aka, blinking lamps), resource lists service (including OMA/RCS extensions), user location states notification and replication, audio/video conference mixer notifications, a.s.o. - embedded XCAP server – management of user contact lists, presence policies, user agent configuration files, a.s.o. There is also an XCAP client extension - embedded HTTP server – for admin and user interaction with the service via pure HTTP or XMLRPC requests - embedded MSRP relay – for relaying and fine controlling of the message-based content of SIP sessions - IRC-style instant messaging conference via imc module - storage of instant messages for offline users and relay to them when they become again online via msilo module I need your help to start. I went through the document a little without understanding much. Thanks for your help

1 0

Kamailio with asterisk on private lan
by Mark Boyce 28 Jan '18

28 Jan '18

Hi all I’m trying to create a relatively simple setup with Kamailio dual homed on public/private ip and asterisk on private ip only. The idea is load balance / fail over asterisk boxes. Following the real-time tutorial I have clients registering with Kamailio, Kamailio registering on clients behalf with asterisk as well as invites going through. However what I’m seeing is that when an invite occurs asterisk offers media on its private ip, as it would. However this is making its way through Kamailio all the way to the client. After a bit of searching all I can find is people trying to get it working and failing, or putting asterisk on public IP. So questions - am I doing this completely the wrong way? Should Kamailio alter the media ip of asterisk on the way through or do I need to do that by hand? Surely someone somewhere has a write up on this already :-) Thanks Mark

7 8

SEMS with sbc application
by Mojtaba 27 Jan '18

27 Jan '18

Hello, I have a problem today, It's strange for me. Suppose we have this senario: uac1------->SEMS(mo profile)------->Kamailio-------->SEMS(mt profile)---------->uac2 In above topology, we have two interfaces(intern,extern) for SEMS, and just used as SBC (sbc application). if i used port=5060 as external port, every things is right and log file is like this: [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd msg via UDP from 89.165.117.125:42411 vv --++-- REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 89.165.117.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z- Max-Forwards: 70 Contact: <sip:4000@ 89.165.117.125:42411;rinstance=79011092e56e1a09;transport=UDP> To: <sip:4000@kava.shatel.ir;transport=UDP> From: <sip:4000@kava.shatel.ir;transport=UDP>;tag=82820e1f Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 but when i changed port=4080 for external port, The Via header and contact header are changed to my public ip, like this: [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd msg via UDP from 89.165.117.125:42411 vv --++-- REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.1.1.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z- Max-Forwards: 70 Contact: <sip:4000@172.1.1.125:42411;rinstance=79011092e56e1a09;transport=UDP> To: <sip:4000@kava.shatel.ir;transport=UDP> From: <sip:4000@kava.shatel.ir;transport=UDP>;tag=82820e1f Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 Why my public ip address is there here? I just changed external port=4080 in sems server. In SDP protocol i have the same problem when i changed external ip port in sems server. Thanks. --Mojtaba Esfandiari.S

3 3

Transcoding
by Wilkins, Steve 26 Jan '18

26 Jan '18

Hello All, I am currently using Kamailio and Asterisk on Centos 7 servers and trying to enable WebRTC jsSIP clients to be able to do Audio/Video calls with Provider Phones (Purple, Z, Sorenson, etc....), however, the providers do not have vp8 codecs (which is what the WebRTC clients use for Audio) so I believe I will need a media proxy server to resolve the video issues. My question is, can rtpproxy or rtpengine perform this transcoding? If so, and if rtpengine is the way to go, should I use Ubuntu for the rtpengine since it is the only one that seems to have a working installation? Thank you, -Steve

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