- sr-users - kamailio.org

DNS Failover try next destination on receiving code other than 503
by Patrick Karton 30 May '22

30 May '22

Hello, im using DNS Failover and everything works well except i would like to try other resolved ips when i receive a 408 from the previous ones As far as i know only a received 503 will trigger next destinations and there is no config parameters in tm module to handle that behaviour. so how can i implement it ? set a failure route before relay and just call t_relay() after again ? also is there a way in failure route to know how many destinations are left to exit failure route. Thanks for help.

2 3

KEMI Python Syntax
by Olivier Roy 30 May '22

30 May '22

Hello, I am writing some Kamailio scripts using app_python3. I understand that the KSR module is automatically generated by Kamailio. So when I import KSR from KSR I don't get any syntax from my IDE since it doesn't know where KSR is. My question: is there a way to generate a KSR.py file that can be used for autocompletion and syntax highlighting? Thanks. Olivier

2 1

get and store destination ip address and port that the request will be sent to
by Patrick Karton 30 May '22

30 May '22

Hello is there a way or pseudo var to get ip address & port that the request will be sent to and save it in to another variable. i want a way to get it to store it. im not able to save it with to_ip and to_port of the onsend_route. TThanks.

2 1

Find Values in User part of Diversion URI
by Lewis Hutchinson 30 May '22

30 May '22

Sorry if this is a super basic ask but I can't seem to work it out, I have got a work around but for my own sanity wanted to post the question on here. In Kamailio Native 5.5.2 I want to be able to look into this field $(di{uri.user}) and "find" or see if it "contains" a value i.e Diversion: "TEST" sip:112233445566@xxx.xxx.xxx.xxx;user=phone; find 234 or contains 234 I know how to do this in in KEMI Python but for the life of me cant get my head around it in Native. 2ndly is there an opposite to {s.numeric} Removes all non-numeric parts of string. I want the ability to strip all numeric parts. Keep us the good work. Lewis

3 4

TLS connection failure. Not sure if system behavior is correct.
by Володимир Іванець 27 May '22

27 May '22

Hello all! My test Kamailio setup (now updated to version 5.5.4) that was working with MS Teams some time ago, now stopped :) I was wondering if someone could tell me if the current behavior is correct or if there is a problem? Here is a screenshot of the packet capture: [image: image.png] I thought that after MS Teams side sent Change Cipher Spec request, Kamailio should respond with ACK. But instead it just starts sending data (I believe it sends OPTIONS request. sipdump module captured it). Is this correct? There seems to be no issues in the Kamailio debug log: May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0xae9560, 24, 0x1, 0xffffffff) fd_no=20 called May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xae9560, 25, -1, 0x0) fd_no=20 called May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4471]: handle_tcpconn_ev(): sending to child, events 1 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4144]: send2child(): selected tcp worker idx:1 proc:11 pid:28972 for activity on [tls:172.16.30.206:5062], 0x7fad6a70f6c8 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1737]: handle_io(): received n=8 con=0x7fad6a70f6c8, fd=6 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7fad6a5e1eb8: (nil) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2720]: tcpconn_do_send(): sending... May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2753]: tcpconn_do_send(): after real write: c= 0x7fad6a70f6c8 n=2279 fd=6 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2754]: tcpconn_do_send(): buf= May 20 15:29:05 server kamailio[28961]: 7 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): processing io_watch_add(0xb555c0, 6, 2, 0x7fad6a70f6c8) - fd_no=1 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7fad6a5e1eb8: (nil) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:790]: sr_ssl_ctx_info_callback(): SSL handshake done May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:794]: sr_ssl_ctx_info_callback(): SSL disable renegotiation May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:549]: tls_connect(): tls_connect: new connection to 52.114.132.46:5061 using TLSv1/SSLv3 ECDHE-RSA-AES256-GCM-SHA384 256 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:552]: tls_connect(): tls_connect: sending socket: 172.16.30.206:0 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/CN=sip.pstnhub.microsoft.com May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft RSA TLS CA 01 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2720]: tcpconn_do_send(): sending... May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2753]: tcpconn_do_send(): after real write: c= 0x7fad6a70f6c8 n=509 fd=6 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2754]: tcpconn_do_send(): buf= May 20 15:29:05 server kamailio[28961]: ▒4▒▒▒▒.▒2T$-▒▒ o▒w)▒(▒▒^▒D▒U{xn9▒' ▒&▒t▒▒)▒▒?▒d▒=g▒▒?▒▒▒4/▒▒▒▒-R+▒tR▒H,▒o▒D2r ▒5Q?m▒▒9▒▒#▒O▒▒▒\eϣ▒݄"▒z=▒U▒? May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xae9560, 24, -1, 0x0) fd_no=19 called May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4471]: handle_tcpconn_ev(): sending to child, events 1 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4144]: send2child(): selected tcp worker idx:2 proc:12 pid:28973 for activity on [tls:172.16.30.206:5062], 0x7fad6a6f25a0 May 20 15:29:05 server kamailio[28961]: 12(28973) DEBUG: <core> [core/tcp_read.c:1737]: handle_io(): received n=8 con=0x7fad6a6f25a0, fd=6 May 20 15:29:05 server kamailio[28961]: 12(28973) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7fad6a5e1eb8: (nil) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:304]: tcp_read_data(): EOF on connection 0x7fad6a70f6c8 (state: 0, flags: 18) - FD 6, bytes 0, rd-flags 10000 ([52.114.132.46]:5061 -> [52.114.132.46]:0)11(28972) DEBUG: <coreTCP closed event creation triggered (reason: 0) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:192]: tcp_emit_closed_event(): no callback registering for handling TCP closed event May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1503]: tcp_read_req(): EOF May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xb555c0, 6, -1, 0x10) fd_no=2 called May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1878]: handle_io(): removing from list 0x7fad6a70f6c8 id 2 fd 6, state 2, flags 18, main fd 25, refcnt 2 ([52.114.132.46]:5061 -> [52.114.132.46]:0) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1659]: release_tcpconn(): releasing con 0x7fad6a70f6c8, state -1, fd=6, id=2 ([52.114.132.46]:5061 -> [52.114.132.46]:0) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1660]: release_tcpconn(): extra_data 0x7fad6a70f008 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:3574]: handle_tcp_child(): reader response= 7fad6a70f6c8, -1 from 1 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fad6a70f008 Thank you!

1 1

Debian: Kamailio: -> FIN - SYN -> on new port
by Kevin Browne 26 May '22

26 May '22

Hi, am looking for some specific Kamailio help i cant seem to find anywhere ( # running Kamailio on Debian/buster ... ## after receipt of tcp FIN, port is closed. when signaling is required again from Kamailio side. Kam is attempting SYN on FINd (closed) port. this is rejected RST. How to change SYN to re-establish tcp on new ephemeral port? ## another description, same problem: FIN received from (external) cisco (call owner) due to port recycling, not to keep port open after x minutes. everything is as expected re ACKs and such. When Kam wants to send REFER to cisco after the FIN, tcp attempts a SYN (as we'd expect) but sends SYN on old closed port. ## Where to change this Kam/OS/other is where i'm attempting to identify, then figure how to change. Thank you.

2 1

TOPOS + RE-INVITE problem - Contact IP not updated
by c c 24 May '22

24 May '22

It seems that TOPOS is not working properly on RE-INVITE and stop masking IP and contact on some RE-INVITE. When a RE-INVITE is send with Kamailio IP as Request-URI TOPOS is not operating. Note : we are using the HTABLE trick to rewrite $ru and match the call but we would need to have TOPOS re-evaluating the dialog once done. We use a standard configuration with 2 modifications. 1. in route[DISPATCH] to set the HTABLE if(is_method("INVITE")) { $sht(ct=>$ci::$ft) = $sel(contact.uri); } 1. in route[WITHINDLG] to find the call if R-URI = myself if(has_totag() && uri==myself) { if($sht(ct=>$ci::$ft) != $null && $T_reply_code > 400) $shtex(ct=>$ci::$ft) = 10 ; if($sht(ct=>$ci::$tt) != $null) { $ru = $sht(ct=>$ci::$tt); if (is_method("BYE")) $shtex(ct=>$ci::$tt) = 10 ; route(RELAY); } } As you can see in the PCAP the RE-INVITE goes directly to calling stating with the Contact not updated and the 200 OK is also sent to called party with wrong Contact info too. [ https://github.com/kamailio/kamailio/files/8730489/topos-issue.zip ] This behavior breaks the BYE procedure. Any advice ? Chris

2 1

Load balancer and consult transfer
by Anh Cao 24 May '22

24 May '22

Hi all, I'm building an simple PBX with 1 Kamailio as load balancer for several Asterisk servers. My issue is if an extension is in-call on Asterisk#1 wants to consult transfer to another number, it makes another call and Kamailio forward this new call to another Asterisk and cause call tranfer will fail. Does anyone here have tips or sugestions to resolve this? Thanks in advance for your help. Best regards

3 3

Kamailio v5.6.0 Released - new major version is out
by Daniel-Constantin Mierla 23 May '22

23 May '22

Kamailio v5.6.0 is out – it comes with 7 new modules and a considerable set of improvements touching again more than 90 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-6-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.6.0 brings more flexibility and optimizations across many components, plenty of new variables and transformations, load balancer enhancements, generic iterators for SIP headers and body lines, alternative http client, NATS PubSub connector, improvements to STIR/SHAKEN, IMS/VoLTE, rtpengine and tls as well as more functions exported to KEMI interface ... just to list only a few here. Enjoy Kamailio v5.5.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online: June 20-23, 2022 * https://www.asipto.com/sw/kamailio-advanced-training-online/

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Releasing Kamailio v5.6.0
by Daniel-Constantin Mierla 23 May '22

23 May '22

Hello, with some effort directed lately to convert the wiki content to markdown files and host them on github, it's time to now to nail down the path to release nest major release for Kamailio, respectively v5.6.0. What's new in this major release as well as upgrade guidelines have to be built, but I hope to be done in time to release Kamailio v5.6.0 either on Monday, June 23, 2022 or on Wednesday, June 25, 2022. Should anyone consider a different release planning, let's discuss it. If anyone wants to help with them, the content has to be added to the github wiki repository: * https://github.com/kamailio/kamailio-wiki If not there, new files have to be created and linked from the main page. The wiki is built with mkdocs into html and made available for browsing at: * https://kamailio.org/wikidocs/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online: June 20-23, 2022 * https://www.asipto.com/sw/kamailio-advanced-training-online/

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Re: [SR-Users] TLS connection failure. Not sure if system behavior is correct.
by Ahmed Fouad 22 May '22

22 May '22

Hi, You are currently using TLSv1/SSLv3 ECDHE-RSA-AES256-GCM-SHA384 256 Teams stopped using TLSv1/ TLSv1.1 and only allowing TLS1.2+ So change your kamailio TLS settings to enforce only it method = TLSv1.2+ After restarting kamailio your team's connections should be good. -- Best Regards, Ahmed Fouad

1 0

LCR GW
by Antoine DEBARRE 20 May '22

20 May '22

Hi, Is there a way to specify an outbound proxy for just a few lcr gateways added in table lcr_gw ? Thanks in advance ! Antoine

2 6

Path MTU issues over UDP/IPv6
by Rick van Rein 20 May '22

20 May '22

Hello, IPv6 routers never fragment packets. Rather, they drop a packet that is too large for a (local) MTU and send back ICMPv6 "Packet Too Big". This seems to cause loss of larger SIP messages when an ISP tunnels their IPv6 at the expense of the MTU. The pmtu_discovery flag sets Don't Fragment in IPv4 traffic; in IPv6 this is an implied property. Does Kamailio learn a lower MTU from any "Packet Too Big" for IPv6 even if pmtu_discovery is not set? Future resends can then be fragemented appropriately. The udp_mtu setting diverts to another protocol, but that would be a setting as low as the worst peer, impacting all. It would be a weird struggle with a telco serving many. PMTU would be better to rely on, but how does it work in Kamailio? Details on https://www.rfc-editor.org/rfc/rfc3542#section-11.3 https://stackoverflow.com/questions/38817837/how-does-mtu-retransmission-wo… Thanks, -Rick

3 4

Forwarding SIP status code to HTTP client
by Tim Anderson 19 May '22

19 May '22

Hi, I have a kamailio that is has an HTTP message server. Is it possible to send the SIP status code back in the HTTP reply? Is it possible to suspend an xhttp request while some other stuff happens and then send the reply later? The relevant bit of the config currently looks like event_route[xhttp:request]{ if($hu == "/sms"){ $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:1234@example.com"; $uac_req(furi)="sip:4567@example.com"; $uac_req(turi)="sip:1234@example.com"; $uac_req(body)=$rb; uac_req_send(); xhttp_reply("200", "OK", "text/plain", "Sent") } } Regards, Tim

2 1

Kamailio Stop Responding Randomly
by M Arqum CH 19 May '22

19 May '22

Kamailio first gives timeout on ctl unix socket as I am exporting data using prometheus exporter, after a few minutes it stops responding to any sip request on default/production udp port binded with multiple IP addresses for sip traffic. The other listing port which is used by keepalived check script for bing-pong keeps responding to option messages. Its very random behaviour occurs after a few days or after a few months, the system having max load 350 concurrent calls with load average 200-300. I am attaching core dump which is collected before restarting the process. Please let me know if more information is required. version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 Thankyou in advance. -- Regards M Arqum

2 4

Relay wss to Asterisk
by Michal Hajek 18 May '22

18 May '22

Hello, can i configure kamailio for relay wss to asterisk/pjsip? WebRTC client ( chrome ) --> WSS -> Kamailio --> WSS -> Asterisk with pjsip Asterisk accept wss://ip_address:wss_port/ws Thanks Michal Hajek

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Split audio and video into different calls
by Rüdiger, Christoph 18 May '22

18 May '22

Hi everyone, we are running a call center with really good to use hardphones and audio only up to this point. Because of changing regulation, we soon have to deal with video calls as well, but only for a fraction of the calls (below 1%). Instead of switching everyone to clumsy softphones for the rare case of a video call, I’m exploring the ability to branch an incoming call into a pure audio call to the hardphone and optionally a pure video call to the softphone. Has anyone done this in the past and can share some experiences/ideas on 1. Can it be done purely with Kamailio and rtpengine? I don’t have a clear picture of how to do it purely with Kamailio on the SIP side, because we end up with two independent legs (to the hardphone and the softphone). Or do I need to deploy an Asterisk, Yate, freeswitch, etc. and doing it as a 3-way conference where one party gets both RTP streams (the trunk) and the other two parties only get one RTP stream (either audio or video)? 2. Was it worth the effort or a steady candidate for troubles? I’m looking forward to your thoughts, especially on question b). Have a nice weekend, Christoph --- Christoph Rüdiger Digital Operations Architect DTX OU DACH M +49 152 3828 0376 | christoph.ruediger(a)tkelevator.com<mailto:christoph.ruediger@tkelevator.com> TK Aufzüge GmbH | Münchener Str. 100c | 45145 Essen | Deutschland | www.tkelevator.com/de<http://www.tkelevator.com/de> VORSITZENDE DES AUFSICHTSRATS Inge Delobelle | GESCHÄFTSFÜHRUNG Burkhard Schlenker (Vorsitzender) | Christian Fongern | Katja Schrafft | SITZ DER GESELLSCHAFT Neuhausen a. d. F. | REGISTERGERICHT Stuttgart HRB 731152 This e-mail (including any attachments) may contain confidential and/or privileged information. Any unauthorized use or dissemination of this message in whole or in part is strictly prohibited. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail.

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kamailio WebSocket not respond sometimes
by dolphin 18 May '22

18 May '22

Hello, kamailio could not respond to the question: This machine is on aws, there is no high load, there is an external ip listening to the WebSocket 443 port, and sometimes there are some unknown detection signals "SSTP_DUPLEX_POST" that will cause port 443 to fail to respond. Need to restart kamailio Then, it stopped responding again after about a month. Attach the detailed log details: root@sip1-aws:~# /home/pkg/kamailio/sbin/kamailio -V version: kamailio 5.3.9 (x86_64/linux) a5fd0a flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: a5fd0a compiled on 22:31:18 Mar 31 2022 with gcc 5.4.0 Thanks.

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RabbitMQ hangs if connection lost
by Avremy Cohen 17 May '22

17 May '22

Hi When I Publish to RabbitMQ and the connection is down, the server just hangs and there is no timeout. I am using KEMI with python3 and have tried different timeout functions to circumvent the hanging but it doesn't seem to work 1. Is there a way to tell if RabbitMQ is connected? 2. using python is there a way to timeout a KSR function? Thanks guys!

1 0

Kamailio Crashed
by M Arqum CH 17 May '22

17 May '22

Dear Team, Kamailio instance got crashed, I found this error in logs. Please give advice on how I can avoid this in future. *kernel: [52641555.303165] kamailio[123814]: segfault at 30 ip 00007f42f70db770 sp 00007ffd5d95a9f0 error 4 in dialog.so[7f42f7033000+c9000]* version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Thank you in advance. Regards M Arqum

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Failing Audio stream on Kamailio+webrtc with integration to asterisk server
by Kelvin Ezenwaka 16 May '22

16 May '22

Summary: We have a Kamailio server as our sip proxy server, sip firewall with WebSocket and RTP engine configured on it (the Kamailio Server). But we experience one way audio, no audio, hang up after 30s, when we try making calls between internal extensions (eg extension 100 to call extension 105) and external calls (eg extension 100 to call mobile number 09056925668) as well. The call flow is as: Webrtc client<------->Kamailio1+rtpengine<-----Asterisk---->Kamailio2<-------->Telco Provider The asterisk communication between all three boxes is via local IP (All ports open between them). The Kamailio 1 box where we have our sip registration cache, has rtp ports and wss port open on the internet. Asterisk 18.9.0 Kamailio 5.5.3 + RTPengine 10.4.0.0 Debian 11 bullseye Kindly see attached below a diagram depicting the VoIP network flow and also attached are logs files for the webrtc client and server side. Ezenwaka Kelvin Arinze VOIP Engineer Tel: (+234) 9056925668 DISCLAIMER: The message and its attachments are for designated recipient(s) only and may contain privileged, proprietary and private information. If you have received it in error, kindly delete it and notify the sender immediately. PressOne accepts no liability for any loss or damage resulting directly and indirectly from the transmission of this e-mail message.

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Question on pike_check_ip(ipaddr)
by Joel Serrano 13 May '22

13 May '22

Hello, One of our customers has lots of extensions that register to our platform using all the same IP. Sometimes they get caught in our pike velocity rules but if we increase the values to accommodate this client then all the others could abuse the system. I was wondering if pike_check_ip(ipaddr) needs strictly an IP as parameter or can it be a string? I was thinking of using pike instead of by $si by something like $fU-$si but I don't know if this would work? I need to either handle separately different extensions from the same IP or a way to adjust the pike config values just for one specific IP... I'm not sure either of these is possible with pike? My alternative is just use table and mimic the functionality myself which I don't think is too complicated, but I'd love to know if I'm missing a fast-path here... Thanks! Joel.

2 1

Accept IP/Domain Register
by Social Boh 12 May '22

12 May '22

Hello, maybe is a very basic question; How Kamailio can accept any kind of register, over IP or domains. Example, accept authenticate REGISTER from ip 1.2.3.4 or sip.domain.org or sip1.domain.org, domains associated with ip 1.2.3.4 Thank you in advance. Regards -- --- I'm SoCIaL, MayBe

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Info: converting dokuwiki content to markdown files
by Daniel-Constantin Mierla 12 May '22

12 May '22

Hello, it was discussed a couple of times during the past Kamailio development meetings and several days ago the works started to convert the Kamailio Wiki content to markdown format and host the files on Kamailio github.com organisation. The Dokuwiki engine has served us very well during more than a decade and a half, but open editing had to be disabled many years ago due to bad actors pushing inappropriate content. The contributors had to make a dedicated account on our portal only for Wiki editing. More recently we had add another layer of protection requiring to grant manually edit permissions to newly created wiki accounts because bots started to work around CAPTCHA and other security extensions, posting again inappropriate content. All these added to operational overhead, but also made it difficult for new people to contribute, specially for small editing to fix typos or add usage examples. Thanks to Kenneth Stowe, who undertook the first big effort of converting the Dowkuwiki files to markdown format, we are now in the position of moving further with the plans of using github for hosting the content of the wiki and enable easier contributions via pull requests. The history of changes done via Dokuwiki show up now as git commits. The github repository for the wiki content is available at: * https://github.com/kamailio/kamailio-wiki It is still work in progress, therefore we ask for the help of community to speed up the conversion, among the tasks that need to be done: * convert relative links between wiki pages – the conversion tools used relative paths without file extensions, like: [Install Kamailio v4.4.x From GIT](/install/4.4.x/git) They have to be converted to relative path with .md extension in order to be able to browse directly :n the github.com portal — practically the leading / has to be removed and .md added at the end, like: [Install Kamailio v4.4.x From GIT](install/4.4.x/git.md) * some section heading markers were not detected/converted, onee may notice many equal signs (=) around a section name, it has to be replaced with the appropriate markdown format based on hash signs (#). For example: ======= Install And Maintain Kamailio v4.4.x From GIT ======= Should be converted to: # Install And Maintain Kamailio v4.4.x From GIT * some Dokuwiki paragraphs used special formatting based on plugins, for example the notes, warnings, … had special colors. The common markdown format does not specify a syntax to color the text, the propose solution is to use emoji at the start of the paragraph to signal its special purpose. For example, the Dokuwiki content: <color blue>//IMPORTANT: Content for older releases than 3.2.0 is available at the **Old Wiki Site**://</color> Has to be transformed to: :fire:*IMPORTANT: Content for older releases than 3.2.0 is available at the **Old Wiki Site**:* You are welcome to propose a better syntax for this kind of transformation. * review config examples and evaluate what improvements can be done (e.g., in terms of formatting and syntax highlighting) Any of these changes do not require any C programming knowledge, nor much Kamailio config knowledge. Can be your first contribution to the project and it will be very appreciated. It can also help you learn more about Kamailio, because you may discover documentation that you were not aware of. A very important decision is how to present the wiki markdown content on kamailio.org website. Besides being able to see it on github repository, we want to process the markdown pages somehow and make them available in html format on kamailio.org website for easier navigation and better indexing (e.g., with tables of content for large pages). If you are aware of a suitable cli or web app that can convert markdown to html and generate table of content, let us know. Any questions and suggestions should be addressed to sr-users(a)lists.kamailio.org mailing list. Looking forward to your contributions on this community effort! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online * https://www.asipto.com/sw/kamailio-advanced-training-online/

5 15

unable to get local issuer certificate
by Володимир Іванець 12 May '22

12 May '22

Hello all! I'm trying to establish TLS connection but getting the following error. Can anyone point me in the right direction, please? tls_dump_verification_failure(): verification failure: unable to get local issuer certificate Kamailio version is 5.5.1. System runs on CentOS 7. At the moment tls.cfg configuration file looks like this: [server:default] method = TLSv1+ require_certificate = no verify_certificate = no private_key = /var/kamailio/certificates/default/server/key.pem certificate = /var/kamailio/certificates/default/server/cert.pem ca_list = /var/kamailio/certificates/default/CA/cert.pem [client:default] method = TLSv1+ require_certificate = no verify_certificate = no private_key = /var/kamailio/certificates/default/server/key.pem certificate = /var/kamailio/certificates/default/server/cert.pem ca_list = /var/kamailio/certificates/default/CA/cert.pem ca_list file contains root and intermediate certificates. Certificate was issued by Sectigo. It can be successfully verified with OpenSSL tool: # openssl verify -verbose -CAfile /var/kamailio/certificates/default/CA/cert.pem /var/kamailio/certificates/default/server/cert.pem /var/kamailio/certificates/default/server/cert.pem: OK Here is a fragment of Kamailio debug output: May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: <core> [core/tcp_main.c:1993]: tcp_send(): no open tcp connection found, opening new one May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): tcpconn_new: new tcp connection: 52.114.132.46 May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 5061, type 3, socket -1 May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: <core> [core/tcp_main.c:1498]: tcpconn_add(): hashes: 3678:784:0, 230 May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tls [tls_server.c:244]: tls_complete_init(): completing tls connection initialization May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tls [tls_server.c:207]: tls_get_connect_server_name(): xavp with outbound server name not found May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tls [tls_server.c:180]: tls_get_connect_server_id(): xavp with outbound server id not found May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tls [tls_server.c:187]: tls_get_connect_server_id(): outbound server id not set May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tls [tls_server.c:274]: tls_complete_init(): Using initial TLS domain TLSc<default> (dom 0x7f1cca178720 ctx 0x7f1cca29dbd0 sn []) May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7f1cca29dbd0: (nil) May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tls [tls_domain.c:778]: sr_ssl_ctx_info_callback(): SSL handshake started May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: <core> [core/tcp_main.c:2888]: tcpconn_1st_send(): pending write on new connection 0x7f1cca41fe18 sock 11 (-1/517 bytes written) (err: 11 - Resource temporarily unavailable) May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: tm [uac.c:686]: send_prepared_request_impl(): uac: 0x7f1cca40bd50 branch: 0 to 52.114.132.46:5061 May 5 06:51:03 server kamailio[3834]: 5(3844) DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/tcp_main.c:3793]: handle_ser_child(): read response= 7f1cca41fe18, 5, fd 26 from 5 (3844) May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xae4760, 26, 2, 0x7f1cca41fe18), fd_no=20 May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0xae4760, 26, 0x1, 0xffffffff) fd_no=21 called May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0xae4760, 24, 0x1, 0xffffffff) fd_no=21 called May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xae4760, 26, -1, 0x0) fd_no=21 called May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/tcp_main.c:4457]: handle_tcpconn_ev(): sending to child, events 1 May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/tcp_main.c:4130]: send2child(): selected tcp worker idx:3 proc:13 pid:3852 for activity on [tls:X.X.X.X:5062], 0x7f1cca41fe18 May 5 06:51:03 server kamailio[3834]: 13(3852) DEBUG: <core> [core/tcp_read.c:1737]: handle_io(): received n=8 con=0x7f1cca41fe18, fd=6 May 5 06:51:03 server kamailio[3834]: 13(3852) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7f1cca29dbd0: (nil) May 5 06:51:03 server kamailio[3834]: 14(3853) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0xae4760, 25, 0x1, 0xffffffff) fd_no=20 called May 5 06:51:03 server kamailio[3834]: 13(3852) DEBUG: <core> [core/tcp_main.c:2706]: tcpconn_do_send(): sending... May 5 06:51:03 server kamailio[3834]: 13(3852) DEBUG: <core> [core/tcp_main.c:2739]: tcpconn_do_send(): after real write: c= 0x7f1cca41fe18 n=5103 fd=6 May 5 06:51:03 server kamailio[3834]: 13(3852) DEBUG: <core> [core/tcp_main.c:2740]: tcpconn_do_send(): buf= May 5 06:51:03 server kamailio[3834]: ? May 5 06:51:03 server kamailio[3834]: 13(3852) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xb50560, 6, 2, 0x7f1cca41fe18), fd_no=1 May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7f1cca29dbd0: (nil) May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_domain.c:790]: sr_ssl_ctx_info_callback(): SSL handshake done May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_domain.c:794]: sr_ssl_ctx_info_callback(): SSL disable renegotiation May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_server.c:549]: tls_connect(): tls_connect: new connection to 52.114.132.46:5061 using TLSv1/SSLv3 ECDHE-RSA-AES256-GCM-SHA384 256 May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_server.c:552]: tls_connect(): tls_connect: sending socket: X.X.X.X:0 May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/CN=sip.pstnhub.microsoft.com May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft RSA TLS CA 01 May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_server.c:558]: tls_connect(): WARNING: tls_connect: server certificate verification failed!!! May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: tls [tls_dump_vf.c:104]: tls_dump_verification_failure(): verification failure: unable to get local issuer certificate May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: <core> [core/tcp_main.c:2706]: tcpconn_do_send(): sending... May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: <core> [core/tcp_main.c:2739]: tcpconn_do_send(): after real write: c= 0x7f1cca41fe18 n=513 fd=6 May 5 06:51:04 server kamailio[3834]: 13(3852) DEBUG: <core> [core/tcp_main.c:2740]: tcpconn_do_send(): buf= Thank you very much!

2 4

Using TLS in Kamailio behind AWS NLB
by Aditya Uppuluri 12 May '22

12 May '22

Hello Team, I want to understand if there is any recommendation on installing kamailio behind AWS NLB. I want to enable TLS at the NLB (using the certificate generated via Aws Certificate Manager) but still use the TLS module functionalities in kamailio (like listening on 443 port/updating the correct via header and all). Because NLB by default doesn't understand sips/TLS, I need a way to differentiate whether the request came on a secure line or on a plain UDP line. How do I configure this? Do I need to generate new certificates for kamailio ? Thanks in advance Regards, Aditya

1 0

Detect RTP stream end (Kamailio + rtpengine)
by Rawad Kassir 11 May '22

11 May '22

Hello, I am trying to play early media purely relaying on kamailio and i have accomplished that using the below: rtpengine_manage(); append_to_reply("Content-Type: application/sdp\r\n"); $var(body)= $(rb{re.subst,/(IP4.).*/\127.0.0.1/g}); set_reply_body($var(body), "application/sdp"); send_reply("100","Trying"); send_reply("183","Session In Progress"); play_media("totag=$ft file=/home/rawad/test.wav"); My problem is that once the RTP stream starts the call is kept open so my question is is there a way for kamailio to detect the end of rtp stream so that i can send BYE or can kamailio do that automatically.

1 0

Proxy for ACK
by Rick van Rein 09 May '22

09 May '22

Hello, RFC 3261 gives (by way of example) an ACK that is sent directly to a remote phone via its Contact: header address. Is this the _only_ way of doing this, or may ACK also be passed through a SIP proxy? I found that my Kamailio setup does not forward ACK, and I wonder why; whether that is a common refusal, or my fault in modifying the proxy script. I am working on an IPv4-only to IPv6-only translation proxy, and would like to leave the DNS-related work to a more advanced SIP proxy. The translator is supposed to be small/simple, so it can be integrated into home routers. I would like to avoid DNS resolution (and its delays) for Contact: URIs. Proxy script, https://gitlab.com/arpa2/mkroot/-/blob/master/contrib/sip-trunk/etc/kamaili… SIPproxy64 translator, https://gitlab.com/0cpm/sipproxy64 Thanks, -Rick

2 4

dns_query container no found
by David Villasmil 06 May '22

06 May '22

Hello guys, I'm trying to use: if(dns_query("$du", "xyz")) And all I get is dns_update_pv(): container not found: xyz Any ideas? I'm not really sure what that means. I know there's supposed to be a $dns(xyz) but it looks like it's not there? I'm using 5.0 BTW. Thanks all & Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

calling failure_route
by David Villasmil 06 May '22

06 May '22

Hello guys, Is it possible to call a failure_route like a normal route? i.e.: { t_on_failure("myroute"); } ... route("myroute"); failure_route[myroute] { ... } Let me explain: I have this scenario where a DNS name may get deleted and a dns resolution will fail. There's no pinging to that gateway so i need to: - skip that gateway when dispatching to the setid - detect whether the domain is up and available And failover to the next gateway. I don't know of any other way so on the DISPATCH route, I'm doing: route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); if(dns_query("$du", "xyz")) { t_on_failure("RTF_DISPATCH"); route(RELAY); } else { route(RTF_DISPATCH); } exit; } to try to resolve and if it fails, it will just go to the failover route. makes sense? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 4

BYE is being sent using the Dialog ID (did)
by mack＠dopensource.com 06 May '22

06 May '22

I configured the dialog module to match routing based on SIP elements versus the (Dialog ID) did. It looks like it’s using the callid SIP element, but it’s still looking it up using the dialog module. I was hoping it would route on the RURI of the BYE. But, it looks like it’s using the dialog lookup table, which has the wrong RURI because that was the original server we sent the request to. How can I get Kamailio to route based on the RURI within the BYE. May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: rr [rr_cb.c:94]: run_rr_callbacks(): callback id 0 entered with <lr> May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:887]: internal_get_dlg(): ref dlg 0x7f6d97faf200 with 1 -> 3 May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:891]: internal_get_dlg(): dialog callid='oeIojv5OA59S9FYCCu7BcQ..' found on entry 777, dir=2 to-tag='sip+3+f36a00a5+44b60208' May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_profile.c:536]: set_current_dialog(): setting current dialog [777:1428] May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:1290]: next_state_dlg(): dialog 0x7f6d97faf200 changed from state 4 to state 5, due event 7 (ref 3) May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:1071]: dlg_ref_helper(): ref op on 0x7f6d97faf200 with 1 from dlg_handlers.c:1441 May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:1075]: dlg_ref_helper(): ref dlg 0x7f6d97faf200 with 1 -> 4 May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_handlers.c:1452]: dlg_onroute(): BYE successfully processed May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:1091]: dlg_unref_helper(): unref op on 0x7f6d97faf200 with 2 from dlg_handlers.c:1476 May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:1095]: dlg_unref_helper(): unref dlg 0x7f6d97faf200 with 2 -> 2 May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:1091]: dlg_unref_helper(): unref op on 0x7f6d97faf200 with 1 from dlg_hash.c:1109 May 5 06:28:13 sbc-atlanta /usr/sbin/kamailio[27066]: DEBUG: dialog [dlg_hash.c:1095]: dlg_unref_helper(): unref dlg 0x7f6d97faf200 with 1 -> 1 Thanks in advance

2 1

Tips on picking random element from a list
by Joel Serrano 05 May '22

05 May '22

Hi Everyone, I have to implement a flow where outbound calls get their caller-id set with a random number from a pool. I've been looking around and, as most of the time, there are tons of ways to implement something like this... This is a simple implementation, the number pool is fixed and will not change (so I could have it hard-coded in the config if required). My initial approach was: 1- Try to keep it simple, with no dependencies, and all done in the config script. 2- If #1 is "ugly", then I'd probably go with some inline code using python/lua/etc 3- If #2 is "ugly", then I'd go with having the info in a DB and just run a query I think #2 and #3 might be overkill for something so "simple" but I haven't had to work too much with randomness in Kamailio, so any tips/suggestions on where to store the number list and how to pick one randomly would be awesome! Thanks!! Joel.

5 7

failover on dns resolve error
by David Villasmil 05 May '22

05 May '22

Hello guys, I have a scenario where a dns name I've added as a dispatcher endpoint may get removed completely, i.e.: dns resolution fails completely. How would you recommend implementing failover in this case? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

iterating over htable with lua
by David Villasmil 05 May '22

05 May '22

Hello guys, I'm trying to iterate over an htable in lus, this is what i have: =================================== loadmodule "app_lua.so" modparam("app_lua", "load", "/etc/kamailio/myscript.lua") modparam("htable", "htable", "mydata=>size=16;autoexpire=0;dbtable=mydata;dbmode=0;") route { lua_run("match_mydata"); } ... function match_mydata() KSR.htable.sht_iterator_start("i1", "mydata") while KSR.htable.sht_iterator_next("i1") do KSR.log("err","WHILE: iterating on: " .. KSR.htable.sht_get("mydata", "i1")) end KSR.htable.sht_iterator_end("i1"); end But i keep getting: 0(1255) DEBUG: app_lua [app_lua_api.c:666]: app_lua_run_ex(): executing Lua function: [[match_mydata]] 0(1255) DEBUG: app_lua [app_lua_api.c:668]: app_lua_run_ex(): lua top index is: 12 0(1255) ERROR: app_lua [app_lua_api.c:719]: app_lua_run_ex(): error from Lua: /etc/kamailio/myscript.lua:11: attempt to call field 'sht_iterator_start' (a nil value) 0(1255) ERROR: app_lua [app_lua_api.c:734]: app_lua_run_ex(): error executing: match_mydata (err: 2) This almost exact same thing works on the cfg: sht_iterator_start("i1", "msrn"); while(sht_iterator_next("i1")) { xlog("data: $shtitkey(i1)\n"); } sht_iterator_end("i1"); Anyone know what I could be doing wrong? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 1

Development is open for future release of v5.7.x series
by Daniel-Constantin Mierla 05 May '22

05 May '22

Hello, the branch 5.6 was created, therefore the master branch is open for adding new features, to be part of future release series v5.7.x (or whatever version is decided for next series). Any bug fix committed to master that applies to 5.6.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.6 or 5.5. Expect that v5.6.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.7.0 should be out after another 6-10 months of development, plus 1-2 months of testing, so sometime by end of 2022 or in the first part of 2023. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

Branch 5.6 has been created
by Daniel-Constantin Mierla 05 May '22

05 May '22

Hello, the branch 5.6 has been created, to be used for releasing v5.6.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.6 cd kamailio-5.6 git checkout -b 5.6 origin/5.6 Pushing commits in this branch: git push origin 5.6:5.6 Note that 5.6 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.5 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.6 will be out, respectively Kamailio v5.6.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

Development for v5.6.x is frozen
by Daniel-Constantin Mierla 05 May '22

05 May '22

Hello, the formal notification that the development for the next major version 5.6.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.6 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 5.6. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added for 5.6. Once the branch 5.6 is created, new features can be pushed again to master branch as usual. From that moment, the v5.6.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.6 is going to be built at: * https://www.kamailio.org/wiki/features/new-in-5.6.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.5.x-to-5.6.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.5.x to 5.6.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

unexpected error log when KSR.xhttp.xhttp_reply(404...)
by hanxiaofang 05 May '22

05 May '22

Hello Kamailio community kamailio v5.5.4 debian10 Lua Scripting Routing: listen=tcp:127.0.0.1:5060 tcp_accept_no_cl=yes loadmodule "xhttp.so" loadmodule "app_lua.so" modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") cfgengine "lua" function ksr_xhttp_event() KSR.set_reply_no_connect() KSR.set_reply_close() KSR.xhttp.xhttp_reply(404, "Not Found", "text/plain", "Not Found") end curl http://127.0.0.1:5060 -vvv * Expire in 0 ms for 6 (transfer 0x555d93fd9fb0) * Trying 127.0.0.1... * TCP_NODELAY set * Expire in 200 ms for 4 (transfer 0x555d93fd9fb0) * Connected to 127.0.0.1 (127.0.0.1) port 5060 (#0) > GET / HTTP/1.1 > Host: 127.0.0.1:5060 > User-Agent: curl/7.64.0 > Accept: */* > < HTTP/1.1 404 Not Found < Sia: SIP/2.0/TCP 127.0.0.1:54944 < Content-Type: text/plain < Server: kamailio (5.4.4 (x86_64/linux)) < Content-Length: 9 < * Connection #0 to host 127.0.0.1 left intact The output of curl is OK, but have some unexpected error in kamailio.log ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 16) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 404 Not Found#015#012Sia: SIP/2.0/TCP 127.0.0.1:54946#015#012Content-Type: text/plain#015#012Server: kamailio (5.4.4 (x86_64/linux))#015#012Content-Length: 9#015#012#015#012Not Found> ERROR: <core> [core/msg_translator.c:3235]: build_sip_msg_from_buf(): parsing failed now change to Native Scripting Routing: listen=tcp:127.0.0.1:5060 tcp_accept_no_cl=yes loadmodule "xhttp.so" event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); xhttp_reply("404", "Not Found", "text/plain", "Not Found"); exit; } curl http://127.0.0.1:5060 -vvv everything is ok. here is the Navite config file: http://220.231.180.131:12296/kam/native/ here is the Lua config file: http://220.231.180.131:12296/kam/kemi/ hanxiaofang

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sanity_reply not available from sanity module
by Patrick Karton 04 May '22

04 May '22

Hello , the sanity_reply function from sanity is not available in kamailio script so i get an error when i want to use it. in the module source i see this static cmd_export_t cmds[] = { {"sanity_check", (cmd_function)w_sanity_check, 0, 0, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"sanity_check", (cmd_function)w_sanity_check, 1, fixup_igp_null, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"sanity_check", (cmd_function)w_sanity_check, 2, fixup_igp_igp, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"sanity_check", (cmd_function)w_sanity_reply, 0, 0, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"bind_sanity", (cmd_function)bind_sanity, 0, 0, 0, 0 }, {0, 0, 0, 0, 0, 0} }; why is sanity_check linked again in w_sanity_reply ? is this an error ? i want to setup specific responses for some errors and let the default ones from module work as usual. Thanks.

2 2

FAILEDOVER ROUTING WITH DNS SRV Records
by Patrick Karton 04 May '22

04 May '22

Hello, im using kamaliio 5.5.3 and I want to do failedover routing with DNS SRV Records. in my lab i have these SRV records ;; ANSWER SECTION: voip.oov-obs-tools-dev.svc.cluster.local. 5 IN SRV 0 33 5060 100-68-7-74.voip.oov-obs-tools-dev.svc.cluster.local. voip.oov-obs-tools-dev.svc.cluster.local. 5 IN SRV 0 33 5060 100-68-7-75.voip.oov-obs-tools-dev.svc.cluster.local. voip.oov-obs-tools-dev.svc.cluster.local. 5 IN SRV 0 33 5060 100-68-7-76.voip.oov-obs-tools-dev.svc.cluster.local. ;; ADDITIONAL SECTION: 100-68-7-76.voip.oov-obs-tools-dev.svc.cluster.local. 5 IN A 100.68.7.76 100-68-7-75.voip.oov-obs-tools-dev.svc.cluster.local. 5 IN A 100.68.7.75 100-68-7-74.voip.oov-obs-tools-dev.svc.cluster.local. 5 IN A 100.68.7.74 my config script use_dns_cache=on use_dns_failover=on dns_srv_lb=on $du = "voip.oov-obs-tools-dev.svc.cluster.local"; t_relay(); after t_relay() is called the INVITE is send to one the 3 ips corresponding to fqdn but if the first ip does not send response. the two others ips are not tried. what i want is that all the ips resolved are tried until one succeeds. is there something to do to achieve that ? do i need to interact with failure route to try next ips ? Thanks for help. version: kamailio 5.5.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1

2 1

function 'sl_send_reply' is not available
by David Villasmil 04 May '22

04 May '22

Hello guys, I'm starting to try KEMI and i already hit a snag... right now i'm simply testing with: loadmodule "app_lua.so" modparam("app_lua", "load", "/etc/kamailio/myscript.lua") cfgengine "lua" route { lua_run("send_reply"); } But i'm always getting: 0(118) ERROR: app_lua [app_lua_api.c:1633]: sr_kemi_lua_modf(): function 'sl_send_reply' is not available I'm guessing that function is not exported by the module.. if that's the case, how can i use it? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Dispatcher SRV load balancing
by Nicolas Chapleau 03 May '22

03 May '22

Just having a hard time trying to see if the Dispatcher module does LB when SRV records are used as the destination. Will the destinations be picked based on the SRV list or will it simply use the first name received? Read lots of docs, but can't seem to figure it out Thanks, Nic.

1 0

Problem with Record-Route in Public to Private Network
by Carlos Escalona 03 May '22

03 May '22

Hello! I wish everyone who reads this message a good day. I would appreciate some guidance with the configuration of my Kamailio. Initially, I have the following architecture: Endpoints <----Public IP----> Kamailio (listen PrivateIP advertise PublicIP) <----Private IP----> Asterisk CONFIG: https://gist.github.com/Carlos-Escalona94/3a289af05b05c69ae563ab29c9ec4710 In general, calls initiated from outside the system work perfectly, but I have a problem with calls initiated from Asterisk. According to what I was able to investigate and understand, the problem arises from the fact that Kamailio uses the public IP for the entire routing system that involves the Record-Route and Route Headers. This in turn causes Asterisk to get lost trying to send messages outside of the INITIAL INVITE transaction, for example, the ACK response to the 200 Ok received from the Endpoint since it tries to use Kamailio's public IP to which Asterisk doesn't have access. I tried to force Asterisk to send all messages to Kamailio's private IP regardless of their nature, but it doesn't seem like an appropriate solution. On the other hand, I tried to modify the architecture a bit so that it was something similar to this: Endpoints <---- Public IP----> Kamailio (listen PrivateIP:P1 advertise PublicIP) <---- PrivateIP-----> Kamailio (listen PrivateIP:P2) <---- PrivateIP ----> Asterisk CONFIG: https://gist.github.com/Carlos-Escalona94/4d681bb189c6190941d291965e123889 It seems to me that this would solve the problem, but I have two doubts about this architecture, the first is that I would like to know if there is an easier way to solve the problem that I am not considering, and on the other hand, I have not found a way to identify from which interface is receiving the message to perform the routing properly. Thanks for the attention. -- *Esta mensagem pode conter informação confidencial ou privilegiada, sendo seu sigilo protegido por lei. Se você não for o destinatário ou a pessoa autorizada a receber esta mensagem, não pode usar, copiar ou divulgar as informações nela contidas ou tomar qualquer ação baseada nessas informações. Se você recebeu esta mensagem por engano, por favor, avise imediatamente ao remetente, respondendo o e-mail e em seguida apague-a. Agradecemos sua cooperação. ** * *This message may contain confidential or privileged information and its confidentiality is protected by law. If you are not the addressed or authorized person to receive this message, you must not use, copy, disclose or take any action based on it or any information herein. If you have received this message by mistake, please advise the sender immediately by replying the e-mail and then deleting it. Thank you for your cooperation. *

3 2

Getting "uac.reg_add" to work (documentation unclarities)
by Rick van Rein 29 Apr '22

29 Apr '22

Hello, I am trying to make UAC register with a SIP telco. But when I run... shell$ kamctl rpc uac.reg_add mydomain.com 0123456789 myteldomain.com 0123456789 sip6.telco.nl telcoreseller.nl 999999 secretpw . udp:sip6.telco.nl 360 0 10 'udp:[fd8:9::10]:5090' ...it gives me the cryptic response... ERROR: jsonrpcs [jsonrpcs_mod.c:666]: jsonrpc_scan(): field is not a string - type 3 The module docs only give "..." as example of attributes. And, according to... http://www.kamailio.net/docs/modules/5.4.x/modules/uac.html#uac.r.uac.reg_a… https://github.com/kamailio/kamailio/blob/5.4.4/src/modules/uac/uac_reg.c#1… ...it also seems to be behind on diff_expires, timer_expires, reg_init. What I am missing is the format, and in some cases the purpose of attributes. Could you help me to an example, perhaps based on the command I am trying? Thanks, -Rick P.S. It's been years since I worked with OpenSIPS. It feels like Kamailio is friendlier, and easier to use.

3 4

Dispatcher Usage and change of $rU
by Benoît Panizzon 29 Apr '22

29 Apr '22

Hi Gang I have a observation which I don't understand... Let's assume we have an inbound $rU = 12345 And I want to change $rU outbound to 0012345 Doing: $rU = "0012345"; ds_select_dst("2000"),"6"); route(RELAY) The call is being sent out with unaltered R-URI Username: 12345 Doing: $rU = "0012345"; ds_select_domain("2000", "6"); route(RELAY); The call is being sent out with the intended new R-URI Username: 0012345 Looking at the docs: https://kamailio.org/docs/modules/5.4.x/modules/dispatcher.html#dispatcher.… I can not figure out, why I see this difference. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 3

Incorrect Socket and Tcpconn-Id with Mid-Registrar and WebSockets
by Grant Bagdasarian 29 Apr '22

29 Apr '22

Hi, We're attempting to implement a mid-registrar for websocket connections but are running into some issues where the tcpconn-id and socket attributes are not stored correctly in memory (usrloc). REGISTER Flow: UA1 -> P1 (Mid-Registrar) ~> P2 (Core Proxy/Registrar) INVITE Flow: UA2 -> P2 -> P1 -> UA1 UA1: baresip cli connecting to P1 using websockets UA2: softphone Everything is running locally on a dev machine, and we're using Docker to launch P1 and P2. The below code snippet is what we use on P1, and has been simplified. The general idea is as following: 1) UA1 sets up the websocket connection and sends a REGISTER request to P1 2) P1 checks if this request has already been sent to P2 2.a) If not, then relay the request to P2. Upon receiving a 200 OK from P2 we save the registration in usrloc 2.b) If so, then absorb the request (save) and generate a reply 3) UA2 tries to call UA1 via P1 and P2 The problem occurs at 2.a since the REGISTER request is forwarded using UDP and the 200 OK is also received using UDP. When the registration is stored in reply route, the Socket (udp:x.x.x.x:xxxx) and Tcpconn-Id (-1) are not set to what we would prefer (see Output 1) TCP was also tried as the transport between P1 and P2. Upon receiving the reply from P2 the Tcpconn-Id and Socket of the connection between P1 and P2 was saved in the binding. When 2.b occurs the Socket and Tcpconn-Id are set to the correct values (see Output 2) Making a call when the binding state is at 2.a we receive the following error: "TCP/TLS connection (id: 0) for WebSocket could not be found" Making a call when the binding state is at 2.b works perfectly fine since the correct Tcpconn-Id is set. We did find a solution, or rather a workaround to this issue: 1) Always call the save function in request_route 2) If P2 responds with an error (non-200 OK) use the unregister function to remove the binding We also tried the tcpops tcp_set_otcpid and tcp_set_otcpid_flag functions after lookup, but this didn't do anything. KEMI Script def ksr_request_route(self, msg): if kemi.is_REGISTER(): aor = "aor" if result := kemi.htable.sht_is_null("registrations", aor) == 1: kemi.tm.t_on_reply("ksr_register_reply_route") kemi.dispatcher.ds_select_dst(1, 4) kemi.tm.t_relay() return 1 kemi.registrar.save("registrations", 1) return 1 if kemi.is_INVITE(): aor = "aor" if result := kemi.registrar.lookup_uri("registrations", aor): kemi.tm.t_relay() return 1 def ksr_register_reply_route(self, msg): if kemi.tm.t_check_status("200") == 1: kemi.registrar.save("registrations", 1) aor = "aor" kemi.htable.sht_sets("registrations", aor, "") return 1 Output 1: Binding state when request is being forwarded kamcmd ul.dump { Domains: { Domain: { Domain: registrations Size: 1024 AoRs: { Info: { AoR: grant(a)127.0.0.1 HashID: 409622189 Contacts: { Contact: { Address: sip:grant-0x7fe915a79138@172.16.99.12:9;transport=ws Expires: 13 Q: 1.000000 Call-ID: 7cdfbf93f23bb54f CSeq: 34818 User-Agent: n/a Received: sip:192.168.80.1:62184;transport=ws Path: [not set] State: CS_NEW Flags: 1 CFlags: 20 Socket: udp:10.2.0.15:5060 Methods: -1 Ruid: uloc-626a80d3-46-1 Instance: <urn:uuid:c67df38d-d5e0-e834-1d98-a8df2d3f291b> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 0 Last-Keepalive: 1651146965 KA-Roundtrip: 0 Last-Modified: 1651146965 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } kamcmd core.tcp_list { id: 1 type: WS state: CONN_OK timeout: 2 lifetime: 120 ref_count: 2 src_ip: 192.168.80.1 src_port: 62184 dst_ip: 192.168.80.15 dst_port: 80 } kamcmd ws.dump { connections: { 1: ws:192.168.80.1:62184 -> ws:192.168.80.15:80 (state: OPEN, last used 3s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } Output 2: Binding state when request is being absorbed kamcmd ul.dump { Domains: { Domain: { Domain: registrations Size: 1024 AoRs: { Info: { AoR: grant(a)127.0.0.1 HashID: 409622189 Contacts: { Contact: { Address: sip:grant-0x7fe915a79138@172.16.99.12:9;transport=ws Expires: 9 Q: 1.000000 Call-ID: 7cdfbf93f23bb54f CSeq: 34819 User-Agent: baresip v2.0.2 (x86_64/darwin) Received: sip:192.168.80.1:62184;transport=ws Path: [not set] State: CS_NEW Flags: 1 CFlags: 20 Socket: tcp:192.168.80.15:80 Methods: 5087 Ruid: uloc-626a80d3-46-1 Instance: <urn:uuid:c67df38d-d5e0-e834-1d98-a8df2d3f291b> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 1 Keepalive: 0 Last-Keepalive: 1651146979 KA-Roundtrip: 0 Last-Modified: 1651146979 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } kamcmd core.tcp_list { id: 1 type: WS state: CONN_OK timeout: 3 lifetime: 120 ref_count: 2 src_ip: 192.168.80.1 src_port: 62184 dst_ip: 192.168.80.15 dst_port: 80 } kamcmd ws.dump { connections: { 1: ws:192.168.80.1:62184 -> ws:192.168.80.15:80 (state: OPEN, last used 1s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } }

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TOPOS, 183 (before a 200OK) and SIP INFO
by Giovanni Tommasini 28 Apr '22

28 Apr '22

Hi kamailiers, I ask your help for a strange situation with topos, 183 (before a 200OK) and SIP INFO My topology is kamailio in front of asterisks; so kamailio has 2 interfaces (1Public to talk with customers and carriers and 1 Private to talk with asterisk). In the image: - the first column is the customer - the second is the public interface of kamailio - the third is the private interface of kamaiio - the fourth is asterisk As you can see in the image, the customer send an INVITE and the carrier give back the IVR messages in a 183 (before a 200OK) asking for an IVR selection. The customer (configured to use SIP INFO) types a key and sends in INFO, but as you can see kamailio sends the INFO to himself to private interface instead of sending it to asterisk as if the contact that handled topos contained incorrect information. [image: Schermata 2022-04-26 alle 19.40.32.png] - If the customer is configured with a RFC2833 - or if the IVR messages is not in a 183 but is sent after a 200OK - or if I disabe TOPOS *all works fine* I used kamailio 5.5.3, I also try to upgrade to 5.5.4 but nothing change Maybe could be connected with https://github.com/kamailio/kamailio/issues/2344 https://github.com/kamailio/kamailio/issues/1077 Could be a kamailio bug for this particular situation or am I getting lost in a glass of water? thanks a lot cheers Giovanni Tommasini *evoseed s.r.l.* phone: +39 340 11 28 800 mail: giovanni.tommasini(a)evoseed.io site: evoseed.io address: Via Lucrezio 13, Trieste Italy

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Kamailio functionality clarification - call forwarding no replay (call is ringing but not answered)
by Pavel Veselovsky 28 Apr '22

28 Apr '22

Hallo all, I would need help with Kamailio. I hope, you can give me some hints. I would like use Kamailio for CFNRy feature call forward no replay, i.e. - call is forked to multiple destination - 180 is received, call is ringing, but not answered, no 200 OK is received from destinations - after predefined time (CFNRy timer), the specific branch has to be rerouted to another destination Expected behaviour is described in this flow: [image: image.png] My current behavior: - Kamailio TM module is involved for parallel forking ( https://www.kamailio.org/docs/modules/stable/modules/tm.html#:~:text=1.-,Ov… ) - all branches are CANCEL-ed, it is not possible send CANCEL just to one specific branch [image: image.png] My questions: - is it possible to CANCEL just specific branch of parallel forked calls? - is it possible start timer just for specific branch? - do you think is it possible implement this feature with Kamailio? Thanks a lot in advance for any comments. Pavel.

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t_check_trans failed on ACK
by Linux Vince 28 Apr '22

28 Apr '22

I am using Kamailio as registrar and proxy for underlying asterisk server. I am able to call but ACK on 200 OK respond from client phones are not validating transaction and thus not relaying correctly. Below are my Invites and 200 OK responses. Invite from Asterisk to Kamailio INVITE sip:31313004@rigel.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com:5060> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 INVITE Contact: <sip:65.20.69.72:5060> max-forwards: 70 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 228 v=0 o=Clarent 155866 155867 IN IP4 65.20.69.72 s=Clarent CallManager c=IN IP4 65.20.69.72 t=0 0 m=audio 45168 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Invite from Kamailio to Phone INVITE sip:31313004@65.20.67.118;lhst=180.211.119.50:25759;lm=midreg SIP/2.0 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Via: SIP/2.0/UDP 65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;received=65.20.69.72;rport=5060;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com:5060> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 INVITE Contact: <sip:65.20.69.72:5060> max-forwards: 69 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 228 v=0 o=Clarent 155866 155867 IN IP4 65.20.69.72 s=Clarent CallManager c=IN IP4 65.20.69.72 t=0 0 m=audio 45168 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 200 OK from Phone to Kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP 65.20.67.118;received=65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 From: "R31313005 N31313005" <sip:31313005@65.20.69.72 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com >;tag=4ecc1f21f1544fd783965f27ff6d95d2 CSeq: 1 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:31313004@180.211.119.50:57935;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 322 v=0 o=- 3860150297 3860150298 IN IP4 180.211.119.50 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4016 RTP/AVP 0 101 c=IN IP4 180.211.119.50 b=TIAS:64000 a=rtcp:4017 IN IP4 180.211.119.50 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1461220511 cname:3b657c275d2a43f6 200 OK from kamailio to asterisk SIP/2.0 200 OK Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 From: "R31313005 N31313005" <sip:31313005@65.20.69.72 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com >;tag=4ecc1f21f1544fd783965f27ff6d95d2 CSeq: 1 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:31313004@180.211.119.50:57935;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 322 v=0 o=- 3860150297 3860150298 IN IP4 180.211.119.50 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4016 RTP/AVP 0 101 c=IN IP4 180.211.119.50 b=TIAS:64000 a=rtcp:4017 IN IP4 180.211.119.50 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1461220511 cname:3b657c275d2a43f6 ACK from phone to Kamailio ACK sip:31313004@180.211.119.50:57935;ob SIP/2.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport;branch=z9hG4bK111270de2729ade9e048afa0f9fbb8df Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com >;tag=4ecc1f21f1544fd783965f27ff6d95d2 Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 ACK Contact: <sip:65.20.69.72:5060> max-forwards: 70 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE This ACK is failing on transaction check and not routing correctly to phone. Any help? Thanks

2 3

Modify session name in SDP
by ZZ Wave 28 Apr '22

28 Apr '22

Hello. Kamailio 5.2.1, Debian 10. I need to change session name (s=) in SDP before relaying it to next hop, and for unknown reason I can't do it with virtually any TEXTOPS function, everything replace*, subst*, *body*... For example this from TEXTOPS documentation: subst_body('/^s=(.*)/s=root/') just before 'final' t_relay(). Either Kamailio just adds "s=root" to SDP leaving first "s=" intact, or adds just "root", or does nothing. Also I've tried msg_apply_changes() before and after these manipulations, still no result. Is it possible to make this in Kamailio without source code modification?

1 0

High performance
by Muborakshohi Masoud 27 Apr '22

27 Apr '22

Hello! I have kamailio 5.5 configured with digest authentication. DB Mysql, 2 rtpengine one on the same server with kamailio, the other is separate. Rtpengins work in bridge mode, internal/external/external-ipV6. Kamailio listen 3 address: internal/external/external-ipV6 work in mixed mode: ipV4-ipV4, ipV4-ipV6, ipV6-ipV6. OS Debian 10, CPU 1 core, Ram 4gb, Disk not ssd. internal address is used for the gateway model of the connection to the PSTN. Before sending a call to PSTN, a request is sent to Billing by Diameter (request over the network, ims_charging ), Billing responds in the region of 10-17 milliseconds. usrloc db_mode = 0; UAS contacts are received according to the PUSH scheme (we send PUSH via http and get a REGISTER). children = 8 for each; When running synthetic tests with sipp, I get 36 cps as a maximum, but there are a lot of retrans. This is a test environment. Product: I plan to get 100 cps; 20k concurrent calls, hardware will be 128GB RAM ddr4-2400; Intel Xeon Silver 4216, Optane SSD. 2 such servers, kamailio/rtpengine are configured exactly according to the scheme of the test environment. Question: Can you please tell me how many children, shm/pv do I need? Is it worth going in the direction of asynchrony, or is the standard one enough for my purpose? I will be very glad for any help. Thank you! -- Отправлено из Mail.ru для Android

2 1

Intermittent error seen when using a python script in kamailio
by Wojciech Kopciewicz 27 Apr '22

27 Apr '22

Hi, Intermittently we are observing the following error when checking if given header is present in SIP INVITE: SystemError: returned a result with an error set The complete stack trace is (some names redacted): 11(28) ERROR: {1 195 INVITE nMHZDN.10nOIDt4ldOwz5cqz0XnMepA9} app_python3 [python_support.c:156]: python_handle_exception(): apy_exec: modify_headers((null)): Unhandled exception in the Python code: TypeError: an integer is required (got type NoneType) The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 25, in modify_headers self.modify_header("SomeHeader") File "/etc/kamailio/kamailio.py", line 29, in modify_header present = KSR.hdr.is_present(header_name) SystemError: returned a result with an error set Kamailio version (kamailio -v output): version: kamailio 5.4.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 Operation system: Linux kamailio-5c7484c74d-vhhr8 5.4.0-1056-azure #58~18.04.1-Ubuntu SMP Wed Jul 28 23:14:18 UTC 2021 x86_64 GNU/Linux Are you able to help me find the root cause of this issue? Pozdrawiam / Best regards, Wojtek -- *For more information on how and why we collect your personal information, please visit our Privacy Policy <https://www.motorolasolutions.com/en_us/about/privacy-policy.html?elqTrackI…>.*

3 9

LCR issues (request_uri)
by Antoine DEBARRE 26 Apr '22

26 Apr '22

Hi, I'm trying to use LCR with kamailio v5.5.4 gw, rule and rule_target are stored on mysql. for lcr rules, do you know if prefix is mandatory ? I first wanted to start using request_uri instead and leave the prefix column null/empty because prefix seems not to support regexp, I don't want to add the same rule with different prefixes. but I'm facing issues my database is fulfilled with : lcr_gw +----+--------+--------------+---------------+----------+------+--------+------------+-----------+-------+--------+-------------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+--------------+---------------+----------+------+--------+------------+-----------+-------+--------+-------------+-------+---------+ | 1 | 1 | carrier1 | 10.78.51.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier1 | 0 | NULL | | 2 | 1 | carrier2 | 10.78.51.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier2 | 0 | NULL | | 3 | 1 | carrier3 | 10.192.50.212 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 6 | 1 | carrier4 | 10.192.50.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier4 | 0 | NULL | | 10 | 1 | carrier5 | 10.192.50.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier5 | 0 | NULL | | 18 | 1 | carrier6 | 10.192.50.214 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 20 | 1 | carrier7 | 10.192.50.210 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 30 | 1 | carrier8 | 10.192.50.211 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 31 | 1 | carrier9 | 10.192.50.216 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 32 | 2 | media1 | 10.192.50.232 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 33 | 2 | media2 | 10.192.50.233 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 34 | 2 | media3 | 10.192.50.234 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 35 | 2 | media4 | 10.192.50.235 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 36 | 2 | media5 | 10.192.50.240 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 37 | 2 | media6 | 10.192.50.241 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 38 | 2 | media7 | 10.192.50.242 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 39 | 3 | xivo-231 | 10.192.50.231 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | cps=5;cc=50 | 0 | NULL | +----+--------+--------------+---------------+----------+------+--------+------------+-----------+-------+--------+-------------+-------+---------+ lcr_rule +----+------------------------+-----------------------------------+------------------+--------+----------+-----------+---------+---------+ | id | desc | request_uri | lcr_id | prefix | from_uri | mt_tvalue | stopper | enabled | +----+------------------------+-----------------------------------+------------------+--------+----------+-----------+---------+---------+ | 1 | Default | .+ | 1 | NULL | NULL | NULL | 0 | 1 | | 2 | France GSM | (0|(|00|\+)33)[67]\d{8} | 1 | NULL | NULL | NULL | 1 | 1 | | 3 | France | (0|(|00|\+)33)[1-59]\d{8} | 1 | NULL | NULL | NULL | 1 | 1 | | 4 | International default | (00|\+)[1-9].+ | 1 | NULL | NULL | NULL | 1 | 1 | | 5 | Italy | (00|\+)39.+ | 1 | NULL | NULL | NULL | 1 | 1 | | 14 | to_oldmedias | .+ | 2 | NULL | NULL | NULL | 0 | 1 | +----+------------------------+-----------------------------------+-------------------+--------+----------+-----------+---------+---------+ lcr_rule_target +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 25 | 1 | 3 | 3 | 1 | 1 | | 26 | 1 | 3 | 18 | 2 | 1 | | 28 | 1 | 3 | 30 | 3 | 1 | | 29 | 1 | 2 | 30 | 4 | 1 | | 30 | 1 | 2 | 3 | 2 | 1 | | 31 | 1 | 2 | 18 | 3 | 1 | | 32 | 1 | 2 | 20 | 1 | 1 | | 33 | 1 | 1 | 30 | 1 | 1 | | 34 | 1 | 12 | 1 | 1 | 1 | | 35 | 1 | 12 | 10 | 1 | 1 | | 37 | 2 | 14 | 32 | 1 | 1 | | 38 | 2 | 14 | 33 | 1 | 1 | | 39 | 2 | 14 | 34 | 1 | 1 | | 40 | 2 | 14 | 35 | 1 | 1 | | 41 | 2 | 14 | 36 | 1 | 1 | | 42 | 2 | 14 | 37 | 1 | 1 | | 43 | 2 | 14 | 38 | 1 | 1 | | 45 | 1 | 2 | 10 | 3 | 1 | +----+--------+---------+-------+----------+--------+ to test gws, I use kamcmd for example : kamcmd lcr.load_gws 1 s:0600000001 s:02111111111 s:0600000001 arg 1 : prefix or $rU arg 2 : from_uri arg 3 : request_uri result is OK with 5 gateways selected but in real test situation with the following config file : kamailio.cfg : request_route { ... if(!load_gws(1,$rU,$fU)) { sl_send_reply("500", "Cannot load gateways"); exit; } ## - dumping the list of lcr gateways $var(i) = 0; while( $(avp(lcr_gw_uri)[$var(i)]) != $null) { xlog("L_ERR","loaded gw_uri_avp[$var(i)]=’$(avp(lcr_gw_uri)[$var(i)])’\n"); $var(i) = $var(i) + 1; } if(!next_gw()) { sl_send_reply("503", "No gateways found"); exit; } } ..... => we receive a 503 "No gateways found", which indicate that load_gws() is OK, but wasn't able to parse avp lcr_gw_uri (we have no xlog output) Then I noticed that some calls are OK and others are KO (with same pattern for $rU) => lcr rule id 2 So I with debug=3 *call to 0611111111 => load_gws OK and next_gw OK* lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3004]: do_from_gw(): request game from gw lcr [lcr_mod.c:3009]: do_from_gw(): added tag_avp <cps=5;cc=50> lcr [lcr_mod.c:2264]: ki_load_gws_furi(): load_gws(1, 0611111111, 0211111111) lcr [lcr_mod.c:2313]: ki_load_gws_furi(): from uri <0211111111> did not match to from regex <AMD> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0611111111@10.192.50.227:5060> did not match to request regex <(00|\+)39.+> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0611111111@10.192.50.227:5060> did not match to request regex <(00|\+)[1-9].+> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0611111111@10.192.50.227:5060> did not match to request regex <(0|(|00|\+)33)[1-59]\d{8}> lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[0]=[4, 0, 3, 2329862] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[1]=[5, 0, 1, 5553909] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[2]=[8, 0, 3, 7256660] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[3]=[7, 0, 2, 3129551] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[4]=[6, 0, 4, 2234577] lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <6|sip:|0|||3543318538||5060||;transport=udp|0|2> with weight <2234577> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <4|sip:|0||fd=carrier5|53657610||5060||;transport=udp|0|2> with weight <2329862> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <8|sip:|0|||3593650186||5060||;transport=udp|0|2> with weight <7256660> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <7|sip:|0|||3560095754||5060||;transport=udp|0|2> with weight <3129551> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <5|sip:|0|||3526541322||5060||;transport=udp|0|2> with weight <5553909> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.210:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2904]: ki_next_gw(): added ruri_user_avp <0611111111> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3526541322> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.212:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3560095754> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.214:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3593650186> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.3:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <fd=carrier5> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <53657610> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.211:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3543318538> the right gws are selected and call are routed to them *call to 0600000001 => load_gws OK and next_gw KO* lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3004]: do_from_gw(): request game from gw lcr [lcr_mod.c:3009]: do_from_gw(): added tag_avp <cps=5;cc=50> lcr [lcr_mod.c:2264]: ki_load_gws_furi(): load_gws(1, 0600000001, 0211111111) lcr [lcr_mod.c:2313]: ki_load_gws_furi(): from uri <0170619309> did not match to from regex <AMD> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0600000001@10.192.50.227:5060> did not match to request regex <(00|\+)39.+> the process stops after trying to match rule with regexp <(00|\+)39.+> Don't know if it's a bug or if I misunderstood something ! Thanks in advance ! Best regards, -- <https://app.livestorm.co/comdata-group/metaverse-what-will-cx-look-like-in-…>

3 8

Local Registration Cache
by Linux Vince 26 Apr '22

26 Apr '22

Hello, I am trying to implement scenario described in this thread by Daniel-Constantin Mierla. I am able to cache registration using save but i am not able to handle unregistration. Unregister function of registrar module is not available in onreply routes and I cannot unregister until upstream registrar returns 200 OK on Expire 0 message. Is there any workaround to fix this? Thanks,

3 5

SSL errors with websocket clients
by Vinicius Kwiecien Ruoso 22 Apr '22

22 Apr '22

Hi all! How are you? I have a kamailio instance and some users are having issues connecting via websocket using TLS. The logs show SSLv3 errors. Cannot find why that error would show up if SSLv2/3 is not enabled. Double checked it via SSLLabs that only TLSv1.2 is allowed in the service. Error sample: ``` 15(36) ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown 15(36) ERROR: <core> [core/tcp_read.c:1512]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fafc8768190 r: 0x7fafc8768278 (-1) ``` This instance has close to 400 websocket connections opened during the day. I have a high log level enabled. I was not able to identify any reason why this happens. The certificate is a Letsencrypt certificate. The issue is that sometimes clients have a hard time connecting to the websocket and that maybe related. Below are a few more details about the config. Please let me know if you have any pointers on how to debug those connection issues. 8<----- TLS config: modparam("tls", "tls_method", "TLSv1.2+") modparam("tls", "verify_certificate", 0) modparam("tls", "require_certificate", 0) modparam("tls", "low_mem_threshold1", 0) modparam("tls", "low_mem_threshold2", 0) modparam("tls", "private_key", "/etc/certs/tls.key") modparam("tls", "certificate", "/etc/certs/tls.crt") version: kamailio 5.3.9 (x86_64/linux) 8<----- This is related to the issue https://github.com/kamailio/kamailio/issues/3085 (as pointed out it is not a problem in Kamailio code). Thanks, Vinicius

2 2

How to disable video in invite with both audio and video
by Simone de Blasiis 22 Apr '22

22 Apr '22

Hello to everyone, we have a Kamailio 5.5 + RTPEngine deployment. Some devices will send an invite with both audio and video in the offer and we are trying to disable the video portion of the invite by either setting the video port to 0 or by setting a=inactive in the video portion of the SDP packet. We know that sdpops has a function that would be able to remove the video portion but it violates standards so we can't use it. We need to disable the video in the SDP offers because one of the pstn providers we are using doesn't accept invites that contain a video offer. I've spent days searching but I can't find an obvious way of achieving this. We were originally doing it with textops but we've been having some issues where the sdp wouldn't be modified correctly in reinvites after the original call is in place so we're looking for alternatives. Can someone please help us? Thanks in advance Simone de Blasiis

3 4

DMQ sync issue between 5.4 and 5.5
by David Villasmil 20 Apr '22

20 Apr '22

Hello guys, We have a cluster of about 10 kamilios 5.4 with several million htable records. Restarting any server is not a problem, CPU goes ip for a second but everything comes back down to normal. We added a 5.5 by mistake and it won't start, it'll just eat up memory until it bails out with qm_malloc(): qm_malloc(0x149dd5ebb000, 16016) called from tm: h_table.c: build_cell(334), module: tm; Free fragment not found! Is there any known issue between DMQ 5.4 and 5.5? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 1

"Re: TOPOS (5.1.[23]) breaks PRACK"
by Simone de Blasiis 20 Apr '22

20 Apr '22

Hello, we noticed the exact behaviour on Kamailio 5.5.4 and topos. If we disable topos the UPDATE messages are being recognized as loose in the WITHINDLG route and everything works. When topos is enabled the UPDATE messages fail the loose_route() check and even if we force the record_route() the message is still not being forwarded. I had just posted my own thread here <https://lists.kamailio.org/pipermail/sr-users/2022-April/114710.html> because I didn't see this one. I'll proceed opening an issue on GitHub so we can track this. Best regards Simone

1 1

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Simone de Blasiis 20 Apr '22

20 Apr '22

Hello, we noticed the exact behaviour on Kamailio 5.5.4 and topos. If we disable topos the UPDATE messages are being recognized as loose in the WITHINDLG route and everything works. When topos is enabled the UPDATE messages fail the loose_route() check and even if we force the record_route() the message is still not being forwarded. I had just posted my own thread here <https://lists.kamailio.org/pipermail/sr-users/2022-April/114710.html> because I didn't see this one. I'll proceed opening an issue on GitHub so we can track this. Best regards Simone

1 0

Problems with topos and UPDATE messages
by Simone de Blasiis 20 Apr '22

20 Apr '22

Hello to everyone, we are experiencing some issues with UPDATE messages being sent by the PSTN after an early media announcement. To reproduce the issue I have to call a specific number that has an early media announcement, since it's a paid phone number it has to announce how much it would cost before picking up the call. In this initial part the call is being negotiated in g711a. As soon as the announcement ends, the PSTN sends an UPDATE message containing the new desidered codec from the callee which is g729. At this moment Kamailio doesn't route the UPDATE message to the destination and it actually replies with a 404 message. I've attached a pcap with anonymized IPs to make it harder for bots. Here is a legend for the IPs: - - 12.34.123.123 is Kamailio - 62.23.19.74 is the PSTN - 1.2.130.40 is the client making the call I found a similar report <https://github.com/kamailio/kamailio/issues/2659> in the GitHub issues of the repository and we tried forcing the record_route() as suggested but, as the OP reported in his last message, it's not working. As soon as we disable topos, the UPDATE message is correctly relayed to the caller. In the documentation we can't find anything that could give us a hint on what's happening. Does anyone have any idea on what could be happening? Thanks in advance Simone

1 0

rcd using secspid
by Jon Bonilla (Manwe) 20 Apr '22

20 Apr '22

Hi all I've been checking for adding rcd info to te stir/shaken stuff using secspid module. I've only found this thread: https://lists.kamailio.org/pipermail/sr-dev/2021-June/063156.html I'd like to know if someone has working code snippet using this function to create/sign the json containing the rcd and creating the Identity header instead of calling the all_in_one secsipid_add_identity function I'm using now. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

PostgreSQL - module sqlops - working with prepared statements ?
by Chaigneau, Nicolas 20 Apr '22

20 Apr '22

Hello, I'm using sql_xquery from sqlops module to execute requests on a PostgreSQL database. This work fine. Now, for performance reasons, I would like to use prepared statements. I'm wondering if this is possible using sql_xquery. I've looked at the code and documentation, it seems it is not supported. Can you confirm ? If not possible with sql_xquery, is it possible with another existing function ? In the code, I've searched for < PQprepare > / < PQexecPrepared >. These can be found only in < modules/db_postgres/pg_cmd.c >, in function < pg_cmd_exec > which is exported from < modules/db_postgres/pg_mod.c > (see below). I have no idea how to use these though... :/ Any ideas ? /* * Postgres module interface */ static cmd_export_t cmds[] = { {"db_ctx", (cmd_function)NULL, 0, 0, 0, 0}, {"db_con", (cmd_function)pg_con, 0, 0, 0, 0}, {"db_uri", (cmd_function)pg_uri, 0, 0, 0, 0}, {"db_cmd", (cmd_function)pg_cmd, 0, 0, 0, 0}, {"db_put", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_del", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_get", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_upd", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_sql", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_res", (cmd_function)pg_res, 0, 0, 0, 0}, {"db_fld", (cmd_function)pg_fld, 0, 0, 0, 0}, {"db_first", (cmd_function)pg_cmd_first, 0, 0, 0, 0}, {"db_next", (cmd_function)pg_cmd_next, 0, 0, 0, 0}, {"db_setopt", (cmd_function)pg_setopt, 0, 0, 0, 0}, {"db_getopt", (cmd_function)pg_getopt, 0, 0, 0, 0}, {"db_bind_api", (cmd_function)db_postgres_bind_api, 0, 0, 0, 0}, {0, 0, 0, 0, 0, 0} }; Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

3 4

Update: Issue when Kamailio is running as a Docker container and the Docker network=overlay
by Steve Wilkins 17 Apr '22

17 Apr '22

If I don't use 'listen', then packets do arrive on 5060.

1 0

Issue when Kamailio is running as a Docker container and the Docker network=overlay
by Steve Wilkins 16 Apr '22

16 Apr '22

Hello, I am attempting to get Kamailio to run as a docker container using a docker network of type network=overlay (everything works great if network=host). The issue is that incoming calls are rejected because of 'connection refused'. I noticed that when running on an overlay network, netstat shows the following => tcp6 0 0 :::5060 :::* LISTEN 10001/docker-proxy-current (Note that I have tcp6 disabled on the server and must leave it disabled) I was attempting to find a workaround for this issue, but have not been successful. Does any one on the board have experience with Docker overlay networks and Kamailio? If so, I would appreciate any suggestions on how to resolve this issue. Thank you,

1 0

BLF: modules parameters for specific SIP phones
by Liviu Andron 14 Apr '22

14 Apr '22

Hello, I am working on a BLF (Busy Lamp Field, dialog presence) integration using Kamailio. I am not sure how certain settings can be used on a single generic (any phone) Kamailio instance. * https://www.kamailio.org/docs/modules/devel/modules/presence_dialoginfo.htm… "If the phone does not support this, you can activate this parameter." modparam("presence_dialoginfo", "force_single_dialog", 1) * https://kamailio.org/docs/modules/devel/modules/pua_dialoginfo.html#idm152 "This is a workaround for the buggy Linksys SPA962 phones. SNOM phones work well with the default setting." modparam("pua_dialoginfo", "caller_confirmed", 1) How can I activate these settings only for the phones that have problems? Do I need a different Kamailio instance for each buggy phone that appears from a client ? For example, a Kamailio instance with modparam("pua_dialoginfo", "caller_confirmed", 1) for some (like Linksys SPA962) and the opposite for all the others (like SNOM) ? Thank you,

1 0

append new a= in SDP (without rtpengine)
by Lewis Hutchinson 14 Apr '22

14 Apr '22

version: kamailio 5.5.3 Using KEMI Python I have read several posts on this and tried several approaches but can not seem to get it to work. As it stands I am trying to add a=record:on | a=record:off | a=record:pause In the SDP because FreeSWITCH creating the INVITE isn't able to do this in the way required. When it is sent from FreeSWITCH a=record:on is send a=record I have looked at SDPOPS, TEXTOPS and a few others but non of the offer the option to just inject and a= into the SDP body. So as a result I tried a couple of options where the INVITE is created with a=record (unsure why FreeSWITCH wont let us just add :off etc to the end but that's another point that if anyone know would be helpful) and then using replace_body or search_append_body but the result is TEXTOPS is unable to action the request because of a SDP parse error DEBUG: <core> [core/parser/sdp/sdp.c:599]: parse_sdp_session(): ignoring unknown type in a= line: `a=record#015#012a=ptime:20...' Does anyone have any thoughts or tricks on this to give me a few options at ways to handle this? Lewis

1 2

UAC parameter for enabling and disabling remote registration
by Ahmed Fouad 14 Apr '22

14 Apr '22

Hi, I was wondering if this patch was ever merged to the current release(s) https://github.com/kamailio/kamailio/commit/49f354276df3cfc2cdb0f85b5a1839f… I'd like to start Kamailio with all remote registration disabled then enable based on the code logic Thanks in advance,

2 9

Roadmap to the next major release - v5.6.0
by Daniel-Constantin Mierla 14 Apr '22

14 Apr '22

Hello, it is the time to plan a bit the road to the next major release, to be versioned 5.6.0. The 5.5.0 was release about one year ago, therefore it is time to set the milestones for getting 5.6.0 out. I would propose to freeze the development on Thursday, April 14, 2020, test till mid of May or so, then release v5.6.0. There is a lot of development to existing components and a couple of new modules. If anyone wants a different time line towards 5.6.0, let's discuss and choose the one that suits most of the developers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 3

kamailio fail to load sip-router.cfg
by bayu p 14 Apr '22

14 Apr '22

Dear Sir, Could you please help me, please to share how to load sip-router.cfg file inside running kamailio ? the file is necessary to be load during rerouting kamailio gateway. I have 2 platform kamailio , both of them are failed to load sip-router.cfg file. They are apt kamailio installation source and the second is tar.gz kamailio installation source, which reload / restart kamailio process (and installation) were failed to load sip-router.cfg file. Best Regards,

2 5

Re: [SR-Users] Kamailio IMS heavy load
by Henning Westerholt 12 Apr '22

12 Apr '22

Hello, checking CPU/I/O performance of the database during the load test would be a start. You can also monitor database access e.g. with a network debugging tools. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Евгений Мастратов <dgeka25594(a)gmail.com> Sent: Tuesday, April 12, 2022 10:03 AM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio IMS heavy load Hey Henning! Thanks for the advice, I understand about logging, I will make edits and re-check. And how can I check the fact that what are the gaps when reading / writing the database? I am using MySQL. вт, 12 апр. 2022 г. в 10:52, Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>>: Hello, as mentioned, I would start by monitoring my dependencies, in particular related to I/O, like database or monitoring. Just to be sure, you are not running the tests with enabled debug log level? This would certainly cause problems. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of ??????? ????????? Sent: Monday, April 11, 2022 5:23 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Kamailio IMS heavy load Hi! Previously, I created a ticket on github https://github.com/kamailio/kamailio/issues/3071 and tried to deal with the problem there, but they sent me here for advice. I set up Kamailio IMS , in the end everything turned out great, calls are made, and I decided to check how Kamailio copes with the load. I am creating a load using the sipp application. There is no problem with a low call generation rate, but if you raise the call generation to more than 20 calls per second, some of the calls fail. I compared the calls and saw that the ICSCF does not send a 180 Ringing message towards the S-CSCF [..] As I found out further and suggested, the problem is that some calls are cut on the S-CSCF and this is probably due to some kind of timer, but I could not figure out which one. Full logs and dump are attached to the issue on github, below is a log sample by bad dialog 0x7fc133ce61d8 ``` Apr 5 08:24:33 ims-core /usr/local/sbin/kamailio[26061]: DEBUG: ims_dialog [dlg_handlers.c:284]: dlg_iuid_sfree(): freeing dlg iuid [465:10905] (0x7fc133ce61d8) Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1025]: link_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:251]: run_create_callbacks(): dialog=0x7fc133ce61d8 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:789]: dlg_onreq(): dialog [0x7fc133ce61d8] added to tm callbacks Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7fc133ce61d8 changed from state 1 to state 6, due event 4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:1489]: dlg_onreply(): dialog 0x7fc133ce61d8 failed (negative reply) Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 0 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): ref <=0 for dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:196]: destroy_dlg(): destroying dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:207]: destroy_dlg(): removed timer for dlg 0x7fc133ce61d8 [3791:1419] with clid '43-3057631(a)10.2.16.63<mailto:43-3057631@10.2.16.63>' and tags '78885556600' Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=8192 ``` What could be the reasons for this behavior and how to diagnose them? I did not observe problems with MEM or CPU during the tests. I will be grateful for any advice.

1 0

Kamailio IMS heavy load
by Евгений Мастратов 12 Apr '22

12 Apr '22

Hi! Previously, I created a ticket on github https://github.com/kamailio/kamailio/issues/3071 and tried to deal with the problem there, but they sent me here for advice. I set up Kamailio IMS , in the end everything turned out great, calls are made, and I decided to check how Kamailio copes with the load. I am creating a load using the sipp application. There is no problem with a low call generation rate, but if you raise the call generation to more than 20 calls per second, some of the calls fail. I compared the calls and saw that the ICSCF does not send a 180 Ringing message towards the S-CSCF [image: image.png] [image: image.png] As I found out further and suggested, the problem is that some calls are cut on the S-CSCF and this is probably due to some kind of timer, but I could not figure out which one. Full logs and dump are attached to the issue on github, below is a log sample by bad dialog 0x7fc133ce61d8 ``` Apr 5 08:24:33 ims-core /usr/local/sbin/kamailio[26061]: DEBUG: ims_dialog [dlg_handlers.c:284]: dlg_iuid_sfree(): freeing dlg iuid [465:10905] (0x7fc133ce61d8) Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1025]: link_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:251]: run_create_callbacks(): dialog=0x7fc133ce61d8 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:789]: dlg_onreq(): dialog [0x7fc133ce61d8] added to tm callbacks Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7fc133ce61d8 changed from state 1 to state 6, due event 4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:1489]: dlg_onreply(): dialog 0x7fc133ce61d8 failed (negative reply) Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 0 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): ref <=0 for dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:196]: destroy_dlg(): destroying dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:207]: destroy_dlg(): removed timer for dlg 0x7fc133ce61d8 [3791:1419] with clid '43-3057631(a)10.2.16.63' and tags '78885556600' Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=8192 ``` What could be the reasons for this behavior and how to diagnose them? I did not observe problems with MEM or CPU during the tests. I will be grateful for any advice.

2 1

How to submit updates for core documentation?
by Ben Kaufman 11 Apr '22

11 Apr '22

Hello, I had a small patch merged for 5.6 to add preprocessor macros #!trydefenv and #!trydefenvs. (https://github.com/kamailio/kamailio/issues/2967), but they need documentation. I believe that they should ultimately end up on this page: https://www.kamailio.org/wiki/cookbooks/devel/core. I don't see this in the /doc section of the repo. What's the process for me to submit this documentation? Ben Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

2 4

reply sent
by Daniel Paape 11 Apr '22

11 Apr '22

Hello, Is there a function that can be called from a module that will return the last response code sent, regardless of whether it was a local or relayed reply? thanks, Daniel

2 1

LCR from_gw function
by Antoine DEBARRE 11 Apr '22

11 Apr '22

Hi, Kamailio 5.5.4 from deb packages I tried to use the lcr function from_gw(lcr_id[, ip_addr, proto]) I've declared several gws with different lcr_id Unfortunately from_gw doesn't work with lcr_id > 1 For example : if (from_gw(2,$si,1)) { .... } ... error message => ki_from_gw(): invalid lcr_id parameter value 2 Thanks in advance ! -- <https://app.livestorm.co/comdata-group/metaverse-what-will-cx-look-like-in-…>

2 1

Kamailio + Asterisk + Voip SIP Trunk = Incorrect IP on ACK
by Luciano Motti 09 Apr '22

09 Apr '22

Hello! I have the following architecture: - Kamailio with RTPEngine (with public access on IP *YY.YY.42.207*) - Asterisk (with no public access, internal IP *172.31.69.198*) I am using a Voip SIP (with public access on IP *54.XXX.XXX.44*) configured as a Trunk on Kamailio using UAC Module. The issue I am facing is that the SIP messages replies I get from the VOIP provider are being destined to Kamailio's public IP (YY.YY.42.207) instead of Asterisk's IP (as sent on the message). For example, this is the 200 message I sent to the VOIP provider (I clearly state that the contact is Asterisk sip:172.31.69.198:5080): 2022/04/07 14:17:19.260864 172.31.32.7:5060 -> 54.XXX.XXX.44:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 54.XXX.XXX.44:5060;rport=5060;branch=z9hG4bK90b6.f7a6b0a4.0 Record-Route: <sip:YY.YY.42.207;lr;ftag=p0BjH21gpa3rc> Call-ID: 464f21c3-3120-123b-91b2-026ef2fede4c From: "419XXXXX998" <sip:38XXXX02@172.31.32.169>;tag=p0BjH21gpa3rc To: <sip:38XXXX02@54.XXX.XXX.44>;tag=8c3bb09e-a909-4cb4-99a7-3c57aa8f946e CSeq: 50116727 INVITE Server: Asterisk PBX 18.11.0 *Contact: <sip:172.31.69.198:5080 <http://172.31.69.198:5080>>* Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 239 v=0 o=- 1649321756 1649321759 IN IP4 172.31.69.198 s=Asterisk c=IN IP4 172.31.69.198 t=0 0 m=audio 10010 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv But the ACK I get from them is destined to Kamailio instead of Asterisk: 2022/04/07 14:17:19.263469 54.XXX.XXX.44:5060 -> 172.31.32.7:5060 *ACK sip:YY.YY.42.207:5060 SIP/2.0* Route: <sip:YY.YY.42.207;lr;ftag=p0BjH21gpa3rc> Via: SIP/2.0/UDP 54.XXX.XXX.44:5060;branch=z9hG4bK90b6.f7a6b0a4.2 Max-Forwards: 69 From: "419XXXXX998" <sip:38XXXX02@172.31.32.169>;tag=p0BjH21gpa3rc To: <sip:38XXXX02@54.XXX.XXX.44>;tag=8c3bb09e-a909-4cb4-99a7-3c57aa8f946e Call-ID: 464f21c3-3120-123b-91b2-026ef2fede4c CSeq: 50116727 ACK Contact: <sip:voip.voip-sbc.6792.fab37901@54.XXX.XXX.44:5060> Content-Length: 0 This causes Kamailio to not know where to forward the message to... Asterisk never gets the reply. This is happening to one of the SIP Trunks I am using, with the other, everything is fine. Is there anything I can do to work around it (other than contacting the provider to fix on their end)? Thanks! ps: I have attached the full SIP messages trail to help.

2 1

Htable reload
by David Villasmil 07 Apr '22

07 Apr '22

Hello all, I’ve been trying to figure out how htable reloads happen. Does it replaces the existing one or creates a new one, populates it from the db and then renames it? The latter would be (I think) ideal, even if it uses more resources. Thanks! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 6

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

Hello Expert, We faces problem in utilizing lcr command line in "kamctl lcr show_gws", whereas kamailio version is 5.5.4 which are new kamailio release, "kamctl lcr show_gws" shown result "command disabled". in this case, LCR module is already been uploaded and the systemd has been running well active kamailio. Could you share, what probably wrong in my kamailio configuration? Best Regards,

2 4

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

Hello Expert, I have problem in utilizing lcr command line in "kamctl lcr show_gws", whereas my kamailio version is 5.5.4 which are new kamailio release, "kamctl lcr show_gws" shown result "command disabled". in this case, LCR module is already been uploaded and the systemd has been running active kamailio. Here detailed screenshot : [image: image.png] Could you share, what probably wrong in my kamailio configuration? Best Regards,

1 1

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

problem in "kamctl lcr show_gws" resulting "command disabled". Kamailio uses LCR module uploaded and running active kamailio. what probably wrong in kamailio? [image: image.png]

1 0

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

problem in "kamctl lcr show_gws" (kamailio version 5.5.4 / new release) resulting "command disabled". LCR module had been uploaded on running well active kamailio. Could you share, what probably wrong in my kamailio configuration? [image: image.png] Best regards

1 0

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

Hello Expert, I have problem in utilize lcr command line "kamctl lcr show_gws", whereas kamailio version is 5.5.4 (new kamailio release), those resulting "command disabled". In this case, LCR module is already been uploaded and the systemd has been running active kamailio. Could you share, what probably wrong in my kamailio configuration? [image: image.png] Best regards

1 0

kamailio 5.5.4 source not found file pcre.h file
by bayu p 07 Apr '22

07 Apr '22

Dear Developer, would you please share which pcre.h file ? due to my compiling source kamailio 5.5.4 found error as captured below. In file included from dialplan.c:59: dialplan.h:33:10: fatal error: pcre.h: No such file or directory 33 | #include <pcre.h> | ^~~~~~~~ compilation terminated. Kind Regards,

2 2

Kamailio preprocessor to get hostname
by David VILLAUME 06 Apr '22

06 Apr '22

Hello all , Do you know any way to get the hostname of my server at kamailio runtime ? The simple idea behind this would be ton import a specific environment file base on hostname. Regards, David

2 1

parse XML nested name-spaces
by Artur Hovsepyan 06 Apr '22

06 Apr '22

good evening everyone, can you please help how to parse following XML body and get the "<gml:pos>31.204359 -44.512358</gml:pos>" in a variable? <?xml version="1.0"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gml="http://www.opengis.net/gml" xmlns:gs="http://www.opengis.net/pidflo/1.0" xmlns:con="urn:ietf:params:xml:ns:geopriv:conf" entity="sip:+6478897789@ims.mnc0xx.mccxxx.3gppnetwork.org"> <dm:device id="Wifi"> <gp:geopriv> <gp:location-info> <gs:Circle srsName="urn:ogc:def:crs:EPSG::5524"> <gml:pos>31.204359 -44.512358</gml:pos> <gs:radius uom="urn:ogc:def:uom:EPSG::8190">10.578000</gs:radius> </gs:Circle> <con:confidence pdf="normal">92</con:confidence> </gp:location-info> <gp:method>DBH_HELO</gp:method> <gp:usage-rules/> </gp:geopriv> <dm:timestamp>2022-04-05T19:16:34Z</dm:timestamp> </dm:device> </presence> i tried few variants: xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/pidf:dm:device/pidf:gp:geopriv/pidf:gp:location-info/pidf:gs:Circle/pidf:gml:pos)\n") xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/pidf:dm\:device/pidf:gp\:geopriv/pidf:gp\:location-info/pidf:gs\:Circle/pidf:gml\:pos)\n"); xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/pidf:dm:device/pidf:gp:geopriv/pidf:gp:location-info/pidf:gs:Circle/pidf:gml:pos)\n"); xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/dm:device/gp:geopriv/gp:location-info/gs:Circle/gml:pos)\n"); and end up with error "unable to evaluate xpath expression": ERROR: xmlops [pv_xml.c:308]: pv_get_xml(): unable to evaluate xpath expression [/pidf:presence/dm:device/gp:geopriv/gp:location-info/gs:Circle/gml:pos/70] thanks in advance,Art

2 2

least cost routing module activation error
by bayu p 06 Apr '22

06 Apr '22

Dear Kamailio Expert, I already add module lcr, but found error when searching pvar "ddn" inside source code core/pvapi.c at line 923, as shown error log captured below : Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/pvapi.c:923]: pv_parse_spec2(): error searching pvar "ddn" Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/pvapi.c:1126]: pv_parse_spec2(): wrong char [n/110] in [$ddn] at [3 (0)] Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: xlog [xlog.c:511]: xdbg_fixup_helper(): wrong format[Domain of destination: $ddn] Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:901 Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:903 Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:918 The question are, how to define correct "INVITE" methode (or correct format) inside route[LCR] which is already defined inside kamailio.cfg: route[LCR] { if (status=="200") { xlog("LCR: Inside the LCR routen"); } if(method=="INVITE") { xlog("We got an invite"); if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Couldn't load gateways"); sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { xlog("GW Selected '$avp(i:709)'n"); xlog("Domain of destination: $ddn"); xlog("To URI: $tun"); } if(!next_gw()) { xlog("Couldn't proceed to next gateway"); sl_send_reply("503", "Service not available, no gateways found"); exit; } else { xlog("Calling the first matched gatewayn"); xlog("ruri_user_avp: '$avp(i:500)'n"); xlog("To URI after next_gw: $tun"); xlog("Request URI: $rUn"); } } } Kind Regards

1 0

escape firt two lines from get_body_part to get xml only part
by Artur Hovsepyan 06 Apr '22

06 Apr '22

hello everyone, i am trying to parse the xml from the message body: get_body_part("application/pidf+xml", "$var(pbody)"); and i am getting the following in my pbody var: Content-Disposition: render;handling=optionalContent-ID: <sip:+6478897789@ims.mnc0xx.mccxxx.3gppnetwork.org><?xml version="1.0"?><presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gml="http://www.opengis.net/gml" xmlns:gs="http://www.opengis.net/pidflo/1.0" xmlns:con="urn:ietf:params:xml:ns:geopriv:conf" entity="sip:+6478897789@ims.mnc0xx.mccxxx.3gppnetwork.org"><dm:device id="Wifi"><gp:geopriv><gp:location-info><gs:Circle srsName="urn:ogc:def:crs:EPSG::5524"><gml:pos>31.204359 -44.512358</gml:pos><gs:radius uom="urn:ogc:def:uom:EPSG::8190">10.578000</gs:radius></gs:Circle><con:confidence pdf="normal">92</con:confidence></gp:location-info><gp:method>DBH_HELO</gp:method><gp:usage-rules/></gp:geopriv><dm:timestamp>2022-04-05T19:16:34Z</dm:timestamp></dm:device></presence> how can i escape the first two lines in order to get the xml part of the content? thanks in advance,Art

2 6

Jansson question
by Barry Flanagan 06 Apr '22

06 Apr '22

Hi I am trying to use the jannson_get function to pull field "secret" from the below json: {"3":{"options":{"secret":"mybigsecret"}}} This works is I use jansson_get("3.options.secret"..... However, I need to know the value of "3" in order to use it. I have tried all the variations on path I can find, but none apart from using the actual root value work for me. How can I either: 1. Construct a PATH which will pick this up, or 2. Obtain this root value ("3" in my example) I have tried testing using https://jsonpath.com/ and a path of "$..secret" and "$..options.secret" works there, but not in jansson kamailio module. Any assistance much appreciated. Thanks -- -Barry

2 1

escape firt two lines from get_body_part to get xml only part
by Does it really matter? 06 Apr '22

06 Apr '22

1 0

Manual usage and example kamailio kamctl
by bayu p 06 Apr '22

06 Apr '22

Dear Expert Kamailio, I use this tutorial kamctl to make kamailio and other sip server communicated https://manpages.ubuntu.com/manpages/xenial/man8/kamctl.8.html but, i cannot found further tutorial how use kamctl. Could you please share, the detail/further tutorial and example of kamctl usage.. Since i always found error/failure during execute some command from the link, here captured: (1). # kamctl cr dump -e \E[37;31mERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_rpc.fifo -e \E[37;31mERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters (2). # kamctl lcr show_gws -e \E[37;31mERROR: command disabled (3). # kamctl lcr reload -e \E[37;31mERROR: command disabled (4). # kamctl ping 192.168.31.50 { "jsonrpc": "2.0", "error": { "code": 400, "message": "Invalid request uri \"192.168.31.50\"" }, "id": 76608 } (5). # kamctl cr reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" }, "id": 76659 } Warm Regards, >

5 16

Kamailio stop processing SIP in high call load with dlg replication via DMQ
by José Seabra 05 Apr '22

05 Apr '22

Hi there, I've two kamailio servers sharing dlg vars between them and I've noticed that during high call load (~50-70 CPS) the kamailio that is processing all the calls and sharing the dialogs with the other kamailio, stops processing SIP. If I set the parameter enable_dmq to 0 I cannot replicate the issue so I assume that it's related to dialog replication through DMQ. *Kamailio version: *5.4.7 - I've tested with recent versions but the issue remains *Configurations of Kamailio_1:* # ----- dialog params ----- modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "enable_dmq", 1) modparam("dialog", "h_id_start", 1000) modparam("dialog", "h_id_step", 6000) # ----- dmq params ----- modparam("dmq", "server_address", "sip:INTERNAL_INSTANCE_IP:5060") modparam("dmq", "notification_address", "DMQ_NOTIFICATION_ADDRESS") modparam("dmq", "multi_notify", 1) modparam("dmq", "ping_interval", 10) modparam("dmq", "num_workers",4) *Configurations of Kamailio_2:* # ----- dialog params ----- modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "enable_dmq", 1) modparam("dialog", "h_id_start", 5000) modparam("dialog", "h_id_step", 6000) # ----- dmq params ----- modparam("dmq", "server_address", "sip:INTERNAL_INSTANCE_IP:5060") modparam("dmq", "notification_address", "DMQ_NOTIFICATION_ADDRESS") modparam("dmq", "multi_notify", 1) modparam("dmq", "ping_interval", 10) modparam("dmq", "num_workers",4) In this link you can find the kamctl trap <https://textbin.net/quykw71nyl>output after kamailio stop processing SIP. Thank you Best Regards -- Cumprimentos José Seabra

1 0

Auth_ephemeral Module > username_format not accepted
by Luciano Motti 05 Apr '22

05 Apr '22

Hi! I was able to successfully use the auth_ephemeral module on my environment: Kamailio > Asterisk > ARI I ran into a problem with ARI, where Asterisk wouldn't send the complete from URI to ARI program, just the first part (timestamp), and the most important part (the username) isn't forwarded. For example: Kamailio username: 1649106119:1001 Asterisk only forwards 1649106119 to ARI, which makes it impossible for me to track the user. So I planned on using the deprecated username_format from Auth Ephemeral module: https://kamailio.org/docs/modules/4.1.x/modules/auth_ephemeral.html#auth_ep… By switching the timestamp with the username, my problems would be fixed. After setting the variable, and trying to register on Kamailio with username 1001:1649106119, I get an error message: SIP/2.0 400 Bad From URI Am I doing something wrong? Thanks!

2 2

Re: [SR-Users] More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS)
by Henning Westerholt 05 Apr '22

05 Apr '22

Hello, please keep the list in CC, that others can comment as well. Beside from the obvious (improving the performance of the lua API call), you could use indeed asynchronous processing. If you just use the lua code to emit the REST request, using a native http client module would be probably also faster as doing it over lua. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Giovanni Jose <gio.jose(a)gmail.com> Sent: Wednesday, March 30, 2022 5:17 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS) Hello Henning, Thanks so much for the fast response I just commented the loop and I was hitting the backend and it's slow anyway maybe it's because, for this type of implementation when the destination of the call is decided by the API backend, I need to use the http_async_client module? I am noticing that for high volume calls kamailio slows the traffic and shows errors because there's a delay that is been added when the backend responds for every call. On Tue, Mar 29, 2022 at 11:53 PM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, why are you sleeping for 1s in the loop? This will cause Kamailio to block processing. I also do not see any stop condition on the loop, so it will loop until the limit. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Giovanni Jose Sent: Tuesday, March 29, 2022 9:46 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS) Hello Guys I am having this issue where kamailio is receiving requests from SIPp (we are currently testing the platform and doing QA) but if we do more than 800 concurrent calls kamailio starts failing and slowing the requests, The scenario is the following: I am sending 5000 calls at 100 CPS on TCP and I am having this route in the configuration file: # Handle the calls to api route[CALL_API] { xlog("L_NOTICE"," Call request $var(call_request) method: $rm \n"); $var(loop_true)=1; while ($var(loop_true)) { if(!lua_run("call_request","$var(call_request)","$sht(token=>new_token)")) { xlog("L_NOTICE", "SCRIPT: failed to execute lua function!\n"); } if ($var(loop_true)){ sleep("1"); } } xlog("L_NOTICE", "SCRIPT: Sucess to execute lua function!\n"); } So I am executing a LUA script for the requests and the API response will tell the destination of the call. At 800 concurrent calls, I can notice SLOWLINESS in the SIPp testing tool, and then I will start receiving 408 timeouts. This sounds like a load issue or something because I believe there's a time that it's been added every time we hit the backend and wait for the response. any help will be appreciated Regards Gio

2 1

Jansson question
by Barry Flanagan 05 Apr '22

05 Apr '22

Hi I am trying to use the jannson_get function to pull field "secret" from the below json: {"3":{"options":{"secret":"mybigsecret"}}} This works is I use jansson_get("3.options.secret"..... However, I need to know the value of "3" in order to use it. I have tried all the variations on path I can find, but none apart from using the actual root value work for me. How can I either: 1. Construct a PATH which will pick this up, or 2. Obtain this root value ("3" in my example) I have tried testing using https://jsonpath.com/ and a path of "$..secret" and "$..options.secret" works there, but not in jansson kamailio module. Any assistance much appreciated. Thanks -- -Barry

1 0

rpm.kamailio.org is down
by Riccardo Villa 05 Apr '22

05 Apr '22

Host is up and responding to ping but not to http requests. Riccardo -- -- NOTICE: This electronic mail transmission may contain confidential information and is intended only for the person(s) named. Any use, copying, or disclosure by any other person is strictly prohibited. If you have received this transmission in error, please notify the sender via e-mail.

2 1

Asterisk, unable to find dialog for ACK with route param and call-id
by Trevor Peirce 05 Apr '22

05 Apr '22

Hello, I'm having some trouble with what I think is the dialog module. I've got Kamailio 5.5.4 running well for some time. Recently I added Asterisk to the mix and have come across an issue. Call to Kamailio. Kamailio sends it to Asterisk. Asterisk may decide to "forward" the call. To do this it will place a new outbound call through Kamailio. From the user perspective, this works absolutely fine (invite, ringing, ok, bye). The acc module handles it fine also. However dialog does not, and the events for ok and bye do not fire on the original call. They do fire for the forwarded call. I've added some logging output to try to isolate the problem, but I'm no further ahead. Here's what I have. Any suggestions? ** Call initiated ** INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=100 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=3360 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=100 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=183 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=183 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=3360 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7fa0494006e8 tl->next=(nil) tl->prev=(nil) ERROR: dialog [dlg_hash.c:1329]: update_dlg_timeout(): failed to update dialog lifetime CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7fa0494006e8 tl->next=(nil) tl->prev=(nil) ERROR: dialog [dlg_hash.c:1329]: update_dlg_timeout(): failed to update dialog lifetime ERROR: dialog [dlg_profile.c:1028]: dlg_set_timeout_by_profile(): Unable to set timeout on 3306:12036 NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611824;time_unix=1648611824.346;method=INVITE;from_tag=as292200a1;to_tag=as75834dc4;call_id=7cef233547dca0ac7882ce5249b3b54c;code=200;reason=OK INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=<null> NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611824;time_unix=1648611824.354;method=INVITE;from_tag=as730369e7;to_tag=as3b06b3e2;call_id=53bad4f0034b61a5745163d82daef402;code=200;reason=OK INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=<null> WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param '161.02d' [353:3360] and call-id '53bad4f0034b61a5745163d82daef402' WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param '161.02d' [353:3360] and call-id '53bad4f0034b61a5745163d82daef402' ** The call is answered and endpoints can talk to each other ** ** Now the original caller hangs up ** WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for BYE with route param '161.02d' [353:3360] and call-id '53bad4f0034b61a5745163d82daef402' NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611831;time_unix=1648611831.933;method=BYE;from_tag=as730369e7;to_tag=as3b06b3e2;call_id=53bad4f0034b61a5745163d82daef402;code=200;reason=OK INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=100 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 ERROR: <script>: got result for bye = Bye!; DLG_lifetime = 8 NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611832;time_unix=1648611832.115;method=BYE;from_tag=as292200a1;to_tag=as75834dc4;call_id=7cef233547dca0ac7882ce5249b3b54c;code=200;reason=OK ERROR: dialog [dlg_handlers.c:363]: dlg_terminated_confirmed(): failed to get dialog from params! ** Everything ends cleanly on the other SIP endpoints ** What can I do from here to try to isolate this further? Thanks, -- Trevor Peirce AcroVoice Solutions Inc

2 4

Kamailio as a SBC
by Nileshwar Goswami 05 Apr '22

05 Apr '22

Hi There, I'm looking out for an SBC and came across your platform. Can Kamailio work as a SBC and provide all the functionalities of a SBC ? A quick reply will be appreciated. Best Regards, Nileshwar Goswami

2 1

RTPENGINE : Kamailio - Get status if max sessions reached
by Youssef Boujraf 01 Apr '22

01 Apr '22

Dear, QUESTION : Through kamailio,We are using rtpengine_manage and the return value is true or false but don"t know for which reason. * How to be informed if rtpengine reach the maximum of sessions and will not accept any rtp routing ? * We have seen if rtpengine reached the max-session; The call is accept by kamailio and finally no audio because SDP not processed- kamailio.log Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: ALERT: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} <script>: [NATMANAGE] branch_id:0 ruri: sip:9919993@public:32617;transport=UDP;rinstance=f5d124565504bdb6, method:INVITE, status:<null>, extra_id: z9hG4bK-524287-1---290944f5547c740a0, rtpengine_manage: replace-origin replace-session-connection via-branch=extraMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine_funcs.c:144]: check_content_type(): type <application/sdp> found validMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844)Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch=z9hG4bK-524287-1---290944f5547c740a0Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2691]: rtpp_function_call(): proxy reply: d6:result10:load limit7:message30:Parallel session limit reachedeMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2715]: rtpp_function_call(): proxy udp:192.168.1.33:2223 has reached its load limit (Parallel session limit reached) - trying next oneMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844) Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch=z9hG4bK-524287-1---290944f5547c740a0 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2691]: rtpp_function_call(): proxy reply: d6:result10:load limit7:message30:Parallel session limit reachede Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2715]: rtpp_function_call(): proxy udp:127.0.0.1:2223 has reached its load limit (Parallel session limit reached) - trying next one Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844) Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch=z9hG4bK-524287-1---290944f5547c740a0 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2900]: rtpp_test(): rtpengine instance <udp:127.0.0.1:2223> found, support for it enabled Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2900]: rtpp_test(): rtpengine instance <udp:192.168.1.33:2223> found, support for it enabled Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: ERROR: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3330]: select_rtpp_node(): rtpengine failed to select new for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: ERROR: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2678]: rtpp_function_call(): no available proxies Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21070]: ALERT: {2 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} <script>: [NATMANAGE] branch_id:0 ruri: <null>, method:INVITE, status:100, extra_id: z9hG4bK-524287-1---290944f5547c740a0, rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: ALERT: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} <script>: [NATMANAGE] branch_id:1 ruri: sip:9919993@public:32617;transport=UDP;rinstance=f5d124565504bdb6, method:INVITE, status:<null>, extra_id: z9hG4bK-524287-1---290944f5547c740a0, rtpengine_manage: replace-origin replace-session-connection Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844) Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch= Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2691]: rtpp_function_call(): proxy reply: d7:warning38:Call-ID not found or tags didn't match6:result2:oke Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2784]: rtpp_function_call(): rtpengine hash table insert node=udp:192.168.1.33:2223 for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch= Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2801]: rtpp_function_call(): rtpengine hash table remove entry for callen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch= - rtpengine.log : Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Received command 'offer' from 192.168.1.41:42822 Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Dump for 'offer' from 192.168.1.41:42822: { "supports": [ "load limit" ], "sdp": "v=0 Mar 29 16:08:14 rtp01srv rtpengine[3921]: o=Zoiper 141319690 1 IN IP4 192.168.30.216 Mar 29 16:08:14 rtp01srv rtpengine[3921]: s=Z Mar 29 16:08:14 rtp01srv rtpengine[3921]: c=IN IP4 192.168.30.216 Mar 29 16:08:14 rtp01srv rtpengine[3921]: t=0 0 Mar 29 16:08:14 rtp01srv rtpengine[3921]: m=audio 8000 RTP/AVP 0 101 8 3 97 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=rtpmap:101 telephone-event/8000 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=fmtp:101 0-16 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=rtpmap:97 iLBC/8000 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=fmtp:97 mode=30 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=sendrecv Mar 29 16:08:14 rtp01srv rtpengine[3921]: ", "replace": [ "origin", "session-connection" ], "call-id": "nIdGvZZNiFnt3_r0xSIKpQ..", "via-branch": "z9hG4bK-524287-1---290944f5547c740a0", "received-from": [ "IP4", "public" ], "from-tag": "75983054", "command": "offer" } Mar 29 16:08:14 rtp01srv rtpengine[3921]: ERR: [nIdGvZZNiFnt3_r0xSIKpQ..]: [core] Parallel session limit reached (2) Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Replying to 'offer' from 192.168.1.41:42822 (elapsed time 0.000058 sec) Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Response dump for 'offer' to 192.168.1.41:42822: { "result": "load limit", "message": "Parallel session limit reached" } Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [control] Received command 'ping' from 192.168.1.41:42822 Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [control] Dump for 'ping' from 192.168.1.41:42822: { "command": "ping" } Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [control] Replying to 'ping' from 192.168.1.41:42822 (elapsed time 0.000001 sec) Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [control] Response dump for 'ping' to 192.168.1.41:42822: { "result": "pong" } How to get this information then we could cancel the call or send sip message that the number of max call is reached ? -- Youssef

3 7

Kamailio PCSCF route logic
by Weiqi Yao 30 Mar '22

30 Mar '22

Hello community, We made some modifications to Kamailio PCSCF sample cfg file, so when PCSCF receiving an tcp SUBSCRIBE message it add ;transport=tcp to the end of Contact URI. Below is what we added the route block right after if (is_method("INVITE|SUBSCRIBE")) logic if (is_method("SUBSCRIBE") && proto=='tcp') { $var(original_contact_header) = $ct; $var(new_contact_header) = $(var(original_contact_header){re.subst,/(.*:.*)(>);/\1;transport=tcp>/g}); remove_hf("Contact"); append_hf("Contact: $var(new_contact_header)\r\n"); msg_apply_changes(); } We did see the ;transport=tcp been added but we also seeing a bad line ;alias=[...] is added before the To: line [cid:image001.png@01D84375.8D109B20] Does anyone know what function might cause this ;alias been added before To: line? BR, Weiqi

2 2

Parallel forking call drop scenario
by Koray Vatansever 30 Mar '22

30 Mar '22

Hi everyone, I'm facing the following situation. Clients are connected over TLS. Therefore there is no retransmission in Kamailio. How can I prevent this call drop? +---+ +-----------+ +-----+ +-----+ | A | | Kamailio | | B1 | | B2 | +---+ +-----------+ +-----+ +-----+ | | | | | INVITE | | | |------------------------------------>| | | | | -------------------\ | | | |-| Parallel forking | | | | | |------------------| | | | | | | | | INVITE | | | |---------------------->| | | | | | | | INVITE | | | |--------------------------------------------------->| | | | | | | 180 Ringing | | | |<----------------------| | | | | | | | | 180 Ringing | | |<---------------------------------------------------| | | | ------------------------\ | | | |-| User answers the call | | | | | |-----------------------| | | | | | | | 200 OK | | | |<----------------------| | | | | | | 200 OK | | | |<------------------------------------| | | | | | | | ACK | | | |------------------------------------>| | | | | | | | | ACK | | | |---------------------->| | | ------------------------------\ | | | |-| Session established with B1 | | | | | |-----------------------------| | | | | | | | ----------------------------\ | | | |-| Client changes network | | | | | | LTE to Wifi or vice versa | | | | | | Temporary connection lost | | | CANCEL | | |---------------------------| | |----------------------------------------------X | | | | | -------------------------------------------------\ | | | |-| Reconnected with different IP | | | | | | CANCEL is missed and client continues ringing. | | | | | | User answers the call. | | | | 200 OK | |------------------------------------------------| | |<---------------------------------------------------| | | | | | 200 OK | | | |<------------------------------------| | | | | | | | | | BYE | | |<---------------------------------------------------| | | | | | BYE | | | |<------------------------------------| | | | ---------------------------\ | | | |-| Session terminated by B2 | | | | | | Call Drop | | | | | |--------------------------| | | |

2 2

More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS)
by Giovanni Jose 30 Mar '22

30 Mar '22

Hello Guys I am having this issue where kamailio is receiving requests from SIPp (we are currently testing the platform and doing QA) but if we do more than 800 concurrent calls kamailio starts failing and slowing the requests, The scenario is the following: I am sending 5000 calls at 100 CPS on TCP and I am having this route in the configuration file: # Handle the calls to api route[CALL_API] { xlog("L_NOTICE"," Call request $var(call_request) method: $rm \n"); $var(loop_true)=1; while ($var(loop_true)) { if(!lua_run("call_request","$var(call_request)","$sht(token=>new_token)")) { xlog("L_NOTICE", "SCRIPT: failed to execute lua function!\n"); } if ($var(loop_true)){ sleep("1"); } } xlog("L_NOTICE", "SCRIPT: Sucess to execute lua function!\n"); } So I am executing a LUA script for the requests and the API response will tell the destination of the call. At 800 concurrent calls, I can notice SLOWLINESS in the SIPp testing tool, and then I will start receiving 408 timeouts. This sounds like a load issue or something because I believe there's a time that it's been added every time we hit the backend and wait for the response. any help will be appreciated Regards Gio

2 1

dispatcher socket attrs binding
by David Villasmil 29 Mar '22

29 Mar '22

Hello guys, Im starting kamailio with listen=0.0.0.0:5060 and in my dispatcher list i have in the attrs something like: socket=udp:172.17.0.2:5060 That IP address belongs to kamailio's host and it is binging to it. Is it right i can't set a specific IP in the socket attribute if listen is 0.0.0.0? maybe i should be able to? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 4

kamailio - mariadb hosted on remote server - issue
by Youssef Boujraf 29 Mar '22

29 Mar '22

Dear all, I don't belive to connect Kamailio to a remote Mariadb will take too much time. STEP 1 : KAMAILIO : -------------------------------- $ sudo kamailio -v version: kamailio 5.5.4 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 STEP 2 : Set the kamctlrc config ------------------------------------------------ KAMAILIO IP : 192.168.1.41 MARIADB IP : 192.168.1.39 - /etc/kamailio/kamctlrc SIP_DOMAIN=example.be DBENGINE=MYSQL DBHOST=192.168.1.39 // where the mariadb is installed DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="XX" DBROUSER="kamailioro" DBROPW="XX" DBACCESSHOST=192.168.1.41 // where kamctl is launched (here kamailio srv) DBROOTUSER="root" DBROOTPW="kamailio" CHARSET="latin1" STEP 3 : MARIADB : hosted on different srv than kamailio -------------------------------------------------------------------------------------- IP : 192.168.1.39 - /etc/mysql/mariadb.conf.d/50-server.cnf # this is only for the mysqld standalone daemon [mysqld] # Instead of skip-networking the default is now to listen only on # localhost which is more compatible and is not less secure. #bind-address = 127.0.0.1 bind-address = 0.0.0.0 STEP 4 : TEST : Reote connection with the DB ---------------------------------------------------------------------- REMOTE IP : 192.168.1.41 DB IP : 192.168.1.39 $ sudo mysql -u root -h 192.168.1.39 -p Enter password: Welcome to the MariaDB monitor. Commands end with ; or \g. Your MariaDB connection id is 298 Server version: 10.5.15-MariaDB-1:10.5.15+maria~bullseye mariadb.org binary distribution Copyright (c) 2000, 2018, Oracle, MariaDB Corporation Ab and others. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. MariaDB [(none)]>The connection is established with success. STEP 5 : KAMAILIO DB : creation ---------------------------------------------------- $ sudo kamdbctl create -e \E[37;33mINFO: creating database kamailio ... -e \E[37;33mINFO: granting privileges to database kamailio ... -e \E[37;33mINFO: creating standard tables into kamailio ... -e \E[37;33mINFO: Core Kamailio tables successfully created. Create the presence related tables? (y/n): y -e \E[37;33mINFO: creating presence tables into kamailio ... -e \E[37;33mINFO: Presence tables successfully created. Create the tables for imc cpl siptrace domainpolicy carrierroute drouting userblocklist htable purple uac pipelimit mtree sca mohqueue rtpproxy rtpengine secfilter? (y/n): y -e \E[37;33mINFO: creating extra tables into kamailio ... -e \E[37;33mINFO: Extra tables successfully created. Create the tables for uid_auth_db uid_avp_db uid_domain uid_gflags uid_uri_db? (y/n): y -e \E[37;33mINFO: creating uid tables into kamailio ... -e \E[37;33mINFO: UID tables successfully created. STEP 6 : DB : verify the users creation and rights -------------------------------------------------------------------------- MariaDB [(none)]> select user, host from mysql.user; +-------------+--------------+ | User | Host | +-------------+--------------+ | kamailio | 192.168.1.39 | | kamailioro | 192.168.1.39 | | kamailio | 192.168.1.41 | | kamailioro | 192.168.1.41 | | root | 192.168.1.41 | | kamailio | localhost | | kamailioro | localhost | | mariadb.sys | localhost | | mysql | localhost | | root | localhost | +-------------+--------------+ 10 rows in set (0.001 sec) MariaDB [(none)]> SHOW GRANTS FOR kamailio(a)192.168.1.41; +--------------------------------------------------------------------------------------------------------------------+ | Grants for kamailio(a)192.168.1.41 | +--------------------------------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO `kamailio`(a)`192.168.1.41` IDENTIFIED BY PASSWORD '*157PASSWORD' | | GRANT ALL PRIVILEGES ON `kamailio`.* TO `kamailio`(a)`192.168.1.41` | +--------------------------------------------------------------------------------------------------------------------+ 2 rows in set (0.000 sec) MariaDB [(none)]> SHOW GRANTS FOR kamailio(a)192.168.1.39; +--------------------------------------------------------------------------------------------------------------------+ | Grants for kamailio(a)192.168.1.39 | +--------------------------------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO `kamailio`(a)`192.168.1.39` IDENTIFIED BY PASSWORD '*157PASSWORD' | | GRANT ALL PRIVILEGES ON `kamailio`.* TO `kamailio`(a)`192.168.1.39` | +--------------------------------------------------------------------------------------------------------------------+ 2 rows in set (0.000 sec) MariaDB [(none)]> SHOW GRANTS FOR kamailio@localhost; +-----------------------------------------------------------------------------------------------------------------+ | Grants for kamailio@localhost | +-----------------------------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO `kamailio`@`localhost` IDENTIFIED BY PASSWORD '*157PASSWORD' | | GRANT ALL PRIVILEGES ON `kamailio`.* TO `kamailio`@`localhost` | +-----------------------------------------------------------------------------------------------------------------+ 2 rows in set (0.000 sec)STEP 7 : ADDRESS TABLE : add new ip address$ sudo systemctl restart kamailio sipsecure@kamailio:/etc$ sudo kamctl address add 0 192.168.1.26 32 0 HP -e \E[37;33mINFO: execute '/sbin/kamctl address reload' to synchronize cache and database STEP 8 : DB : verify if the new ip address is created ? Yes MariaDB [(none)]> select * from kamailio.address; +----+-----+--------------+------+------+------+ | id | grp | ip_addr | mask | port | tag | +----+-----+--------------+------+------+------+ | 1 | 0 | 192.168.1.26 | 32 | 0 | HP | +----+-----+--------------+------+------+------+ 1 row in set (0.000 sec) STEP 9 : Reload the content of the DB into kamailio. It failed ---------------------------------------------------------------------------------------- The host: '192.168.1.41' is kamailio (remote client)$ sudo kamctl address reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Reload failed." }, "id": 12397 } In /var/log/daemon.log Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 281 [Warning] Aborted connection 281 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 273 [Warning] Aborted connection 273 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 295 [Warning] Aborted connection 295 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 264 [Warning] Aborted connection 264 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Now, if I am trying to restart "kamailio" service, I got an error : Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: DEBUG: <core> [core/sr_module.c:940]: init_mod(): permissions Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: DEBUG: permissions [address.c:179]: reload_address_db_table(): Number of rows in address table: 1 Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: DEBUG: permissions [address.c:191]: reload_address_db_table(): failure during checks of database value 1 (group) in address table Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: ERROR: permissions [address.c:233]: reload_address_db_table(): database problem - invalid record Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: CRITICAL: permissions [address.c:536]: init_addresses(): reload of address table failed Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: ERROR: permissions [permissions.c:657]: mod_init(): failed to initialize the allow_address function Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module permissions (/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so) Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/cfg.y:1860]: yyparse(): loading module permissions.so Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:513]: ksr_locate_module(): found module to load </usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so> Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:563]: load_module(): trying to load </usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so> Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/kemi.c:3090]: sr_kemi_modules_add(): adding module: permissions Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1963]: pp_define(): defining id: MOD_permissions Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:112]: set_mod_param_regex(): 'permissions' matches module 'permissions' Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:811]: find_param_export(): found <db_url> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:128]: set_mod_param_regex(): found <db_url> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:112]: set_mod_param_regex(): 'permissions' matches module 'permissions' Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:811]: find_param_export(): found <load_backends> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:128]: set_mod_param_regex(): found <load_backends> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:31 kamailio kamailio: DEBUG: <core> [core/sr_module.c:719]: find_mod_export_record(): found export of <allow_source_address> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:31 kamailio kamailio: DEBUG: <core> [core/sr_module.c:719]: find_mod_export_record(): found export of <allow_source_address_group> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.soIf I remove the record from the database manually, I can restart kamailio but no way to add any address. Thanks if someone could highlight my path. Best Regards, -- Youssef

2 4

avps on regexps?
by David Villasmil 28 Mar '22

28 Mar '22

Hello guys, Is it possible to use apvs on regexps like: $avp(myvar)=$(re{uri.user}{re.subst,/^\+$avp(some_var)/$avp(some_var)/}); doesn't seem possible Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

proxy and forward
by jaflong jaflong 28 Mar '22

28 Mar '22

2 1

nathelper OPTIONS ping
by Xuo Guoto 28 Mar '22

28 Mar '22

Hello all, I am trying to periodically send SIP OPTIONS to all connected WebRTC clients from Kamailio. The functionality is similar to qualify=yes of Asterisk. Following are the configuration changes I have made to get this working. #!define FLB_NATSIPPING 7 <snip/> loadmodule "nathelper.so" <snip/> # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "natping_interval", 20) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") <snip/> # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("L_INFO", "Processing REGISTER in route[REGISTRAR]\n"); if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # do SIP NAT pinging xlog("L_INFO", "Setting FLB_NATSIPPING\n"); setbflag(FLB_NATSIPPING); } if (!save("location")) { sl_reply_error(); } xlog("L_INFO", "Successfully processed REGISTER in route[REGISTRAR]\n"); exit; } } When the WebRTC client registers, I can see the log: Setting FLB_NATSIPPING, but SIP OPTIONS packets are not seen. I am checking it using the Chrome console, at client side as well as sipdump module in server side. Do I have to do any additional configuration? I am not posting the full config file here so that its easy to focus on the relevant parts, but can do that if needed. Thanks and regards, X.

2 1

Re: [SR-Users] Kamailio : kamctl address reload with HAProxy
by Henning Westerholt 28 Mar '22

28 Mar '22

Hello, please keep the list in CC – that others can comment as well. Can you actually connect from the kamailio system e.g. with mysql client? If yes, as I said – I would try to debug the issue further with network analysis tools. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Paclinio <thomas.leutscher(a)gmail.com> Sent: Monday, March 28, 2022 1:11 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio : kamctl address reload with HAProxy Hello, KAMCTL : I've resolved the problem with kamctl that now can communicate with my remote Cluster by adding this line in kamctlrc. // HAPROXY = 192.168.1.37 DBHOST=192.168.1.37 DBACCESSHOST=192.168.1.37 I can now add and reload new addresses with success. KAMAILIO + HAPROXY: But now, I've this issue when restarting kamailio. Mar 28 10:54:39 kamailio /usr/sbin/kamailio[8416]: ERROR: permissions [address.c:233]: reload_address_db_table(): database problem - invalid record Mar 28 10:54:39 kamailio /usr/sbin/kamailio[8416]: ERROR: permissions [permissions.c:657]: mod_init(): failed to initialize the allow_address function Mar 28 10:54:39 kamailio /usr/sbin/kamailio[8416]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module permissions (/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so) My DBURL is #!substdef "!DB_USER!kamailio!g" #!substdef "!DB_PASS!XXXXX!g" #!substdef "!DB_HOST!192.168.1.37!g" // HA PROXY #!substdef "!DB_PORT!MY_PORT_DB!g" #!substdef "!DB_NAME!kamailio!g" #!define DBURL "mysql://DB_USER:DB_PASS@DB_HOST/DB_NAME" KAMAILIO + REMOTE DB : Even I using a remote database I'm getting the same issue. I've discovered the connection between kamailio and remote database is aborded instantly after restarting kamailio. Mar 28 11:47:30 db03srv mariadbd[874]: 2022-03-28 11:47:30 44076 [Warning] Aborted connection 44076 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Thank you [Das Bild wurde vom Absender entfernt.]<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…> Le dim. 27 mars 2022 à 22:52, Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> a écrit : Hello, it seems the data that gets delivered to the Kamailio is corrupted. Analyse it further e.g. with a network trace, wireshark etc.. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Paclinio Sent: Friday, March 25, 2022 5:22 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Kamailio : kamctl address reload with HAProxy Dear colleagues, I've installed a Galera cluster with a HAProxy that kamailio interacts with. Kamailio's DBURL is linked to the proxy that will contact the databases. Kamailio works fine except when I need to reload something from the database (Using kamctl). When cache and database are the same I receive sudo kamctl address reload { "jsonrpc": "2.0", "result": "Reload OK", "id": 13111 } Now when I have something using kamctl, it still works and is replaced on all my databases. sipsecure@kamailio:~$ sudo kamctl address add 0 192.168.48.15 32 0 Test -e \E[37;33mINFO: execute '/sbin/kamctl address reload' to synchronize cache and database Then when i try to reload i receive sipsecure@kamailio:~$ sudo kamctl address reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Reload failed." }, "id": 13504 } And when I'm going to read my logs to see a error, I only see that little sentence Mar 25 17:14:40 kamailio /usr/sbin/kamailio[13343]: ERROR: permissions [address.c:233]: reload_address_db_table(): database problem - invalid record Could I have some explanation ? Thank you. [Das Bild wurde vom Absender entfernt.]<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 2

t_suspend / t_continue & insert_hf
by Fuad Trle 28 Mar '22

28 Mar '22

Hello, I'm inserting some headers after getting value from http_async_query. If a call is relayed, I see them. But If I t_suspend and t_continue call (before time out), I do not see added headers. Here is flow: route(START) t_newtran(); #.. prep for query http_async_query("url","HTTP"); route(HTTP) #.. collect and using data from query insert_hf("$var(key): $var(value)\r\n"); t_store(); if (registered("location") { route(RELAY); } else { t_suspend(); exit; } route(AWAKE) t_continue(); I see that the branch index ($T_branch_idx) is 0 at start. 1 after http query, +1 if there is forking or location. +1 after t_suspend. It looks like t_continue uses a branch with index after t_suspend. I tried with t_save_lumps() to carry over changes. Any idea how to do this?

3 3

What is the status of nat_traversal.so
by hanxiaofang 28 Mar '22

28 Mar '22

Hello Kamailio community, What is the status of nat_traversal.so? https://www.kamailio.org/wiki/alphaindexes/devel/modfunctions Could not find any functions for this module here. Regards hanxiaofang

2 1

Kamailio : kamctl address reload with HAProxy
by Paclinio 27 Mar '22

27 Mar '22

Dear colleagues, I've installed a Galera cluster with a HAProxy that kamailio interacts with. Kamailio's DBURL is linked to the proxy that will contact the databases. Kamailio works fine except when I need to reload something from the database (Using kamctl). When cache and database are the same I receive sudo kamctl address reload > { > "jsonrpc": "2.0", > "result": "Reload OK", > "id": 13111 > } Now when I have something using kamctl, it still works and is replaced on all my databases. sipsecure@kamailio:~$ sudo kamctl address add 0 192.168.48.15 32 0 Test > -e \E[37;33mINFO: execute '/sbin/kamctl address reload' to synchronize > cache and database Then when i try to reload i receive sipsecure@kamailio:~$ sudo kamctl address reload > { > "jsonrpc": "2.0", > "error": { > "code": 500, > "message": "Reload failed." > }, > "id": 13504 > } And when I'm going to read my logs to see a error, I only see that little sentence Mar 25 17:14:40 kamailio /usr/sbin/kamailio[13343]: ERROR: permissions > [address.c:233]: reload_address_db_table(): database problem - invalid > record Could I have some explanation ? Thank you. <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…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

2 1

Using DNS log in
by Steve 26 Mar '22

26 Mar '22

Hello, I have a very basic Kamailio 5.4 configuration on Debian 11 with RTPProxy 2.2 behind a firewall. It works fine with the sip client logging into the server public ip address. However, the sip client is not able to log in when using a dns domain name directed to that public ip address. Is there a specific setting in the Kamailio and/or RTPProxy config files for using a dns domain name instead of a numeric ip address? Also, is there a Kamailio module that allows use of encrypted keys instead of passwords when logging in? Thanks for any info, Steve

2 2

Forward and proxy
by jaflong 26 Mar '22

26 Mar '22

Hi Kamailio community Any advise on how to archive this sip.mydomain.com sip.mypbx.com sip_phone =======sip/tls=======> KAMAILIO =========sip/udp=======>PBX sip_phone should register to Kamilio with tls Kamailio should proxy register to PBX with udp sip_phone invites should be proxied to PBX in sip_client setting he will point to sip.mydomain.com as his server Regards

2 1

(no subject)
by jaflong jaflong 26 Mar '22

26 Mar '22

2 1

t_suspend / t_continue & insert_hf
by Fuad Trle 25 Mar '22

25 Mar '22

Hello, I'm inserting some headers after getting value from http_async_query. If a call is relayed, I see them. But If I t_suspend and t_continue call (before time out), I do not see added headers. Here is flow: route(START) t_newtran(); #.. prep for query http_async_query("url","HTTP"); route(HTTP) #.. collect and using data from query insert_hf("$var(key): $var(value)\r\n"); t_store(); if (registered("location") { route(RELAY); } else { t_suspend(); exit; } route(AWAKE) t_continue(); I see that the branch index ($T_branch_idx) is 0 at start. 1 after http query, +1 if there is forking or location. +1 after t_suspend. It looks like t_continue uses a branch with index after t_suspend. I tried with t_save_lumps() to carry over changes. Any idea how to do this?

1 0

Kamailio SCSCF forward SIP INVITE over TCP
by Weiqi Yao 24 Mar '22

24 Mar '22

Hello community, We currently have some issues with Kamailio SCSCF forwarding a large SIP UDP INVITE packet(size around 3000bytes), it got dropped somewhere in the ethernet. I tried increase the MTU size on both sides' interface and even define it on the switch port, but still no luck. So I am thinking to force SCSCF to send this INVITE over TCP, I followed this link: https://andersbrownworth.com/cms/264/, enabled naptr in SCSCF config and configured NAPTR and SRV DNS record, but the SCSCF still stop even forward it out. I think I did the DNS part right cause when I switch the NAPTR record to point to UDP it worked. Anyone have some experience with configure SCSCF forwarding SIP through TCP? BR, Weiqi

1 0

JsSIP:UA Request-URI does not point to us
by Xuo Guoto 24 Mar '22

24 Mar '22

Hello all, I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows: INVITE -----------------------------> <-------------------------------SIP/2.0 100 Trying <-------------------------------SIP/2.0 180 Ringing <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------BYE 404 Not Found ----------------------> When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us From another thread with similar issue at https://groups.google.com/g/sip_js/c/uiaXS_qc2n8 it could be that JsSIP is not recognizing the GRUU is pointing towards it. In the INVITE message I can see a line "Record-Route: > <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> < <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> mailto:;expires=600;received=mailto:;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 < <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com> < mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> < mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com>

2 4

Re: [SR-Users] sr-users Digest, Vol 202, Issue 25
by 韩小仿 24 Mar '22

24 Mar '22

Hi, Henning Westerholt Pls close case1 Thanks! hanxiaofang From: sr-users-request Date: 2022-03-24 19:00 To: sr-users Subject: sr-users Digest, Vol 202, Issue 25 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: {param.in, name[, delimiter]} error when delimiter is comma (Henning Westerholt) 2. TOPOH is generating new Call-ID for BYE (Marrold) 3. Re: SIPS Errors on Kamailio (Christopher Vincent) 4. bug in tm module? (Daniel Paape) 5. Re: bug in tm module? (Henning Westerholt) 6. carrierroute with non-numeric characters (Arek ?azarczyk) 7. Re: carrierroute with non-numeric characters (Henning Westerholt) 8. Re: carrierroute with non-numeric characters (Arek ?azarczyk) 9. Re: carrierroute with non-numeric characters (Henning Westerholt) ---------------------------------------------------------------------- Message: 1 Date: Wed, 23 Mar 2022 12:49:40 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: ??? <13715209697(a)139.com> Subject: Re: [SR-Users] {param.in, name[, delimiter]} error when delimiter is comma Message-ID: <PAXPR05MB855899124FF2FD96E43C7E7FBF189(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="utf-8" Hello, Please have a look to the documentation, the issue is described there, and a workaround also suggested. http://www.kamailio.org/wiki/cookbooks/devel/transformations#parameters_lis… Cheers, Henning -- Henning Westerholt ? https://skalatan.de/blog/ Kamailio services ? https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of ??? Sent: Wednesday, March 23, 2022 11:12 AM To: sr-users <sr-users(a)lists.kamailio.org> Subject: [SR-Users] {param.in, name[, delimiter]} error when delimiter is comma Hi, all Kamailio5.5.4 $var(s) = "a=1;b=2;c=3"; $var(s1) = $(var(s){param.value,b}); // work well $var(s) = "a=1|b=2|c=3"; $var(s1) = $(var(s){param.value,b,|}); // work well // but $var(s) = "a=1,b=2,c=3"; $var(s1) = $(var(s){param.value,b,,}); ERROR: <core> [core/pvapi.c:1131]: pv_parse_spec2(): invalid parsing in [$(var(s){param.value,b,,})] at (4) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 366, column 13-38: Can't get from cache: $(var(s){param.value,b,,}) how to fix this? Thanks! ________________________________ hanxiaofang -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/836492af/…> ------------------------------ Message: 2 Date: Wed, 23 Mar 2022 15:05:58 +0000 From: Marrold <kamailio(a)marrold.co.uk> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] TOPOH is generating new Call-ID for BYE Message-ID: <CAC-Lcd8H5H=V4ua7y7Fv-t=LFoC=JZC9FZaKH5JtO+K1QJNxcw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi, I have enabled the TOPOH module with mask_callid enabled, but it's generating a new Call-ID for a BYE which gets rejected with a 481 downstream. If I disable the TOPOH module it works as expected. Does anyone know where I should start looking? SIP-wise the initial BYE seems compliant. This is Kamailio 5.5.3 on Debian 11 Thanks Matthew -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/be811b10/…> ------------------------------ Message: 3 Date: Wed, 23 Mar 2022 18:13:54 +0000 From: Christopher Vincent <CDV(a)redwoodtech.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] SIPS Errors on Kamailio Message-ID: <3C0E2FDC7BD2334492F46C0DA6E07428014E8D1BD0(a)RTG-Exchange01.Redwood.com> Content-Type: text/plain; charset="utf-8" Recompiled Kamailio from source kamailio -v version: kamailio 5.5.4 (x86_64/linux) 4c8938 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4c8938 compiled on 14:21:51 Mar 23 2022 with gcc 4.8.5 Compiled a newer version of OpenSSL openssl version OpenSSL 1.1.1m 14 Dec 2021 Then followed through this http://www.kamailio.org/wiki/tutorials/tls/howto-openssl-1-0 Errors changed slightly, but issue still persists kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(3878) ERROR: <core> [core/sr_module.c:570]: load_module(): could not open module </usr/local/lib64/kamailio/modules/tls.so>: /usr/local/lib64/kamailio/modules/tls.so: undefined symbol: OPENSSL_sk_num 0(3878) CRITICAL: <core> [core/cfg.y:3684]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 106, column 12-19: failed to load module 0(3878) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 253, column 72: Can't set module parameter 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 254, column 72: Can't set module parameter 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 255, column 68: Can't set module parameter 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column 39: Can't set module parameter ERROR: bad config file (5 errors) (parsing code: 0) 0(3878) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized sed /usr/local/etc/kamailio/kamailio.cfg -n -e 106p -e 253p -e 254p -e 255p -e 260p loadmodule "tls.so" modparam("tls", "private_key", "/etc/kamailio/star.redwoodtest.com.pem") modparam("tls", "certificate", "/etc/kamailio/star.redwoodtest.com.pem") modparam("tls", "ca_list", "/etc/kamailio/star.redwoodtest.com.pem") modparam("tls", "tls_method", "TLSv1+") Thanks, Chris From: Christopher Vincent Sent: 21 March 2022 09:41 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Christopher Vincent <CDV(a)redwoodtech.com> Subject: RE: [SR-Users] SIPS Errors on Kamailio Hi Karsten, Sergey, Thank you for your responses. Kamailio module appears to be installed rpm -ql kamailio-tls /usr/lib64/kamailio/modules/auth_identity.so /usr/lib64/kamailio/modules/tls.so /usr/lib64/kamailio/openssl_mutex_shared /usr/lib64/kamailio/openssl_mutex_shared/openssl_mutex_shared.so /usr/share/doc/kamailio/modules/README.auth_identity /usr/share/doc/kamailio/modules/README.tl Kamailio was installed from the repository using yum / dnf https://rpm.kamailio.org/centos/kamailio.repo Load module order has been corrected to as below loadmodule "sl.so" loadmodule "tls.so" loadmodule "jsonrpcs.so" loadmodule "db_mysql.so" loadmodule "kex.so" ... Kamailio ?c still showed the same errors Thanks, Chris From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Safarov Sent: 19 March 2022 07:38 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] SIPS Errors on Kamailio SECURITY WARNING: This email is from an external source - do not open or click any attachments if you suspect the email is suspicious. Please report any suspicious emails to Information Security (InformationSecurity(a)redwoodtech.com<mailto:InformationSecurity@redwoodtech.com>) In your config "tls" module after "sl" module. loadmodule "sl.so" loadmodule "tls.so" you need load tls module before "jsonrpcs" module. On Fri, Mar 18, 2022 at 9:59 PM Karsten Horsmann <khorsmann(a)gmail.com<mailto:khorsmann@gmail.com>> wrote: Hi, are you sure the Kamailio tls module is on your system? Check the module path for tls.so like this or if you have build it from source? rpm -ql kamailio-tls /usr/lib64/kamailio/modules/auth_identity.so /usr/lib64/kamailio/modules/tls.so /usr/lib64/kamailio/openssl_mutex_shared /usr/lib64/kamailio/openssl_mutex_shared/openssl_mutex_shared.so /usr/share/doc/kamailio/modules/README.auth_identity /usr/share/doc/kamailio/modules/README.tls Christopher Vincent <CDV(a)redwoodtech.com<mailto:CDV@redwoodtech.com>> schrieb am Fr., 18. M?rz 2022, 12:37: Hi, Kamailio / RTPEngine was set up on CentOS 8 running SIP to SIPS and RTP to SDES SRTP conversion. This worked as expected Attempted to duplicate the setup on RHEL but errors were seen. These errors were present on both RHEL 7 / RHEL 8. The errors seen were as below kamailio -c loading modules under config path: /usr/lib64/kamailio/modules/ 0(9165) ERROR: tls [tls_init.c:611]: tls_pre_init(): Unable to set the memory allocation functions 0(9165) ERROR: tls [tls_init.c:613]: tls_pre_init(): libssl current mem functions - m: 0x7f7a77c367a0 r: 0x7f7a77c367f0 f: 0x7f7a77c36770 0(9165) ERROR: tls [tls_init.c:615]: tls_pre_init(): module mem functions - m: 0x7f7a72db7653 r: 0x7f7a72db769f f: 0x7f7a72db76fc 0(9165) ERROR: tls [tls_init.c:617]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(9165) ERROR: <core> [core/sr_module.c:590]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(9165) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 137, column 12-19: failed to load module 0(9165) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 249, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 250, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 251, column 68: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 256, column 39: Can't set module parameter ERROR: bad config file (5 errors) (parsing code: 0) 0(9165) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized The kamailio config was exactly the same as on the CentOS systems and started as below /* Server ports: */ #!substdef "!SIP_PORT!5060!g" #!substdef "!SIPS_PORT!5061!g" /* Listen addresses */ #!substdef "!UDP_LOCAL_ADDR!udp:SERVER_IP_ADDR:SIP_PORT!g" #!substdef "!TCP_LOCAL_ADDR!tcp:SERVER_IP_ADDR:SIPS_PORT!g" /* Server connections: */ #!ifndef MAX_CONNECTIONS #!define MAX_CONNECTIONS 8192 #!endif ##!define WITH_DEBUG /* Transaction and branch flags: FLT_ - per transaction (message) flags FLB_ - per branch flags */ #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_OUT 8 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define KAMAILIODBURL1 "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_TLS enable_tls=1 listen=tls:<ipaddr>:5062 ####### Global Parameters ######### ### LOG Levels: ALERT=-5, BUG=-4, CRIT=-3, ERR=-1, WARN=0, NOTICE=1, INFO=2, DBG=3 #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 /* display memory usage on exit */ mem_summary=15 /* join free memory fragments */ mem_join=1 /* proxy will fork and run in daemon mode */ /* one process will be created for each network interface the proxy listens to and for each protocol (TCP/UDP), multiplied with the value of 'children' parameter */ fork=yes children=8 listen=TCP_LOCAL_ADDR listen=UDP_LOCAL_ADDR /* life time of TCP connection when there is no traffic - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* sip over websockets may not specify a content length header */ tcp_accept_no_cl=yes /* buffer size used for tcp reads, limits the maximum message size (SIP, HTTP) that can be received over tcp */ tcp_rd_buf_size=65536 /* max number of tcp connections */ tcp_max_connections=MAX_CONNECTIONS ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib64/kamailio/modules/" loadmodule "jsonrpcs.so" loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "cfgutils.so" loadmodule "textopsx.so" loadmodule "nathelper.so" loadmodule "uac.so" loadmodule "ipops.so" loadmodule "debugger.so" loadmodule "exec.so" loadmodule "avpops.so" loadmodule "sqlops.so" loadmodule "rtpengine.so" loadmodule "sl.so" loadmodule "tls.so" # ----------------- setting module-specific parameters --------------- # ----- usrloc params ----- # store contacts in memory only modparam("usrloc", "db_mode", 0) # hash size of 16,384 modparam("usrloc", "hash_size", 14) # removes contact if ws disconnects modparam("usrloc", "handle_lost_tcp", 1) modparam("tm|usrloc", "xavp_contact", "ulattrs") # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) #default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- uac params ----- modparam("uac", "restore_mode", "none") # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "max_contacts", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # disable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direction of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- dispatcher params ----- modparam("dispatcher", "db_url", KAMAILIODBURL1) modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:<address>.com<sip:%3caddress%3e.com>") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 3) # ----- pv params ----- modparam("pv", "shvset", "maintenance=i:0") modparam("pv", "shvset", "virtualIP1=i:0") modparam("pv", "shvset", "virtualIP2=i:0") # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is only being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") #modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") #modparam("rtpengine", "force_send_interface", SERVER_IP_ADDR) #modparam("rtpengine", "setid_default", -1) #modparam("rtpengine", "rtp_inst_pvar", "$avp(RTPENGINE)") #modparam("rtpengine", "rtpengine_retr", 5) #modparam("rtpengine", "queried_nodes_limit", 5) #modparam("rtpengine", "rtpengine_allow_op", 1) #modparam("rtpengine", "hash_table_size", MAX_CONNECTIONS) #modparam("rtpengine", "hash_table_tout", 7200) modparam("tls", "private_key", "<cert path>") modparam("tls", "certificate", "<cert path>") modparam("tls", "ca_list", "<cert path>") # modparam("tls", "ca_list", "<cert path>") modparam("tls", "tls_method", "TLSv1+") ####### Routing Logic ######## If load module lines for TLS are move to near the top of the config file, config will parse and non-SIPS calls will work loadmodule "sl.so" loadmodule "tls.so" But logs will show WARNING: <core> [main.c:2985]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) WARNING: <core> [main.c:2987]: main(): disabling tls... Presumably loading the module before configuring it just gives it default values so the latter config is ignored Any advice on the matter would be appreciated Thanks in advance, Chris __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/4d2d15a7/…> ------------------------------ Message: 4 Date: Wed, 23 Mar 2022 20:16:03 +0000 From: Daniel Paape <dpaape(a)amtelco.com> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] bug in tm module? Message-ID: <CO6PR13MB5321D941E99DE2BA6702E24BC9189(a)CO6PR13MB5321.namprd13.prod.outlook.com> Content-Type: text/plain; charset="iso-8859-1" Hello I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset. So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires an authorization (local database sqlite). After the 401 is sent back, the callback via the module event route is called as expected. All good except that the module event route script fragment is never executed again after the first call. I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route again. thanks for any help, Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/a0311764/…> ------------------------------ Message: 5 Date: Wed, 23 Mar 2022 20:21:26 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] bug in tm module? Message-ID: <PAXPR05MB8558B125061A99B7E6A3D352BF189(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="us-ascii" Hello, If you think there is a bug in the tm module, you could create an issue in our tracker here: https://github.com/kamailio/kamailio/issues/new?assignees=&labels=&template… Please also add some description of the wrong behaviour that you are seeing in the server to it. Thanks, Henning From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Daniel Paape Sent: Wednesday, March 23, 2022 9:16 PM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] bug in tm module? Hello I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset. So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires an authorization (local database sqlite). After the 401 is sent back, the callback via the module event route is called as expected. All good except that the module event route script fragment is never executed again after the first call. I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route again. thanks for any help, Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/61051c0f/…> ------------------------------ Message: 6 Date: Thu, 24 Mar 2022 07:39:26 +0100 From: Arek ?azarczyk <arek(a)lazarczyk.org.pl> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] carrierroute with non-numeric characters Message-ID: <a20892cf-fa42-eefe-2bea-b75a74e3196d(a)lazarczyk.org.pl> Content-Type: text/plain; charset=UTF-8; format=flowed Hi, I'm in a process of long overdue update of kamailio from 4.3 to 5.5, after update i can not start kamailio casue: Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:456]: add_route(): adding prefix TR, prob 1.000000 Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:469]: add_route(): found carrier and domain, now adding route Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_domain.c:166]: add_route_to_tree(): cannot insert route flags into d-trie Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: <core> [dtrie.c:138]: dtrie_insert(): cannot insert non-numerical character Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data I know that string TR is non-numeric data ;) but i have thoes routes deep in my infrastructure logic so i just can not change them to some numbers, is there any way i can resolve this without changeing routes in my whole infrastrucure (almost 50 servers)? Please help :) -- Arek ?azarczyk ------------------------------ Message: 7 Date: Thu, 24 Mar 2022 07:07:35 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Message-ID: <PAXPR05MB8558847862B2480F6CC9E2EABF199(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="utf-8" Hello, give this parameter a try: https://kamailio.org/docs/modules/5.5.x/modules/carrierroute.html#idm173 If it also not works with this parameter set to 128 and its used to work on 4.3, its probably a code regression. Please open an issue on our GitHub tracker about it then. Cheers, Henning -- Henning Westerholt ? https://skalatan.de/blog/ Kamailio services ? https://gilawa.com -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Arek Lazarczyk Sent: Thursday, March 24, 2022 7:39 AM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] carrierroute with non-numeric characters Hi, I'm in a process of long overdue update of kamailio from 4.3 to 5.5, after update i can not start kamailio casue: Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:456]: add_route(): adding prefix TR, prob 1.000000 Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:469]: add_route(): found carrier and domain, now adding route Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_domain.c:166]: add_route_to_tree(): cannot insert route flags into d-trie Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: <core> [dtrie.c:138]: dtrie_insert(): cannot insert non-numerical character Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data I know that string TR is non-numeric data ;) but i have thoes routes deep in my infrastructure logic so i just can not change them to some numbers, is there any way i can resolve this without changeing routes in my whole infrastrucure (almost 50 servers)? Please help :) -- Arek ?azarczyk __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ Message: 8 Date: Thu, 24 Mar 2022 08:42:20 +0100 From: Arek ?azarczyk <arek(a)lazarczyk.org.pl> To: Henning Westerholt <hw(a)gilawa.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Message-ID: <97be0b82-e083-673b-1e2b-f11a62692f34(a)lazarczyk.org.pl> Content-Type: text/plain; charset=UTF-8; format=flowed Thanks, that worked :) , i've must missed that parameter during initial config. BR, A?. W dniu 24.03.2022 o?08:07, Henning Westerholt pisze: > Hello, > > give this parameter a try: > > https://kamailio.org/docs/modules/5.5.x/modules/carrierroute.html#idm173 > > If it also not works with this parameter set to 128 and its used to work on 4.3, its probably a code regression. Please open an issue on our GitHub tracker about it then. > > Cheers, > > Henning > -- Arek ?azarczyk ------------------------------ Message: 9 Date: Thu, 24 Mar 2022 07:58:10 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: Arek ?azarczyk <arek(a)lazarczyk.org.pl>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Message-ID: <PAXPR05MB8558EA2CE9C0F379042C9DBABF199(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="utf-8" Great, thanks for the feedback. Henning -----Original Message----- From: Arek ?azarczyk <arek(a)lazarczyk.org.pl> Sent: Thursday, March 24, 2022 8:42 AM To: Henning Westerholt <hw(a)gilawa.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Thanks, that worked :) , i've must missed that parameter during initial config. BR, A?. W dniu 24.03.2022 o?08:07, Henning Westerholt pisze: > Hello, > > give this parameter a try: > > https://kamailio.org/docs/modules/5.5.x/modules/carrierroute.html#idm1 > 73 > > If it also not works with this parameter set to 128 and its used to work on 4.3, its probably a code regression. Please open an issue on our GitHub tracker about it then. > > Cheers, > > Henning > -- Arek ?azarczyk ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 202, Issue 25 *****************************************

1 0

TOPOH is generating new Call-ID for BYE
by Marrold 24 Mar '22

24 Mar '22

Hi, I have enabled the TOPOH module with mask_callid enabled, but it's generating a new Call-ID for a BYE which gets rejected with a 481 downstream. If I disable the TOPOH module it works as expected. Does anyone know where I should start looking? SIP-wise the initial BYE seems compliant. This is Kamailio 5.5.3 on Debian 11 Thanks Matthew

2 2

core parsing of SIP message failed
by Seyyed Soroosh Hosseinalipour 24 Mar '22

24 Mar '22

ITNOA Description I try to connect Kamailio and Freeswitch together, and initiate call from Zoiper5 soft phone through Kamailio My Kamailio config is below #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases - it can be set many times */ # alias="sip.mydomain.com" /* listen sockets - if none set, Kamailio binds to all local IP addresses * - basic prototype (full prototype can be found in Wiki - Core Cookbook): * listen=[proto]:[localip]:[lport] advertise [publicip]:[pport] * - it can be set many times to add more sockets to listen to */ listen=udp:172.28.72.99:5060 advertise 172.28.72.99:5060 #listen=tcp:172.28.72.99:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_JSONRPC tcp_accept_no_cl=yes #!endif #!ifdef WITH_TLS enable_tls=no /* upper limit for TLS connections */ tls_max_connections=2048 #!endif /* set it to yes to enable sctp and load sctp.so module */ enable_sctp=no ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_JSONRPC loadmodule "xhttp.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "textopsx.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) /* set it to 0 to disable Path handling */ modparam("registrar", "use_path", 1) /* save Path even if not listed in Supported header */ modparam("registrar", "path_mode", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); # handle requests within SIP dialogs # route(WITHINDLG); ### only initial requests (no To tag) # authentication # route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers #remove_hf("Route"); #if (is_method("INVITE|SUBSCRIBE")) { # record_route(); #} # account only INVITEs #if (is_method("INVITE")) { # setflag(FLT_ACC); # do accounting #} # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests #route(PRESENCE); # handle registrations # route(REGISTRAR); #if ($rU==$null) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; #} # dispatch destinations to PSTN #route(PSTN); # user location service #route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE #!ifdef WITH_MSGREBUILD # apply changes in case the request headers or body were modified msg_apply_changes(); #!endif if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-Hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # JSONRPC over HTTP(S) routing #!ifdef WITH_JSONRPC event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } As you can see I just one thing in request_route request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); The 172.28.69.52 is ip of my Freeswitch After I try to call from Zoiper5, I get below error in Kamailio log and Zoiper does not work. 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) Troubleshooting Reproduction * pull official Kamailio docker image kamailio/kamailio 5.4.6-buster * then run below command docker run --net=host --name kamailio \ -v /home/ssoroosh/kamailio:/etc/kamailio \ kamailio/kamailio:5.4.6-buster Debugging Data I do not have any debug data Log Messages Listening on udp: 172.28.72.99 [172.28.72.99]:5060 advertise 172.28.72.99:5060 Aliases: 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2919]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 13(19) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/19 15(21) INFO: ctl [io_listener.c:210]: io_listen_loop(): io_listen_loop: using epoll_lt as the io watch method (auto detected) 2(8) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) SIP Traffic My pcap is attached sip.zip<https://github.com/kamailio/kamailio/files/8326814/sip.zip> Possible Solutions I have not any solution yet. Additional Information * Kamailio Version - output of kamailio -v version: kamailio 5.4.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 * Operating System: Linux master 5.4.0-104-generic #118-Ubuntu SMP Wed Mar 2 19:02:41 UTC 2022 x86_64 GNU/Linux root@master:/home# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster How to resolve this? thanks

2 1

carrierroute with non-numeric characters
by Arek Łazarczyk 24 Mar '22

24 Mar '22

Hi, I'm in a process of long overdue update of kamailio from 4.3 to 5.5, after update i can not start kamailio casue: Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:456]: add_route(): adding prefix TR, prob 1.000000 Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:469]: add_route(): found carrier and domain, now adding route Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_domain.c:166]: add_route_to_tree(): cannot insert route flags into d-trie Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: <core> [dtrie.c:138]: dtrie_insert(): cannot insert non-numerical character Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data I know that string TR is non-numeric data ;) but i have thoes routes deep in my infrastructure logic so i just can not change them to some numbers, is there any way i can resolve this without changeing routes in my whole infrastrucure (almost 50 servers)? Please help :) -- Arek Łazarczyk

2 3

bug in tm module?
by Daniel Paape 23 Mar '22

23 Mar '22

Hello I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset. So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires an authorization (local database sqlite). After the 401 is sent back, the callback via the module event route is called as expected. All good except that the module event route script fragment is never executed again after the first call. I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route again. thanks for any help, Dan

2 1

SIPS Errors on Kamailio
by Christopher Vincent 23 Mar '22

23 Mar '22

Hi, Kamailio / RTPEngine was set up on CentOS 8 running SIP to SIPS and RTP to SDES SRTP conversion. This worked as expected Attempted to duplicate the setup on RHEL but errors were seen. These errors were present on both RHEL 7 / RHEL 8. The errors seen were as below kamailio -c loading modules under config path: /usr/lib64/kamailio/modules/ 0(9165) ERROR: tls [tls_init.c:611]: tls_pre_init(): Unable to set the memory allocation functions 0(9165) ERROR: tls [tls_init.c:613]: tls_pre_init(): libssl current mem functions - m: 0x7f7a77c367a0 r: 0x7f7a77c367f0 f: 0x7f7a77c36770 0(9165) ERROR: tls [tls_init.c:615]: tls_pre_init(): module mem functions - m: 0x7f7a72db7653 r: 0x7f7a72db769f f: 0x7f7a72db76fc 0(9165) ERROR: tls [tls_init.c:617]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(9165) ERROR: <core> [core/sr_module.c:590]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(9165) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 137, column 12-19: failed to load module 0(9165) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 249, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 250, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 251, column 68: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 256, column 39: Can't set module parameter ERROR: bad config file (5 errors) (parsing code: 0) 0(9165) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized The kamailio config was exactly the same as on the CentOS systems and started as below /* Server ports: */ #!substdef "!SIP_PORT!5060!g" #!substdef "!SIPS_PORT!5061!g" /* Listen addresses */ #!substdef "!UDP_LOCAL_ADDR!udp:SERVER_IP_ADDR:SIP_PORT!g" #!substdef "!TCP_LOCAL_ADDR!tcp:SERVER_IP_ADDR:SIPS_PORT!g" /* Server connections: */ #!ifndef MAX_CONNECTIONS #!define MAX_CONNECTIONS 8192 #!endif ##!define WITH_DEBUG /* Transaction and branch flags: FLT_ - per transaction (message) flags FLB_ - per branch flags */ #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_OUT 8 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define KAMAILIODBURL1 "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_TLS enable_tls=1 listen=tls:<ipaddr>:5062 ####### Global Parameters ######### ### LOG Levels: ALERT=-5, BUG=-4, CRIT=-3, ERR=-1, WARN=0, NOTICE=1, INFO=2, DBG=3 #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 /* display memory usage on exit */ mem_summary=15 /* join free memory fragments */ mem_join=1 /* proxy will fork and run in daemon mode */ /* one process will be created for each network interface the proxy listens to and for each protocol (TCP/UDP), multiplied with the value of 'children' parameter */ fork=yes children=8 listen=TCP_LOCAL_ADDR listen=UDP_LOCAL_ADDR /* life time of TCP connection when there is no traffic - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* sip over websockets may not specify a content length header */ tcp_accept_no_cl=yes /* buffer size used for tcp reads, limits the maximum message size (SIP, HTTP) that can be received over tcp */ tcp_rd_buf_size=65536 /* max number of tcp connections */ tcp_max_connections=MAX_CONNECTIONS ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib64/kamailio/modules/" loadmodule "jsonrpcs.so" loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "cfgutils.so" loadmodule "textopsx.so" loadmodule "nathelper.so" loadmodule "uac.so" loadmodule "ipops.so" loadmodule "debugger.so" loadmodule "exec.so" loadmodule "avpops.so" loadmodule "sqlops.so" loadmodule "rtpengine.so" loadmodule "sl.so" loadmodule "tls.so" # ----------------- setting module-specific parameters --------------- # ----- usrloc params ----- # store contacts in memory only modparam("usrloc", "db_mode", 0) # hash size of 16,384 modparam("usrloc", "hash_size", 14) # removes contact if ws disconnects modparam("usrloc", "handle_lost_tcp", 1) modparam("tm|usrloc", "xavp_contact", "ulattrs") # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) #default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- uac params ----- modparam("uac", "restore_mode", "none") # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "max_contacts", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # disable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direction of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- dispatcher params ----- modparam("dispatcher", "db_url", KAMAILIODBURL1) modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:<address>.com") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 3) # ----- pv params ----- modparam("pv", "shvset", "maintenance=i:0") modparam("pv", "shvset", "virtualIP1=i:0") modparam("pv", "shvset", "virtualIP2=i:0") # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is only being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") #modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") #modparam("rtpengine", "force_send_interface", SERVER_IP_ADDR) #modparam("rtpengine", "setid_default", -1) #modparam("rtpengine", "rtp_inst_pvar", "$avp(RTPENGINE)") #modparam("rtpengine", "rtpengine_retr", 5) #modparam("rtpengine", "queried_nodes_limit", 5) #modparam("rtpengine", "rtpengine_allow_op", 1) #modparam("rtpengine", "hash_table_size", MAX_CONNECTIONS) #modparam("rtpengine", "hash_table_tout", 7200) modparam("tls", "private_key", "<cert path>") modparam("tls", "certificate", "<cert path>") modparam("tls", "ca_list", "<cert path>") # modparam("tls", "ca_list", "<cert path>") modparam("tls", "tls_method", "TLSv1+") ####### Routing Logic ######## If load module lines for TLS are move to near the top of the config file, config will parse and non-SIPS calls will work loadmodule "sl.so" loadmodule "tls.so" But logs will show WARNING: <core> [main.c:2985]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) WARNING: <core> [main.c:2987]: main(): disabling tls... Presumably loading the module before configuring it just gives it default values so the latter config is ignored Any advice on the matter would be appreciated Thanks in advance, Chris

3 4

{param.in, name[, delimiter]} error when delimiter is comma
by 韩小仿 23 Mar '22

23 Mar '22

Hi, all Kamailio5.5.4 $var(s) = "a=1;b=2;c=3"; $var(s1) = $(var(s){param.value,b}); // work well $var(s) = "a=1|b=2|c=3"; $var(s1) = $(var(s){param.value,b,|}); // work well // but $var(s) = "a=1,b=2,c=3"; $var(s1) = $(var(s){param.value,b,,}); ERROR: <core> [core/pvapi.c:1131]: pv_parse_spec2(): invalid parsing in [$(var(s){param.value,b,,})] at (4) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 366, column 13-38: Can't get from cache: $(var(s){param.value,b,,}) how to fix this? Thanks! hanxiaofang

2 1

bug in tm module?
by dpaape＠frontier.com 22 Mar '22

22 Mar '22

Hello, I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset.So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires and authorization. After the 401 is sent back, the callback via the module event route is called. All good except that the module event route script fragment is never executed again after the first call.I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route. thanks for any help,Dan

1 0

SERIAL FORKING : call to a group number
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear all, My use case is : I defined a username = 112 (for emergency call) This username doesn't exist in subscribers table and unavailable in location table. The idea is when Sopftphone1 call the group = 112 then Kamailio will forward the call to a group using SERIAL FORKING. => Softphone 1 -> Sends INVITE to 112 -> Kamailio -> FORWARD TO SERIAL FORKING (4x softphone all registered and in location table). The q priority could be hybrid that means 2x softphone could have same priority. My issues : - the q priority forced for each softphone is not taking into account. - the first branch created when send INVITE to 112 is still available. My setup Kamailio is behind a NAT. Kamailio version : sipsecure@kamailio:/$ kamailio -version version: kamailio 5.5.4 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 Kamailio.cfg # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 5sec modparam("tm", "fr_timer", 5000) # default invite retransmission timeout after 1xx: 10sec modparam("tm", "fr_inv_timer", 10000) # This is the name of an XAVP that the t_load_contacts() function uses to store contacts of the destination set and that t_next_contacts() function uses to restore those contacts. modparam("tm", "contacts_avp", "tm_contacts") modparam("tm", "contact_flows_avp", "tm_contact_flows") # modparam("tm|usrloc", "xavp_contact", "tm_contacts") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) route[INVITE]{ route(THREAD_CHECK); route(AUTH); route(RECORD_ROUTE); setflag(FLT_ACC); switch($tU) { case "112": xlog("MY_SEVERITY"," -- Calling Group Serial"); route(FORK_SERIAL); break; case "113": xlog("MY_SEVERITY"," -- Calling Group Parralelle"); route(FORK_PARALLEL); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); break; } exit; } route[FORK_SERIAL]{ t_on_reply("MANAGE_REPLY"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(IPV4V6); route(NATMANAGE); $var(uri1)="sip:9919995@192.168.1.41"; $var(uri2)="sip:9919994@192.168.1.41"; $var(uri3)="sip:9919991@192.168.1.41"; $var(uri4)="sip:9919997@192.168.1.41"; lookup("location","$var(uri4)"); append_branch("$ru","0.2"); lookup("location","$var(uri1)"); append_branch("$ru","0.8"); lookup("location","$var(uri2)"); append_branch("$ru","0.5"); lookup("location","$var(uri3)"); append_branch("$ru","0.5"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL"); #!ifdef WITH_SIPSECURE_DEBUG xlog("MY_SEVERITY"," ROUTE_SERIAL : Relaying \n"); #!endif #Relay (aka Forward) the request t_relay(); break; } failure_route[SERIAL]{ xlog("MY_SEVERITY","At failure route - Trying next destination"); if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } t_on_failure("SERIAL"); t_relay(); } Any highlight will be welcome. Best Regards, -- Youssef

1 0

$ru vs. $fu as r/w pseudo variables
by Juha Heinanen 22 Mar '22

22 Mar '22

If I set $ru in config file and then print $ru to log, the new value is printed. If I do the same with $fu, the old value is printed. I haven't found in Pseudo-Variables Wiki any documentation about this. Does it exist somewhere else? -- Juha

3 9

MESSAGE METHOD - Send message to a softphone - $uac_req
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear, I am expecting to send a MESSAGE METHOD to a softphone supporting MESSAGE. route[SEND_MESSAGE_METHOD]{ $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; $uac_req(callid)=$(mb{s.md5}); $uac_req(hdrs)="Subject: Emergency Alert\r\n"; $uac_req(hdrs)=$uac_req(hdrs) + "Content-Type: text/plain\r\n"; $uac_req(body)="Emergency Message"; $uac_req(evroute)=1; uac_req_send(); } event_route[uac:reply] { xlog("MY_SEVERITY"," -- MESSAGE PROCESSING - received reply code is: $uac_req(evcode)\n");} My use case is to send message from KAMAILIO -> SOFTPHONE kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:192.168.1.29:5066 (physical ip address of the softphone) $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; The sip capture : 2022/03/18 02:20:58.947603 192.168.1.29:5066 -> 192.168.1.41:5066 MESSAGE sip:192.168.1.41:5066@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611 Max-Forwards: 70 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41> Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE User-Agent: MicroSIP/3.20.7 Content-Type: text/plain Content-Length: 3 ;-) 2022/03/18 02:20:58.948822 192.168.1.41:5066 -> 192.168.1.29:5066 SIP/2.0 200 Stored Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611;rport=5066 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41>;tag=cbfe6543b9079987bce3b5b9a0d1195a.b63b503b Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 Now, if I want to use his username@domain kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:9919997@192.168.1.41:5066 The message is sent to kamailio itself not to the softphone 2022/03/18 02:30:39.285927 192.168.1.41:5060 -> 192.168.1.41:5066 MESSAGE sip:9919997@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0 To: <sip:9919997@192.168.1.41:5066> From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Max-Forwards: 70 Content-Length: 17 User-Agent: kamailio (5.5.4 (x86_64/linux)) Subject: Emergency Alert Content-Type: text/plain Emergency Message 2022/03/18 02:30:39.287590 192.168.1.41:5066 -> 192.168.1.41:5060 SIP/2.0 200 Stored Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0;rport=5060;received=192.168.1.41 To: <sip:9919997@192.168.1.41:5066>;tag=cbfe6543b9079987bce3b5b9a0d1195a.76a88393 From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 - finaly, how to relay the message from one softphone to another softphone through kamailio proxy? Best Regards -- Youssef

2 3

Re: [SR-Users] Kamailio : Serial forking
by Paclinio 22 Mar '22

22 Mar '22

Following the example of TM Module. https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forki… Code : modparam("tm", "contacts_avp", "tm_contacts"); > > route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com", "0.5"); > append_branch("sip:c@example.com", "0.5"); > append_branch("sip:d@example.com", "1.0"); > > t_load_contacts(); > > t_next_contacts(); > t_on_failure("serial"); > t_relay(); > break; > } > > failure_route["serial"] > { > if (!t_next_contacts()) { > exit; > } > > t_on_failure("serial"); > t_relay(); > } > My code : failure_route[GroupFailureRoute]{ > t_next_contacts(); > while(!route(GROUP_LOCATION)) > { > if (!t_next_contacts()) { > send_reply("408","Nobody available"); > exit; > } > } > t_on_failure("GroupFailureRoute"); > t_relay(); > } > > route[GROUP_RELAY_SER]{ > append_branch("sip:9919995@192.168.1.41","1"); > append_branch("sip:9919992@192.168.1.41", "0.21"); > append_branch("sip:9919997@192.168.1.41", "0.22"); > append_branch("sip:9919991@192.168.1.41", "0.22"); > t_load_contacts(); > t_next_contacts(); > while(!route(GROUP_LOCATION)) > { > if (!t_next_contacts()) { > send_reply("408","Timeout or nobody available"); > exit; > } > } > t_on_failure("GroupFailureRoute"); > t_relay(); > break; > } > The result : [image: image.png] > > public2_addr > 192.168.1.41 192.168.1.24 > 192.168.1.10 > ──────────┬───────── ──────────┬───────── > ──────────┬───────── ──────────┬───────── > > │ INV (192.168.1.41) │ > | > > │ audio 39776 (g711u) │ > | > > │ > ─────────────────────────────── ──────────────> │ > > │ INV > (192.168.1.41) │ > │ > > │─┐ audio 39776 (g711u) │ > │ > > │<┘ > │ > │ > > │─┐ 407 Proxy Authentication R│ │ > > │<┘ │ > │ > > │ > CANCEL │ > │ > > │ ─────────────────────> │ > │ > > │─┐ ACK > │ > │ > > │<┘ > │ > │ > > │ 487 Request Terminated │ > │ > > │ <────────────────────── │ > │ > > │ > ACK │ > │ > > │ ─────────────────────> │ > │ > > │ 200 OK │ │ > > │ <───────────────────── │ > │ > > │ 100 Trying │ > > │ <────────────────────────────────────────────── │ > > │ 180 Ringing │ > > │ <─────────────────────────────────────────── ── │ > │ 180 Ringing > │ │ > │ > │ <─────────────────── │ > │ > │ > On the Second "wave" of INVITE with the parallel only the first one is correctly changed with the lookup("location") and not the second. Could someone give me some tips? Best regards Thomas <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…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Re: [SR-Users] Kamailio : Parallel Forking
by Paclinio 22 Mar '22

22 Mar '22

Dear, Following the configuration available on the TM Modules documentation. Src : https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forki… request_route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com"); > append_branch("sip:c@example.com"); > append_branch("sip:d@example.com"); > > t_relay(); > break; > } > > My code : request_route{ > $var(uri1)="sip:9919995@192.168.1.41"; > $var(uri2)="sip:9919992@192.168.1.41"; > $var(uri3)="sip:9919991@192.168.1.41"; > $var(uri4)="sip:9919994@192.168.1.41"; > > append_branch ("$var(uri1"); > append_branch ("$var(uri2"); > append_branch ("$var(uri3"); > append_branct_("$var(uri4"); > t_relay(); > break; > } > The server send INVITE to himself. Fixing the problem : request_route{ > $var(uri1)="sip:9919995@192.168.1.41"; > $var(uri2)="sip:9919992@192.168.1.41"; > $var(uri3)="sip:9919991@192.168.1.41"; > $var(uri4)="sip:9919994@192.168.1.41"; > > lookup("location","$var(uri4)"); > append_branch(); > lookup("location","$var(uri1)"); > append_branch(); > lookup("location","$var(uri2)"); > append_branch(); > lookup("location","$var(uri3)"); > t_relay(); > break; > > } > Is this the correct way ? Best regards. Thomas <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…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Re: [SR-Users] MESSAGE METHOD - Send message to a softphone - $uac_req
by Boujraf, Youssef 22 Mar '22

22 Mar '22

Dear, then could I use to retrieve the user from location then set the ruri ? $var(uri1)="sip:9919995@192.168.1.41"; > lookup("location","$var(uri1)"); > > $uac_req(ruri)=$ru; > Best Regards *Youssef *

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Re: [SR-Users] ?==?utf-8?q? Table : address - address_reload?==?utf-8?q? (exported MI function) - expect to call this function in kamailio.cfg to reload in cache the address
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear, When using : sudo kamctl address reload sipsecure@kamailio:/$ sudo kamctl address reload [sudo] password for sipsecure: { "jsonrpc": "2.0", "result": "Reload OK", "id": 73482 } I looked for the command I could use in kamailio.cfg route[JSON_ADDRESS_RELOAD]{ xlog("MY_SEVERITY"," RPC REQUEST : RELOAD ADDRESS TABLE"); jsonrpc_exec('{"jsonrpc": "2.0","method": "permissions.addressReload"}'); exit; } Now I am able to call a rpc command from the code ;-) Best Regards -- Youssef

1 0

Periodic keepalive for WebRTC clients
by Xuo Guoto 22 Mar '22

22 Mar '22

Hello all, I have a [SIP client] <-> Kamailio <-> Asterisk installation. Sip clients are both WebRTC and Android app. WebRTC connects via 443 while Android app connects VIA TLS at port 5061. WebRTC client uses jsSIP library. My Kamailio version is 5.5.3 (x86_64/linux) I had been using asterisk directly to terminate WebRTC clients previously and was using the qualify feature to periodically send OPTIONS packet to all connected clients as a keep alive mechanism. This allowed the client to detect any loss of connectivity and reconnect back to server if there is any loss of connectivity. After moving to Kamailio I am looking for a similar feature, where Kamailio sends OPTIONS (or any other packets) periodically to the WebRTC clients. My attempts so far hasn't been a success. I am aware of couple of options for keepliave in kamailio: 1. nat_keeplaive() I have some what naively copied the example from nat_keepalive() documentation into route[NATDETECT] but that is not working. route[NATDETECT] { force_rport(); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); return; } #!endif if (nat_uac_test(19)) { if (is_method("REGISTER")) { fix_nated_register(); } fix_contact(); } setflag(FLT_NATS); if ((method=="REGISTER" || method=="SUBSCRIBE" || (method=="INVITE" && !has_totag())) && client_nat_test("3")) { nat_keepalive(); } return; } 2. keepalive module This looks like what I want, but I am not sure how to add a new WebRTC connection to the list of endpoints for pinging. This is the handler for xhttp:request, should the ka_add_destination be added here? event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { exit; } } xhttp_reply("404", "Not Found", "", ""); } My search so far has not returned any other means to send options packets to WebRTC clients. Been banging my head around this for couple of days, and any help will be much appreciated. X.

1 0

Dialog module errors with dmq
by Olli Attila 22 Mar '22

22 Mar '22

Hello, I have two Kamailio nodes. Couple of days of ago I enabled dmq to replicate dialogs between these two Kamailio nodes. Now that the dmq is enabled, we are seeing lot of warning and notice messages on kamailio logs related to the dialog module. Is there something I have missed when configuring dlg_manage & dmq replication that might cause this? No calls have been affected negatively by this though. Kamailio 1 node prints lot of this: Mar 4 08:20:25 /usr/sbin/kamailio[14316]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param 'a5d.ebf1' [3418:8126] and call-id 'SDiu98001-dd9c12fea650b9d1408dd672a2c51c9f-v300g00020' Mar 4 08:20:28 /usr/sbin/kamailio[14313]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for BYE with route param 'a5d.ebf1' [3418:8126] and call-id 'SDiu98001-dd9c12fea650b9d1408dd672a2c51c9f-v300g00020' Mar 4 08:21:56 /usr/sbin/kamailio[14315]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param '689.f2f1' [2438:7983] and call-id 'SD8o63c01-47972a87ee1fd8791ddf27b650badcba-v300g00010' Kamailio 2 node prints lot of this: Mar 4 08:20:29 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07b84e980 ref 1) Mar 4 08:20:29 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07bb03360 ref 1) Mar 4 08:21:59 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07b528300 ref 1) (Version: kamailio 5.5.3 (x86_64/linux) 473cef on both kamailio nodes) [root@kamailio-1]# kamcmd dmq.list_nodes { host: 172.19.194.140 port: 5060 proto: * resolved_ip: 172.19.194.140 status: active last_notification: 0 local: 0 } { host: 172.19.194.132 port: 5060 proto: * resolved_ip: 172.19.194.132 status: active last_notification: 0 local: 1 } [root@kamailio-2]# kamcmd dmq.list_nodes { host: 172.19.194.132 port: 5060 proto: * resolved_ip: 172.19.194.132 status: active last_notification: 0 local: 0 } { host: 172.19.194.140 port: 5060 proto: * resolved_ip: 172.19.194.140 status: active last_notification: 0 local: 1 } ### On Kamailio 1: ### #!define DMQ_LOCAL_ADDR "sip:172.19.194.132:5060" #!define DMQ_PEER_ADDR "sip:172.19.194.140:5060" ### On Kamailio 2: ### #!define DMQ_LOCAL_ADDR "sip:172.19.194.140:5060" #!define DMQ_PEER_ADDR "sip:172.19.194.132:5060" ### On Kamailio 1 & 2:### modparam("dmq", "server_address", DMQ_LOCAL_ADDR) modparam("dmq", "notification_address", DMQ_PEER_ADDR) modparam("dialog", "profiles_with_value", "total_dialogs") modparam("dialog", "enable_dmq", 1) route[INITIAL] { ...located at the bottom of the initial route... if(is_method("INVITE") && !has_totag()) { dlg_manage(); } } Cheers, Olli

2 2

About SMSC multi-byte support
by shigeru ishida 22 Mar '22

22 Mar '22

Hi all, I have a little question. I tried SMS over IMS on my smartphones with docker_open5gs (using Kamailio and Open5GS), and the multibyte message (Japanese) was garbled at the recipient smartphone. docker_open5gs URL: https://github.com/herlesupreeth/docker_open5gs Also, docker_open5gs uses Kamailio v5.3. I would be very glad if you could tell me about Kamailio's support for multi-byte SMS. Best regards, --Shigeru

2 2

Kamailio : Parallel Forking
by Paclinio 21 Mar '22

21 Mar '22

I've been trying to do parallel forking but i've a problem. It's when the conversation ends that the user that is in a private network can't send a bye to kamailio and i don't know why. But the on in public network (4G) can. I think it's related to record_route() or Via but I'm not sure. I'm also trying to use lookup on users that are in my domain but the problem is that I don't know how to go through all my append_branch() and then t_relay. My Audio with RTPENGINE works correctly, the only problem is really this SIP method that doesn't go to kamailio. Config : request_route { route(INVITE); exit; } route[INVITE]{ route(RECORD_ROUTE); route(GROUP_RELAY_SER); switch($tU) { case "112": xlog("MY_SEVERITY"," -- Calling Group Serial"); route(GROUP_RELAY_SER); break; case "113": // This number does not exist (It's not a subscriber). xlog("MY_SEVERITY"," -- Calling Group Parallel"); route(GROUP_RELAY_PAR); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); } exit; } route[RECORD_ROUTE]{ # - remove preloaded route headers remove_hf("Route"); if(is_method("INVITE|SUBSCRIBE")) { # record routing for dialog forming requests (in case they are routed) record_route(); } } route[GROUP_RELAY_PAR]{ t_on_reply("MANAGE_REPLY"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(IPV4V6); route(NATMANAGE); append_branch("sip:9919995@192.168.1.24:5066"); append_branch("sip:9919992@192.168.1.26:5066"); append_branch("sip:9919991@192.168.1.10:5067"); // I Would like to do an IP in my domain and then do a lookup but I don't know how to do it. t_on_failure("GroupFailureRoute"); #Relay (aka Forward) the request t_relay(); break; } # User location service route[GROUP_LOCATION] { $avp(oexten) = $rU; if (lookup_branches("location")){ return (1); } else { return (-1); } } failure_route[GroupFailureRoute]{ t_next_contacts(); while(!route(GROUP_LOCATION)){ // Check if the next user (in serial) is logged in) if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } } t_on_failure("GroupFailureRoute"); t_relay(); } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { if(status=~"[12][0-9][0-9]") { if(has_body("application/sdp")){ setbflag(FLB_BRIDGE); } route(NATMANAGE); } } ----------------- Sip Trace : 2022/03/21 18:36:46.469257 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---4108e717927214fd;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53798437 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:36:46.470066 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---4108e717927214fd;rport=9665;received=149.154.195.121 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.6dfd9117 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 1 INVITE Proxy-Authenticate: Digest realm="192.168.1.41", nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/" Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:36:46.500471 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---4108e717927214fd;rport Max-Forwards: 70 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.6dfd9117 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 1 ACK Content-Length: 0 2022/03/21 18:36:46.549993 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75020835629f7238;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/",uri="sip:113@Public_IP ;transport=TCP",response="c8654099b37ca083ace2b245f22ebc71",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53798437 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:36:46.556454 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665;received=149.154.195.121 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:36:46.558035 192.168.1.41:55008 -> 192.168.1.41:5060 INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Record-Route: <sip:Public_IP;transport=tcp;lr> Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.558616 192.168.1.41:5060 -> 192.168.1.41:55008 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.0;i=6;rport=55008;received=192.168.1.41 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.4ecd297d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Proxy-Authenticate: Digest realm="192.168.1.41", nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/" Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:36:46.559281 192.168.1.41:55008 -> 192.168.1.41:5060 ACK sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.0;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.4ecd297d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:36:46.559660 192.168.1.41:5060 -> 192.168.1.24:5066 INVITE sip:9919995@192.168.1.24:5066 SIP/2.0 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.559717 192.168.1.41:5060 -> 192.168.1.26:5066 INVITE sip:9919992@192.168.1.26:5066 SIP/2.0 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.559744 192.168.1.41:5060 -> 192.168.1.10:5067 INVITE sip:9919991@192.168.1.10:5067 SIP/2.0 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.566320 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 100 Trying To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:46.580166 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 180 Ringing To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:46.580487 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:46.640281 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:36:46.658312 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:36:46.740855 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:46.741955 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:46.769015 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:46.769313 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:48.945301 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 165 v=0 o=Z 0 1 IN IP4 192.168.1.26 s=Z c=IN IP4 192.168.1.26 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv 2022/03/21 18:36:48.947154 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 200 OK Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 245 v=0 o=Z 0 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36572 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:36573 2022/03/21 18:36:48.953782 192.168.1.41:5060 -> 192.168.1.24:5066 CANCEL sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL Content-Length: 0 Reason: SIP;cause=200;text="Call completed elsewhere" 2022/03/21 18:36:48.953817 192.168.1.41:5060 -> 192.168.1.10:5067 CANCEL sip:9919991@192.168.1.10:5067 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL Content-Length: 0 Reason: SIP;cause=200;text="Call completed elsewhere" 2022/03/21 18:36:48.964605 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 487 Request Terminated To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:48.964766 192.168.1.41:5060 -> 192.168.1.24:5066 ACK sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:36:48.966945 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 200 OK To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:48.989036 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:36:48.989036 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:36:48.989268 192.168.1.41:5060 -> 192.168.1.10:5067 ACK sip:9919991@192.168.1.10:5067 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:36:49.021133 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---28ec75a236e47eb1;rport Max-Forwards: 70 Route: <sip:Public_IP;transport=tcp;r2=on;lr> Route: <sip:Public_IP:5060;r2=on;lr> Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:36:49.021618 192.168.1.41:5060 -> 192.168.1.26:5066 ACK sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.064a5839a4d237786d792a9b52b0a37c.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---28ec75a236e47eb1;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:37:23.350203 149.154.195.121:9665 -> 192.168.1.41:5060 BYE sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport Max-Forwards: 70 Route: <sip:Public_IP;transport=tcp;r2=on;lr> Route: <sip:Public_IP:5060;r2=on;lr> Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/",uri=" sip:9919992@192.168.1.26:5066 ",response="a7440eb01dfb200b92cc1320808251da",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:37:23.357139 192.168.1.41:5060 -> 192.168.1.26:5066 BYE sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK14b4.ece7c52456dd1d2ba5e46ca9a10b6cee.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/",uri=" sip:9919992@192.168.1.26:5066 ",response="a7440eb01dfb200b92cc1320808251da",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:37:23.359207 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK14b4.ece7c52456dd1d2ba5e46ca9a10b6cee.0;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport=9665 Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 2022/03/21 18:37:23.359668 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 200 OK Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport=9665 Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 Thank you. <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…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Kamailio : Serial forking
by Paclinio 21 Mar '22

21 Mar '22

I've been trying to do serial forking, I've configured something that is working but when I'm looking on github and other sources it seems like my code is a little too complicated and I would like to know if there is a simpler config that I can use. To have more precision, I'm using RTPENGINE and I need to be able to go through NAT. My Config : request_route { route(INVITE); exit; } route[INVITE]{ route(RECORD_ROUTE); route(GROUP_RELAY_SER); switch($tU) { case "112": // This number does not exist (It's not a subscriber). xlog("MY_SEVERITY"," -- Calling Group Serial"); route(GROUP_RELAY_SER); break; case "113": xlog("MY_SEVERITY"," -- Calling Group Parralelle"); route(GROUP_RELAY_PAR); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); } exit; } route[RECORD_ROUTE]{ # - remove preloaded route headers remove_hf("Route"); if(is_method("INVITE|SUBSCRIBE")) { # record routing for dialog forming requests (in case they are routed) record_route(); } } route[GROUP_RELAY_SER]{ t_on_reply("MANAGE_REPLY"); route(IPV4V6); route(NATMANAGE); (RTP Routing is working correctly) append_branch("sip:9919995@192.168.1.41","1"); append_branch("sip:9919992@192.168.1.41", "0.2"); append_branch("sip:75575757@192.168.1.41", "0.4"); append_branch("sip:9919991@192.168.1.41", "0.5"); t_load_contacts(); t_next_contacts(); while(!route(GROUP_LOCATION)) { if (!t_next_contacts()) { send_reply("408","Timeout or nobody available"); exit; } } t_on_failure("GroupFailureRoute"); #Relay (aka Forward) the request t_relay(); break; } # User location service route[GROUP_LOCATION] { $avp(oexten) = $rU; if (lookup_branches("location")){ return (1); } else { return (-1); } } failure_route[GroupFailureRoute]{ t_next_contacts(); while(!route(GROUP_LOCATION)){ if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } } t_on_failure("GroupFailureRoute"); t_relay(); } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { if(status=~"[12][0-9][0-9]") { if(has_body("application/sdp")){ setbflag(FLB_BRIDGE); } route(NATMANAGE); } } --------------------------------- Here's The SIP Trace : 2022/03/21 18:28:45.265009 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:112@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53721112 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:28:45.274652 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport=9665;received=149.154.195.121 To: <sip:112@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.064b0d7d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 1 INVITE Proxy-Authenticate: Digest realm="192.168.1.41", nonce="Yji2+WI4tc0uQkJ6GbQ4I9VeSUQmR6Gk" Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:28:45.335876 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:112@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport Max-Forwards: 70 To: <sip:112@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.064b0d7d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 1 ACK Content-Length: 0 2022/03/21 18:28:45.385686 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:112@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji2+WI4tc0uQkJ6GbQ4I9VeSUQmR6Gk",uri="sip:112@Public_IP ;transport=TCP",response="54837810a66298e4542486bb6ab8ca04",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53721112 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:28:45.418871 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665;received=149.154.195.121 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:28:45.421797 192.168.1.41:5066 -> 192.168.1.24:5066 INVITE sip:9919995@192.168.1.24:5066 SIP/2.0 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53721112 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36502 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36503 2022/03/21 18:28:45.435659 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 100 Trying To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:45.449138 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 180 Ringing To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:45.455211 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066 ;alias=192.168.1.24~5066~1> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:55.445916 192.168.1.41:5066 -> 192.168.1.10:50561 INVITE sip:9919991@192.168.1.10:50561;transport=TCP;rinstance=034d5c22d5466df2 SIP/2.0 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53721112 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36502 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36503 2022/03/21 18:28:55.446096 192.168.1.41:5066 -> 192.168.1.24:5066 CANCEL sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL Content-Length: 0 2022/03/21 18:28:55.457941 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 487 Request Terminated To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:55.458874 192.168.1.41:5066 -> 192.168.1.24:5066 ACK sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:28:55.459394 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 200 OK To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:55.536475 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:28:55.758738 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:28:55.763664 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561 ;transport=TCP;alias=192.168.1.10~50561~2> To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:29:05.761477 192.168.1.41:5066 -> 192.168.1.26:32821 INVITE sip:9919992@192.168.1.26:5066;rinstance=791ce762698e7422;transport=tcp SIP/2.0 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53721112 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36502 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36503 2022/03/21 18:29:05.761547 192.168.1.41:5066 -> 192.168.1.10:50561 CANCEL sip:9919991@192.168.1.10:50561;transport=TCP;rinstance=034d5c22d5466df2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL Content-Length: 0 2022/03/21 18:29:05.799795 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:05.800776 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 487 Request Terminated Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:05.801104 192.168.1.41:5066 -> 192.168.1.10:50561 ACK sip:9919991@192.168.1.10:50561;transport=TCP;rinstance=034d5c22d5466df2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:29:05.844860 192.168.1.26:32821 -> 192.168.1.41:5066 SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:29:06.045373 192.168.1.26:32821 -> 192.168.1.41:5066 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066;transport=tcp> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:29:06.046528 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066 ;transport=tcp;alias=192.168.1.26~32821~2> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:29:11.055497 192.168.1.26:32821 -> 192.168.1.41:5066 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066;transport=tcp> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 165 v=0 o=Z 0 1 IN IP4 192.168.1.26 s=Z c=IN IP4 192.168.1.26 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv 2022/03/21 18:29:11.061482 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 200 OK Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066 ;transport=tcp;alias=192.168.1.26~32821~2> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 245 v=0 o=Z 0 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36518 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:36519 2022/03/21 18:29:11.144532 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:9919992@192.168.1.26:5066;transport=tcp;alias=192.168.1.26~32821~2 SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---f49bbd15731e8f8d;rport Max-Forwards: 70 Route: <sip:Public_IP;transport=tcp;r2=on;lr> Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:29:11.152633 192.168.1.41:5066 -> 192.168.1.26:32821 ACK sip:9919992@192.168.1.26:5066;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.422a25f74a8ad6c743d75a28764bc8b7.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---f49bbd15731e8f8d;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:29:20.514914 192.168.1.26:32821 -> 192.168.1.41:5066 BYE sip:9919993@100.79.133.159:5060;transport=tcp;alias=149.154.195.121~9665~2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.26:5066 ;branch=z9hG4bK-524287-1---404c84aa134e2a04 Max-Forwards: 70 Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066;transport=tcp> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:20.517617 192.168.1.41:5060 -> 149.154.195.121:9665 BYE sip:9919993@100.79.133.159:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bKf831.2578b48bc3463ab1aed37eba97036001.0;i=5 Via: SIP/2.0/TCP 192.168.1.26:5066 ;rport=32821;branch=z9hG4bK-524287-1---404c84aa134e2a04 Max-Forwards: 69 Contact: <sip:9919992@192.168.1.26:5066 ;transport=tcp;alias=192.168.1.26~32821~2> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:20.584230 149.154.195.121:9665 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.41:5060 ;branch=z9hG4bKf831.2578b48bc3463ab1aed37eba97036001.0;i=5 Via: SIP/2.0/TCP 192.168.1.26:5066 ;rport=32821;branch=z9hG4bK-524287-1---404c84aa134e2a04 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:29:20.584431 192.168.1.41:5066 -> 192.168.1.26:32821 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.26:5066 ;rport=32821;branch=z9hG4bK-524287-1---404c84aa134e2a04 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 Thank you. Thomas <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…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Can Kamailio send an ACK to a final response?
by Marrold 21 Mar '22

21 Mar '22

Hello, I'm exploring using Kamailio as a dumb edge proxy, and as such I want to keep things as lean as possible. In some scenarios I would like the "core" proxies to signal to the edge that a request should be dropped without a response. This could be with a custom header in the reply or perhaps a 444 response ( nginx uses this internally to drop a request without responding ) I have discovered that if I drop the response in the reply_route, the edge proxy doesn't send an ACK and keeps retransmitting the request until it times out, at which point it sends a 408 back to the UA. Is there any way to ACK the response from the core and destroy the transaction? Thanks Matthew

3 2

Kamailio WebRTC Call Not Working
by Ankit Jayswal 21 Mar '22

21 Mar '22

Hi, I have configured Kamailio to register a SIP user over websocket. SIP users are getting registered over webphone like JSSIP fine but not able to establish the call scenario below. >From Twilio Trunk - INVITE - KAMAILIO - ASTERISK - play hello world - Dial SIP user 101@Kamailio_IP - Rings 101 and Answer - Hangup. Here, 101 is registered with Kamailio IP, and I am sending a call from Twilio SIP trunk to Kamailio server IP. Asterisk is on port 5080 and Kamailio is on port 5090 both are on the same host 45.118.163.244. This above call scenario is NOT working with Webphones registered via JSSIP. I think the second call leg from Asterisk -> Kamailio is not getting set up properly and ACK is not getting confirmed here, call connects for some time and no voice transmits and then disconnects. The same call scenario is working with users registered at Softphone like Zoiper perfectly fine. No idea why it is not working with a webphone. I have attached kamailio.cfg, Kamailio log and Asterisk's pjsip logs here. So, please guide me on this as soon as possible. -- Thanks & Regards, *Ankit Jayswal* | Specialist - Software Development

1 0

pseudo var for From and To sip uri port
by Patrick Karton 20 Mar '22

20 Mar '22

Hello, is there a pseudo variable for setting only sip uri port for From and To Headers like $rp for R-URI ?

1 0

dialog module issues (BYE Request issued) when topos is enabled
by Patrick Karton 19 Mar '22

19 Mar '22

i am using dialog and topos module. but i have an issue i think with topos module. When topos module is disabled all the call flow works well but when i enable topos module i have this weird behaviour : after Caller sends BYE and complete the call , i receive later a BYE Request issued by the dialog module towards Caller and Callee. Does anyone already had that issue ? modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 1024) modparam("dialog", "rr_param", "oovdlg") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "db_update_period", 30) modparam("dialog", "wait_ack", 1) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "end_timeout", 180) modparam("dialog", "default_timeout", 3600) modparam("dialog", "noack_timeout", 90) modparam("dialog", "early_timeout", 300) modparam("dialog", "dlg_extra_hdrs", "X-OOV-Hint: Dlg-hit\r\n") modparam("db_text", "db_mode", 0) modparam("db_text", "emptystring", 0) modparam("db_text", "file_buffer_size", 8192) modparam("db_text", "max_result_rows", 1000) modparam("topos", "storage", "db") modparam("topos", "db_url", "text:///home/sbc/database") modparam("topos", "mask_callid", 0) modparam("topos", "sanity_checks", 0) modparam("topos", "contact_mode", 0)

1 0

DMQ version compatability
by Ben Kaufman 19 Mar '22

19 Mar '22

Daniel, In a recent thread you stated the following regarding DMQ compatibility between versions: "if you do dmq replication between kamailio systems running different major versions, then it is likely to get memory leaks due to replication of data and most probably cannot be fixed. This is because internal structures of modules (also dmq commands) can change, practically what an instance does is not ensured to happen on the other instance. Just for example, from my mind, htable got some changes during past releases, dmq also has significant enhancements by getting support for more transport protocols." Is it possible to get a bit more clarification on this, particularly as it relates to DMQ_USRLOC? I'm hoping to do rolling updates, but don't want to end up in a situation where I've caused myself more problems than I've solved. Ben Kaufman

3 2

help to configure topos module configuration
by Patrick Karton 18 Mar '22

18 Mar '22

Caller <--> (internal_IP) Kamailio (external_IP) <---> Callee I have this set up of kamailio with 2 interfaces. i use topos module for topology hiding and what i want to do is to send external_IP in Contact Header of request relayed to Callee and internal_IP in Contact Header of response relayed to Caller. i noticed that : when i use topos module with contact_mode to 0 all Record-Route and Caller Via Header are removed and thats great. but i dont find way to put 2 different ip addresses in Request and Response Contact Header? when i use contact_mode to 2 Record-Route and Caller Via Header are not anymore removed thats bad for topology hiding. And its seems we can not use $xavu(_tps_=>contact_host) parameter of module to put 2 different ip addresses in Request and Response Contact Header. What i want is : - remove all Record-Route and Caller Via Header for request like when contact_mode to 0 - and put 2 different ip addresses in Request and Response Contact Header is it possible to achieve it ? Thanks.

2 2

Upgrading from version 5.2 to 5.4 - any unpublished words of wisdom?
by Kevin Greene 18 Mar '22

18 Mar '22

Hi: I have a 5.2 version of Kamailio running and I was thinking to get it upgraded to the 5.4 branch. The 5.2 is working well but I have heard the 5.4 is a bit more stable. Thus the question becomes, is it worth the risk to upgrade and what should I look out for. Thanks. Kevin

2 1

Dialog module and DMQ
by Olle E. Johansson 18 Mar '22

18 Mar '22

Hi! The documentation on the DMQ support in the dialog module is a bit short, so I will ask here for clarifications: If I want to enable a rough limit on concurrent calls - will all servers share a $DLG_count ? Will get_profile_size() return the number of dialogs belonging to a profile on ALL servers? Thank you! /Olle

1 0

Kamailio as stateless failover proxy server and nat
by ppq＠tuta.io 17 Mar '22

17 Mar '22

Hello, I need some help in creating a stateless failover balancer for two asterisk servers using kamailio. I followed in the steps of an old project here: https://txlab.wordpress.com/tag/xlab1/ Kamailio is running in a multihomed host, with public ip facing the internet and private ip facing the asterisk servers. I have partial issues with nat traversal when the external device is also behind nat. All works when the device initates a call/dialog, however when the asterisk server tries to send an "OPTIONS" keep alive to the end device, Kamailio receives the packet from asterisk but tries to send it to the private IP address of the device instead of the public ip:port. Same goes for calls. My kamailio.cfg is below. I am missing something obvious here, but can't find it. Thanks === CUT==== #!KAMAILIO ################################################################ # # This is a front-end SIP proxy configuration for the xlab1 project # See for more details: https://txlab.wordpress.com/tag/xlab1/ # The configuration requires Kamailio version 3.3.x # # This work is licensed under a Creative Commons Attribution 3.0 License # Author: Stanislav Sinyagin <ssinaygin(a)k-open.com> # ################################################################ #!define WITH_DEBUG ####### Defined Values ######### #### define DBURL "text:///opt/sipfe/etc/kamailio_db" #!define DBURL "text:///etc/kamailio/kamailio_db" #!define BACKEND_NET4 10.10.10.0/24 #!define BACKEND_NET6 2a02:200:8:100::/64 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=2 log_stderror=yes #!else debug=1 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=2 auto_aliases=no #listen=udp:195.x.x.78:5060 # to avoid simple attacks, use unconventional port #port=5060 # this makes it listen on all ipv6 addresses auto_bind_ipv6=0 dns_try_ipv6=no # let kamailio choose the right source IP address mhomed=3 # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ## Hardening sip_warning=0 server_signature=no user_agent_header="User-Agent: Kamailio Failover Balancer" ####### Modules Section ######## ## NOTE: ## mpath="/opt/sipfe/lib64/kamailio/modules_k/:/opt/sipfe/lib64/kamailio/modules/" loadmodule "db_text.so" loadmodule "jsonrpcs.so" #loadmodule "mi_fifo.so" loadmodule "tm.so" loadmodule "kex.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" #loadmodule "mi_rpc.so" loadmodule "path.so" loadmodule "domain.so" loadmodule "dispatcher.so" loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") modparam("path", "use_received", 1) modparam("domain", "db_url", DBURL) modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") #modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## route { if (!sanity_check()) { exit; } if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); xlog("L_NOTICE", "$si $rm $ru -- too many hops\n"); exit; } if (af==INET) { force_rport(); if (src_ip != BACKEND_NET4) { # SIP request packet client->backend if( !loose_route() ) { if (!lookup_domain("$td", "dattr_")) { xlog("L_ERR", "$si $rm $ru -- domain \"$td\" is not " + "found in domain table\n"); drop(); } if (!defined $avp(dattr_routeset)) { xlog("L_ERR", "$si $rm $ru -- attribute \"routeset\" is " + "undefined for domain $td\n"); drop(); } if( !ds_select_dst(4000 + $avp(dattr_routeset), "1") ) { drop(); } } if (nat_uac_test("19")) { xlog("L_INFO", "Routing NAT in-dialog $rm from $fu to $du\n"); if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } } add_path_received(); #rtpproxy_manage("cwei"); rtpproxy_manage("cw"); record_route(); } else { # SIP request packet backend->client # Invites from backend contain Route field and it should be used # to reach the registered client loose_route(); #rtpproxy_manage("cwie"); rtpproxy_manage("cw"); record_route(); } } else { # IPv6: no NAT traversal, just pure routing if (src_ip != BACKEND_NET6) { # SIP request packet client->backend if( !loose_route() ) { if (!lookup_domain("$td", "dattr_")) { xlog("L_ERR", "$si $rm $ru -- domain \"$td\" is not " + "found in domain table\n"); drop(); } if (!defined $avp(dattr_routeset)) { xlog("L_ERR", "$si $rm $ru -- attribute \"routeset\" is " + "undefined for domain $td\n"); drop(); } if( !ds_select_dst(6000 + $avp(dattr_routeset), "1") ) { drop(); } } add_path_received(); record_route(); } else { # SIP request packet backend->client record_route(); } } forward(); } onreply_route { if (af==INET) { if(src_ip != BACKEND_NET4) { # SIP reply packet client->backend #rtpproxy_manage("cwie"); rtpproxy_manage("cw"); fix_nated_contact(); } else { # SIP reply packet backend->client #rtpproxy_manage("cwei"); rtpproxy_manage("cw"); } } return(1); } # Local Variables: # mode: sh # sh-indent-after-if: 0 # indent-tabs-mode: nil # tab-width: 4 # End: ====CUT=====

1 0

kamctl monitor : statistics not displayed ?
by Youssef Boujraf 17 Mar '22

17 Mar '22

Dear I am tying to monitor the kamailio with "kamctl monitor" and I got this error. -e [cycle #: 1; if constant make sure server lives] -e Kamailio Runtime Details: kamailio 5.5.4 (x86_64/linux) now: Wed Mar 16 12:44:43 2022 up_since: Wed Mar 16 12:39:51 2022 uptime: 292 -e Transaction Statistics: /usr/sbin/kamctl: 245: column: not found -e Stateless Server Statistics: /usr/sbin/kamctl: 259: column: not found -e UsrLoc Statistics: /usr/sbin/kamctl: 273: column: not found -e Core Statistics: /usr/sbin/kamctl: 287: column: not found -e Shared Memory Statistics: /usr/sbin/kamctl: 301: column: not foundAny idea Best Regards -- Youssef BOUJRAF

2 6

kamctl or kamcmd way to enable/disable sections of configs
by Joshua Young 17 Mar '22

17 Mar '22

#!define MYSECTION" route("1"); #!ifdef MYSECTION route("2"); #!endif I have declarations like this in my kamailio.cfg file, and was curious if there was a way to use kamctl or kamcmd to enable or disable the defined value in a live system. Or if there is another approach/mechanism I'm unaware of the "Serial Forking Based on Q Value" stuff and use it in other parts, but I am looking for a general way to manually manipulate routing on live system using cli commands if possible... Thanks, Joshua

3 2

Kamailio in docker container with TLS enabled using Let's Encrypt
by Giovanni Tommasini 17 Mar '22

17 Mar '22

Hi guys, starting from Fred article https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ we move step forward creating this small PoC to deploy Kamailio in docker container with TLS enabled using Let's Encrypt https://github.com/evoseed/kamailio-tls-letsencrypt What do you think? Any feedback? Any mistakes to correct or improvements to make? Thanks a lot Have a good evening Giovanni Tommasini *evoseed s.r.l.* phone: +39 340 11 28 800 mail: giovanni.tommasini(a)evoseed.io site: evoseed.io address: Via Lucrezio 13, Trieste Italy

2 2

sanity from uri check
by Juha Heinanen 16 Mar '22

16 Mar '22

I have in config: modparam("sanity", "uri_checks", 3) /* RURI, From */ and then config call sanity_check() on incoming requests: if (!sanity_check()) { xnotice("Request from <$var(src_ip)> has invalid syntax\n"); exit; }; When incoming request has: From: sipp <sip:@192.168.255.95:5062>;tag=1 no notice is printed to syslog. >From RFC 3261: SIP-URI = "sip:" [ userinfo ] hostport uri-parameters [ headers ] userinfo = ( user / telephone-subscriber ) [ ":" password ] "@" user = 1*( unreserved / escaped / user-unreserved ) My understanding of that is that there needs to be at least one character before @ sign. -- Juha

2 29

properly removing First Via header received
by Patrick Karton 16 Mar '22

16 Mar '22

i want to remove the First Via Header received from uac to kamailio. so in my script i use remove_hf("Via") but in the relayed request i see that ";received=x.x.x.x;rport=xxxx" is left without any header before it. i think internally kamailio adds a lump before i call the remove_hf function. how to do it properly ? Thanks

2 2

Forking for scripts?
by Matthias Urlichs 16 Mar '22

16 Mar '22

Hello, Is it possible to run a background Python script within Kamailio? I.e. call "fork_process" and then a non-terminating "python_exec()" from the child? -- -- regards -- -- Matthias Urlichs

2 2

Question about a future 5.6 release
by Stefan-Cristian Mititelu 16 Mar '22

16 Mar '22

Hi, I want to ask you if there is an approximate timeline for a kamailio 5.6 release? We are planning an upgrade on our systems and are weighing different options. Thank you Stefan

2 1

typo in documentation
by David Villasmil 15 Mar '22

15 Mar '22

Hello all, I was trying to implement db_text and the documentation states it should be something like modparam("usrloc|auth_db", "db_url", "db_text:///tmp/kamailiodb") but that doesn't work, it's modparam("usrloc|auth_db", "db_url", "text:///tmp/kamailiodb") so this line should probably be updated: https://github.com/kamailio/kamailio/blob/master/src/modules/db_text/doc/db… Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 2

load to htable from flat file
by David Villasmil 15 Mar '22

15 Mar '22

Hello guys, Is this possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 5

RTPEngine inject DTMF
by Bendersky 15 Mar '22

15 Mar '22

Hello Everyone, I am trying inject DTMF and getting error message. ERROR:rtpengine:rtpe_function_call: proxy replied with error: No SSRC context present for DTMF injection I have full call rtpengine debug which I can provide upon request. Rtpengine version rtpengine-9.5.4.1-9.fc32.x86_64 Volga629

1 0

clean an easy way to get sip uri of contact header
by Patrick Karton 15 Mar '22

15 Mar '22

hello, i want a generic way to get the sip uri in contact header. for example if Contact: Sipp <sip:6565@ip;ov=65;gid=45>;tid=700 i want a way to get both <sip:6565@ip;ov=65;gid=45> (without header param) and <sip:6565@ip;ov=65;gid=45>;tid=700 (with header param) Thanks.

3 3

limit the length of Via branch parameter
by Patrick Karton 15 Mar '22

15 Mar '22

Hello, i would like to know if there is a way or option to limit the length of the value of branch parameter in Via header ? Thanks.

2 3

Table : address - address_reload (exported MI function) - expect to call this function in kamailio.cfg to reload in cache the address
by Youssef Boujraf 14 Mar '22

14 Mar '22

Dear, I use the address table to ENABLE UAC doesn't registererd in subscriber table just for emergency call then after the BYE to I remove it from the address table. I 'd like to purge the UAC from address table BUT NEED to RELOAD in CACHE the address table. To use the "kamctl address relaod" works BUT I want to reload the address table from my Kamailio.cfg (route). I found the function "address_reload" ( https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528769) but I have an error when trying to start kamailio. Best regards, -- Youssef BOUJRAF | Managing Director Direct Line: +32 (0)2 880 33 30 Mobile: +32 (0)477 94 86 89 Dubai Mobile : +971 (0)55 9928871 Fax: +32 (0)2 318 82 21 Skype: y_boujraf Email: yboujraf(a)by-research.be Web : www.by-systems.be/ BY-SYSTEMS sprl| Avenue Charles Plisnier 13 | B-1070 Anderlecht | Belgium | TVA: BE 0819.832.617______________________________________________________________________________ This communication, issued by BY-SYSTEMS SPRL, is confidential and we do not waive confidentiality by mistransmission. If you have received it by mistake please notify the sender immediately by reply and then delete it permanently from your system without copying it, using it for any purposes or disclosing its contents to any other person. Thank you for your cooperation. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of BY-SYSTEMS SPRL. Emails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, lost or destroyed, or contain viruses. Anyone who communicates with us by email is taken to accept these risks.

2 1

WebSocket ping is not being sent
by Xuo Guoto 14 Mar '22

14 Mar '22

Hello list, I have kamailio 5.5.3 (x86_64/linux) configured as an SBC in front of asterisk to front end webrtc clients. Browser clients running JsSIP 3.7.0 connect to kamailio via webrtc and kamailio sends the traffic back to asterisk using SIP. rtpengine is configured to relay the RTP packets. I need to send websocket pings every 10 seconds to the browser clients. I have the following config in kamailio: modparam("websocket", "keepalive_mechanism", 1) modparam("websocket", "keepalive_timeout", 10) modparam("websocket", "keepalive_interval", 2) modparam("websocket", "keepalive_processes", 2) modparam("websocket", "verbose_list", 1) If my understanding of the documentation is correct, I do not need to do any additional configuration. But looking at the websocket traffic, I do not see any ping packets in websocket. I am also using sipdump module to capture all packets sent and received by kamailio and the ping packets are not visible there either. What could be wrong here? Do I need to do any additional config? Thanks for every help! X.

3 5

Any clever ways to strip down a user portion of an RURI?
by Brooks Bridges 12 Mar '22

12 Mar '22

Have a need to strip out various parameters from the user portion of the RURI and/or the To (npdi, etc). Does anyone have any clever methods for doing so that don't involve just brute forcing through the individual parameters? Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

3 4

Interconnect with Zoom
by Matthias Urlichs 11 Mar '22

11 Mar '22

Hi, has anybody managed to connect to Zoom with Kamailio, ideally using rtpengine? I'm getting error 500 when I call them … -- -- regards, -- -- Matthias Urlichs

3 4

Kamailio + RADCLI
by Marrold 11 Mar '22

11 Mar '22

Hi, We're currently testing Kamailio 5.5.3 on Debian 11 compiled with RADCLI and we're experiencing an issue with radius authentication: Feb 23 08:54:22 redacted /sbin/kamailio[817172]: WARNING: <script>: RADIUS auth failed for redacted (IP:redacted:5060) Feb 23 08:54:22 redacted /sbin/kamailio[817173]: radcli: rc_avpair_new: rc_avpair_new: no attribute 1/1 in dictionary Feb 23 08:54:22 redacted2 /sbin/kamailio[817173]: ERROR: auth_radius [sterman.c:204]: add_cisco_vsa(): unable to add Cisco-AVPair attribute I don't believe this is directly a Kamailio issue but enabling debug logging hasn't revealed any clues ( The log lines remain the same without additional info ) This was working previously on 5.5.X on Debian 9 Has anyone experienced this before or have any suggestions on where to start looking? Thanks

2 2

Kamailio 5.5 TLS critical log in Debian 11
by mayamatakeshi 11 Mar '22

11 Mar '22

I'm testing with this kamailio version: $ kamcmd version kamailio 5.5.4 (x86_64/linux) 54c9df on: $ cat /etc/issue Debian GNU/Linux 11 \n \l with: $ openssl version OpenSSL 1.1.1k 25 Mar 2021 Whenever I start kamailio I get 3199 log lines like this: 2022-03-10T10:04:39.700676+09:00 lab002201-flip-server /usr/local/src/git/kamailio-5.5/src/kamailio[261703]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f8eb57f7848), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_free(323) - ignoring I searched the issues at github and found: https://github.com/kamailio/kamailio/issues/2560 https://github.com/kamailio/kamailio/issues/2912 So I tried to start kamailio with --atexit=no but those logs remain. With and without using --atexit=no there is no crash and all my tests are OK. But should I be worried to put this in production?

3 3

add a=fmtp: if it does not exist in sdp
by Krzysztof Drewicz 11 Mar '22

11 Mar '22

Hello i try to fix one endpoint, it does neet do have a a=rtpmap:116 telephone-event/8000 a=fmtp:116 0-15 sometimes i got: a=rtpmap:115 telephone-event/8000 and sometimes: a=rtpmap:114 telephone-event/8000 a=fmtp:114 0-15 so - should i use txtops and search for fmtp, and use search_append_body ? ifsearch_body(re) does not find 'a=fmtp:95' (95..127 - dynamic payload here), or am I reinventing the wheel again ? Any hints? -- Krzysztof Drewicz Senior Infrastructure Administrator CLUDO | ul. Grochowska 306/308, 03-840 Warszawa t+48221223977 kdrewicz(a)cludo.pl | www.cludo.pl

3 3

Diversion Header & Header management - $hdrc
by Brandon Armstead 10 Mar '22

10 Mar '22

Thanks for your time in advance! When dealing with multiple headers in different formats, i.e. Diversion: <sip:foo>, <sip:bar> $hdrc(Diversion) returns 1 header instead of 2. Is this expected behavior? What is the best way to manage all scenarios of possible multiple header values? It seems currently there is no easy way to iterate over ALL diversion headers, including scenarios like the following below: Diversion: <sip:abc> or Diversion: <sip:abc>, <sip:def> or Diversion: <sip:abc> Diversion: <sip:def>, <sip:ghi> etc. - Brandon

2 1

Kamailio 5.4.7 crash
by Igor Olhovskiy 10 Mar '22

10 Mar '22

Hello! My Kamailio 5.4.7 on CentOS 7 [root@pbxphone-0866-fe-1 kamailio_crash]# kamailio -V version: kamailio 5.4.7 (x86_64/linux) a19fe1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: a19fe1 compiled on 20:40:47 Nov 22 2021 with gcc 4.8.5 [root@pbxphone-0866-fe-1 kamailio_crash]# uname -a Linux pbxphone-0866-fe-1.company.domain 3.10.0-1160.15.2.el7.x86_64 #1 SMP Wed Feb 3 15:06:38 UTC 2021 x86_64 x86_64 x86_64 GNU/Linux crashed with following backtraces (3 core files) [root@pbxphone-0866-fe-1 kamailio_crash]# gdb kamailio core.kamailio.222447.pbxphone-0866-fe-1.company.name.1646903896 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-120.el7 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /root/kamailio_crash/kamailio...done. warning: exec file is newer than core file. [New LWP 222447] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamail'. Program terminated with signal 6, Aborted. #0 0x00007f04d6e16387 in raise () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install audit-libs-2.8.5-4.el7.x86_64 bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.176-5.el7.x86_64 glibc-2.17-325.el7_9.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-51.el7_9.x86_64 libacl-2.2.51-15.el7.x86_64 libattr-2.4.46-13.el7.x86_64 libcap-2.22-11.el7.x86_64 libcap-ng-0.7.5-4.el7.x86_64 libcom_err-1.42.9-19.el7.x86_64 libdb-5.3.21-25.el7.x86_64 libffi-3.0.13-19.el7.x86_64 libgcc-4.8.5-44.el7.x86_64 libselinux-2.5-15.el7.x86_64 libstdc++-4.8.5-44.el7.x86_64 libunistring-0.9.3-9.el7.x86_64 libuuid-2.23.2-65.el7_9.1.x86_64 libxml2-2.9.1-6.el7_9.6.x86_64 lm_sensors-libs-3.4.0-8.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 net-snmp-agent-libs-5.7.2-49.el7_9.1.x86_64 net-snmp-libs-5.7.2-49.el7_9.1.x86_64 nspr-4.32.0-1.el7_9.x86_64 nss-3.67.0-4.el7_9.x86_64 nss-softokn-freebl-3.67.0-3.el7_9.x86_64 nss-util-3.67.0-1.el7_9.x86_64 openssl-libs-1.0.2k-24.el7_9.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-299.el7_9.x86_64 popt-1.13-16.el7.x86_64 python-libs-2.7.5-90.el7.x86_64 rpm-libs-4.11.3-48.el7_9.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-19.el7_9.x86_64 (gdb) bt full #0 0x00007f04d6e16387 in raise () from /lib64/libc.so.6 No symbol table info available. #1 0x00007f04d6e17a78 in abort () from /lib64/libc.so.6 No symbol table info available. #2 0x000000000041e330 in sig_alarm_abort (signo=14) at main.c:693 __FUNCTION__ = "sig_alarm_abort" #3 <signal handler called> No symbol table info available. #4 0x00007f04d6ed8e27 in syscall () from /lib64/libc.so.6 No symbol table info available. #5 0x00000000004fa848 in futex_get (lock=0x7f042fe11ca0) at core/mem/../futexlock.h:121 v = 1 i = 0 #6 0x0000000000520dda in tcpconn_send_put (c=0x7f042fe11c98, buf=0x7ffd8be90c90 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"..., len=302, send_flags=...) at core/tcp_main.c:2422 tmp = 0x0 fd = 0 response = {140726950757744, 0} n = 32516 do_close_fd = 1 rest_buf = 0x12fe11e10 <Address 0x12fe11e10 out of bounds> t_buf = 0x0 rest_len = 32516 t_len = 794570504 resp = 139658225205256 t_send_flags = {f = 1296, blst_imask = 35817} fd_cache_e = 0x0 use_fd_cache = 1 __FUNCTION__ = "tcpconn_send_put" #7 0x00000000005203e7 in tcp_send (dst=0x7ffd8be90a20, from=0x0, buf=0x7ffd8be90c90 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"..., len=302) at core/tcp_main.c:2323 c = 0x7f042fe11c98 ip = {af = 2, len = 4, u = {addrl = {1765950034, 4294967295}, addr32 = {1765950034, 0, 4294967295, 0}, addr16 = {16978, 26946, 0, 0, 65535, 65535, 0, 0}, addr = "RBBi\000\000\000\000\377\377\377\377\000\000\000"}} port = 46050 fd = 4 response = {139655957499112, 0} n = 0 con_lifetime = 9600 try_local_port = 5061 rest_buf = 0x69d0c8 <dns_a_get_he+86> "\353CH\213Eؾ\001" t_buf = 0x7ffd8be906e0 "\020\a\351\213\375\177" rest_len = 0 t_len = 0 resp = 139655957577728 t_send_flags = {f = 19200, blst_imask = 180} __FUNCTION__ = "tcp_send" #8 0x00007f04b2aa0ceb in ul_ka_send (kamsg=0x7ffd8be90c80, kadst=0x7ffd8be90a20) at ul_keepalive.c:271 __FUNCTION__ = "ul_ka_send" #9 0x00007f04b2aa0b58 in ul_ka_urecord (ur=0x7f042fca3340) at ul_keepalive.c:239 uc = 0x7f042fca3458 kabuf = "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name >;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"... ---Type <return> to continue, or q <return> to quit--- kabuf_len = 302 kamsg = { s = 0x7ffd8be90c90 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"..., len = 302} vaddr = {s = 0x7f04b6e74008 "tone.company.name", len = 12} vport = {s = 0x7f04b6e740f8 "5061", len = 4} sdst = {s = 0x7f042fca3770 "sip:82.66.66.105:46050;transport=tls", len = 36} sproto = {s = 0x81bb4b "TLS", len = 3} duri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x7f042fca3774 "82.66.66.105:46050;transport=tls", len = 12}, port = {s = 0x7f042fca3781 "46050;transport=tls", len = 5}, params = {s = 0x7f042fca3787 "transport=tls", len = 13}, sip_params = {s = 0x7f042fca3787 "transport=tls", len = 13}, headers = {s = 0x0, len = 0}, port_no = 46050, proto = 3, type = SIP_URI_T, flags = (unknown: 0), transport = {s = 0x7f042fca3787 "transport=tls", len = 13}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x7f042fca3791 "tls", len = 3}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} dproto = 3 '\003' he = 0xb88fa0 <he.5288> ssock = 0x7f04b6e73d98 idst = {send_sock = 0x7f04b6e73d98, to = {s = {sa_family = 2, sa_data = "\263\342RBBi\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 58035, sin_addr = {s_addr = 1765950034}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 58035, sin6_flowinfo = 1765950034, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = { 0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 2, __ss_padding = "\263\342RBBi", '\000' <repeats 111 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 3 '\003', proto_pad0 = 0 '\000', proto_pad1 = 0} bcnt = 1 aortype = 0 i = 5 tv = {tv_sec = 1646903836, tv_usec = 846405} tnow = 1646903836 __FUNCTION__ = "ul_ka_urecord" #10 0x00007f04b2a6bb99 in mem_timer_udomain (_d=0x7f042f9c5388, istart=0, istep=1) at udomain.c:1234 ptr = 0x7f042fca3340 t = 0x1ea00000000 i = 28 #11 0x00007f04b2a9dd0e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:966 res = 0 ptr = 0x7f042f9c5288 #12 0x00007f04b2a7ac5a in destroy () at usrloc_mod.c:498 __FUNCTION__ = "destroy" #13 0x00000000005934ee in destroy_modules () at core/sr_module.c:772 t = 0x7f04b6e83828 foo = 0x7f04b6e83338 __FUNCTION__ = "destroy_modules" #14 0x000000000041cda7 in cleanup (show_status=1) at main.c:569 memlog = 0 __FUNCTION__ = "cleanup" #15 0x000000000041e682 in shutdown_children (sig=15, show_status=1) at main.c:712 __FUNCTION__ = "shutdown_children" #16 0x000000000042172a in handle_sigs () at main.c:812 chld = 0 chld_status = 139 any_chld_stopped = 1 memlog = 798824048 __FUNCTION__ = "handle_sigs" #17 0x000000000042babb in main_loop () at main.c:1845 i = 4 pid = 222489 si = 0x0 si_desc = "udp receiver child=3 sock=128.141.18.117:5060 ( tone.company.name:5060)\000\000\000\000\000\000\000\000\200S/\004\177\000\000\000\000\000\000\000\000\000\000\220Q\234/\004\177\000\000\200\034\351\2---Type <return> to continue, or q <return> to quit--- 13\375\177\000\000\000R\234/\004\177\000\000B\227Z\257\004\177\000\000\300\023\006\267\004\177\000" nrprocs = 4 woneinit = 1 __FUNCTION__ = "main_loop" #18 0x000000000043423e in main (argc=9, argv=0x7ffd8be921f8) at main.c:2942 cfg_stream = 0xf6a040 c = -1 r = 0 tmp = 0x7ffd8be935cc "" tmp_len = 1472 port = 960 proto = 32516 ahost = 0x0 aport = 0 options = 0x7d7ef8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1413124693 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffd8be920b0 p = 0xf0b5ff <Address 0xf0b5ff out of bounds> st = {st_dev = 20, st_ino = 16553497, st_nlink = 2, st_mode = 16832, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1646819986, tv_nsec = 228520360}, st_mtim = {tv_sec = 1646903531, tv_nsec = 836398702}, st_ctim = {tv_sec = 1646903531, tv_nsec = 836398702}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 392 times>... option_index = 0 long_options = {{name = 0x7da22f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d56d4 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7da234 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7da23a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7da240 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7da249 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7da253 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7da25d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7da268 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7da271 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7da27c "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x7da282 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" 2nd CORE backtrace [root@pbxphone-0866-fe-1 kamailio_crash]# gdb kamailio core.kamailio.222460.pbxphone-0866-fe-1.company.domain.1646903834 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-120.el7 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /root/kamailio_crash/kamailio...done. warning: exec file is newer than core file. [New LWP 222460] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 0x00007f04af578716 in tls_complete_init (c=0x7f042fe11c98) at tls_server.c:236 236 data->ssl = SSL_new(dom->ctx[process_no]); Missing separate debuginfos, use: debuginfo-install audit-libs-2.8.5-4.el7.x86_64 bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.176-5.el7.x86_64 glibc-2.17-325.el7_9.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-51.el7_9.x86_64 libacl-2.2.51-15.el7.x86_64 libattr-2.4.46-13.el7.x86_64 libcap-2.22-11.el7.x86_64 libcap-ng-0.7.5-4.el7.x86_64 libcom_err-1.42.9-19.el7.x86_64 libdb-5.3.21-25.el7.x86_64 libffi-3.0.13-19.el7.x86_64 libgcc-4.8.5-44.el7.x86_64 libselinux-2.5-15.el7.x86_64 libstdc++-4.8.5-44.el7.x86_64 libunistring-0.9.3-9.el7.x86_64 libuuid-2.23.2-65.el7_9.1.x86_64 libxml2-2.9.1-6.el7_9.6.x86_64 lm_sensors-libs-3.4.0-8.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 net-snmp-agent-libs-5.7.2-49.el7_9.1.x86_64 net-snmp-libs-5.7.2-49.el7_9.1.x86_64 nspr-4.32.0-1.el7_9.x86_64 nss-3.67.0-4.el7_9.x86_64 nss-softokn-freebl-3.67.0-3.el7_9.x86_64 nss-util-3.67.0-1.el7_9.x86_64 openssl-libs-1.0.2k-24.el7_9.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-299.el7_9.x86_64 popt-1.13-16.el7.x86_64 python-libs-2.7.5-90.el7.x86_64 rpm-libs-4.11.3-48.el7_9.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-19.el7_9.x86_64 (gdb) bt full #0 0x00007f04af578716 in tls_complete_init (c=0x7f042fe11c98) at tls_server.c:236 dom = 0x7f042fbcccf8 data = 0x7f042f96dec0 cfg = 0x7f042f96dec0 state = S_TLS_CONNECTING sname = 0x0 srvid = 0x0 __FUNCTION__ = "tls_complete_init" #1 0x00007f04af578d07 in tls_fix_connection_unsafe (c=0x7f042fe11c98) at tls_server.c:297 __FUNCTION__ = "tls_fix_connection_unsafe" #2 0x00007f04af580784 in tls_h_encode_f (c=0x7f042fe11c98, pbuf=0x7ffd8be909e0, plen=0x7ffd8be909d8, rest_buf=0x7ffd8be909e8, rest_len=0x7ffd8be909dc, send_flags=0x7ffd8be909c0) at tls_server.c:772 n = 0 offs = 0 ssl = 0x1 tls_c = 0x7f042fe11ca0 wr_buf = "\026\003\001\002\000\001\000\001\374\003\003\276,'P^\252j\027\021S\351\252\021\342(~\347\216\351\332Hh!\vb\024\361\324+\003\177\a\000\000\254\300\060\300,\300(\300$\300\024\300\n\000\245\000\243\000\241\000\237\000k\000j\000i\000h\000\071\000\070\000\067\000\066\000\210\000\207\000\206\000\205\300\062\300.\300*\300&\300\017\300\005\000\235\000=\000\065\000\204\300/\300+\300'\300#\300\023\300\t\000\244\000\242\000\240\000\236\000g\000@ \000?\000>\000\063\000\062\000\061\000\060\000\232\000\231\000\230\000\227\000E\000D\000C\000B\300\061\300-\300)\300%\300\016\300\004\000\234\000<\000/\000\226\000A\300\022\300\b\000\026\000\023\000\020\000\r\300\r\300\003"... rd = {buf = 0x18 <Address 0x18 out of bounds>, pos = 22930096, used = 1771061403, size = -1947662592} wr = {buf = 0x4b6e74a40 <Address 0x4b6e74a40 out of bounds>, pos = -1226359400, used = 32516, size = -1562640382} ssl_error = 0 err_src = 0x7f04af5afc09 "TLS write:" ip_buf = "\000\000\000\000\000\000\000\000\230\034\341/\004\177\000\000\000\000\000\000\000\000\000\000\370@ \347\266\001\000\000\000\340\b\351\213\375\177\000\000f\211Q\000\000\000\000\000\000\000\000\000\001\000\000\000\060\000\000\000\060\000\000" buf = 0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain >;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"... len = 308 x = 4 __FUNCTION__ = "tls_h_encode_f" #3 0x000000000051e729 in tcp_send (dst=0x7ffd8be90d60, from=0x0, buf=0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"..., len=308) at core/tcp_main.c:2093 c = 0x7f042fe11c98 ip = {af = 2, len = 4, u = {addrl = {1765950034, 4294967295}, addr32 = {1765950034, 0, 4294967295, 0}, addr16 = {16978, 26946, 0, 0, 65535, 65535, 0, 0}, addr = "RBBi\000\000\000\000\377\377\377\377\000\000\000"}} port = 46050 fd = 22 response = {139655966192064, 0} n = 0 con_lifetime = 9600 try_local_port = 5061 rest_buf = 0x0 t_buf = 0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain >;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"... rest_len = 0 t_len = 308 resp = 140726950756352 t_send_flags = {f = 0, blst_imask = 0} __FUNCTION__ = "tcp_send" #4 0x00007f04b2aa0ceb in ul_ka_send (kamsg=0x7ffd8be90fc0, kadst=0x7ffd8be90d60) at ul_keepalive.c:271 __FUNCTION__ = "ul_ka_send" #5 0x00007f04b2aa0b58 in ul_ka_urecord (ur=0x7f042fca3340) at ul_keepalive.c:239 uc = 0x7f042fca3458 kabuf = "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-2c-31dd20---Type <return> to continue, or q <return> to quit--- 1c-6229c21a"... kabuf_len = 308 kamsg = { s = 0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"..., len = 308} vaddr = {s = 0x7f04b6e74008 "tone.company.domain", len = 12} vport = {s = 0x7f04b6e740f8 "5061", len = 4} sdst = {s = 0x7f042fca3770 "sip:82.66.66.105:46050;transport=tls", len = 36} sproto = {s = 0x81bb4b "TLS", len = 3} duri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x7f042fca3774 "82.66.66.105:46050;transport=tls", len = 12}, port = {s = 0x7f042fca3781 "46050;transport=tls", len = 5}, params = {s = 0x7f042fca3787 "transport=tls", len = 13}, sip_params = {s = 0x7f042fca3787 "transport=tls", len = 13}, headers = {s = 0x0, len = 0}, port_no = 46050, proto = 3, type = SIP_URI_T, flags = (unknown: 0), transport = {s = 0x7f042fca3787 "transport=tls", len = 13}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x7f042fca3791 "tls", len = 3}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} dproto = 3 '\003' he = 0xb88fa0 <he.5288> ssock = 0x7f04b6e73d98 idst = {send_sock = 0x7f04b6e73d98, to = {s = {sa_family = 2, sa_data = "\263\342RBBi\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 58035, sin_addr = {s_addr = 1765950034}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 58035, sin6_flowinfo = 1765950034, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = { 0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 2, __ss_padding = "\263\342RBBi", '\000' <repeats 111 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 3 '\003', proto_pad0 = 0 '\000', proto_pad1 = 0} bcnt = 1 aortype = 0 i = 5 tv = {tv_sec = 1646903834, tv_usec = 680904} tnow = 1646903834 __FUNCTION__ = "ul_ka_urecord" #6 0x00007f04b2a6bb99 in mem_timer_udomain (_d=0x7f042f9c5388, istart=0, istep=1) at udomain.c:1234 ptr = 0x7f042fca3340 t = 0x7f0429d60470 <str_record_route_col> i = 28 #7 0x00007f04b2a9dd0e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:966 res = 0 ptr = 0x7f042f9c5288 #8 0x00007f04b2a7b019 in ul_core_timer (ticks=1791845, param=0x0) at usrloc_mod.c:527 __FUNCTION__ = "ul_core_timer" #9 0x00000000004f1e69 in compat_old_handler (ti=28669522, tl=0x7f042f82e060, data=0x7f042f82e060) at core/timer.c:980 t = 0x7f042f82e060 #10 0x00000000004f29a5 in slow_timer_main () at core/timer.c:1105 n = 12 ret = 4294967295 tl = 0x7f042f82e060 i = 308 __FUNCTION__ = "slow_timer_main" #11 0x000000000042a707 in main_loop () at main.c:1765 i = 4 pid = 0 si = 0x0 si_desc = "udp receiver child=3 sock=128.141.18.117:5060 (tone.company.domain:5060)\000\000\000\000\000\000\000\000\200S/\004\177\000\000\000\000\000\000\000\000\000\000\220Q\234/\004\177\000\000\200\034\351\213\375\177\000\000\000R\234/\004\177\000\000B\227Z\257\004\177\000\000\300\023\006\267\004\177\000" nrprocs = 4 woneinit = 1 __FUNCTION__ = "main_loop" #12 0x000000000043423e in main (argc=9, argv=0x7ffd8be921f8) at main.c:2942 cfg_stream = 0xf6a040 ---Type <return> to continue, or q <return> to quit--- c = -1 r = 0 tmp = 0x7ffd8be935cc "" tmp_len = 1472 port = 960 proto = 32516 ahost = 0x0 aport = 0 options = 0x7d7ef8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1413124693 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffd8be920b0 p = 0xf0b5ff <Address 0xf0b5ff out of bounds> st = {st_dev = 20, st_ino = 16553497, st_nlink = 2, st_mode = 16832, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1646819986, tv_nsec = 228520360}, st_mtim = {tv_sec = 1646903531, tv_nsec = 836398702}, st_ctim = {tv_sec = 1646903531, tv_nsec = 836398702}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 392 times>... option_index = 0 long_options = {{name = 0x7da22f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d56d4 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7da234 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7da23a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7da240 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7da249 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7da253 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7da25d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7da268 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7da271 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7da27c "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x7da282 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" 3rd CORE: [root@pbxphone-0866-fe-1 kamailio_crash]# gdb kamailio core.kamailio.276013.pbxphone-0866-fe-1.company.domain.1646903518 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-120.el7 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /root/kamailio_crash/kamailio...done. warning: exec file is newer than core file. [New LWP 276013] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 0x00007f7cfc309478 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at ../../core/atomic/atomic_x86.h:224 224 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) Missing separate debuginfos, use: debuginfo-install audit-libs-2.8.5-4.el7.x86_64 bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.176-5.el7.x86_64 glibc-2.17-325.el7_9.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-51.el7_9.x86_64 libacl-2.2.51-15.el7.x86_64 libattr-2.4.46-13.el7.x86_64 libcap-2.22-11.el7.x86_64 libcap-ng-0.7.5-4.el7.x86_64 libcom_err-1.42.9-19.el7.x86_64 libdb-5.3.21-25.el7.x86_64 libffi-3.0.13-19.el7.x86_64 libgcc-4.8.5-44.el7.x86_64 libselinux-2.5-15.el7.x86_64 libstdc++-4.8.5-44.el7.x86_64 libunistring-0.9.3-9.el7.x86_64 libuuid-2.23.2-65.el7_9.1.x86_64 libxml2-2.9.1-6.el7_9.6.x86_64 lm_sensors-libs-3.4.0-8.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 net-snmp-agent-libs-5.7.2-49.el7_9.1.x86_64 net-snmp-libs-5.7.2-49.el7_9.1.x86_64 nspr-4.32.0-1.el7_9.x86_64 nss-3.67.0-4.el7_9.x86_64 nss-softokn-freebl-3.67.0-3.el7_9.x86_64 nss-util-3.67.0-1.el7_9.x86_64 openssl-libs-1.0.2k-24.el7_9.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-299.el7_9.x86_64 popt-1.13-16.el7.x86_64 python-libs-2.7.5-90.el7.x86_64 rpm-libs-4.11.3-48.el7_9.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-19.el7_9.x86_64 (gdb) bt full #0 0x00007f7cfc309478 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at ../../core/atomic/atomic_x86.h:224 ret = 0 #1 0x00007f7cfc3094c4 in futex_get (lock=0x0) at ../../core/futexlock.h:99 v = 32636 i = 1024 #2 0x00007f7cfc310461 in wsconn_get (id=992) at ws_conn.c:452 id_hash = 992 wsc = 0x100000001 __FUNCTION__ = "wsconn_get" #3 0x00007f7cfc324d71 in ws_frame_transmit (evp=0x7ffcf5f31830) at ws_frame.c:728 wsev = 0x7ffcf5f31870 frame = {fin = 1, rsv1 = 0, rsv2 = 0, rsv3 = 0, opcode = 1, mask = 0, payload_len = 301, masking_key = "\000\000\000", payload_data = 0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., wsc = 0x0} __FUNCTION__ = "ws_frame_transmit" #4 0x00000000004f3762 in sr_event_exec (type=11, evp=0x7ffcf5f31830) at core/events.c:296 ret = 0 i = 50304 #5 0x0000000000544edf in wss_send (dst=0x7ffcf5f31c30, buf=0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., len=301) at core/tcp_main.c:5209 port = 50304 ip = {af = 2, len = 4, u = {addrl = {3232828610, 140724434836000}, addr32 = {3232828610, 0, 4126349856, 32764}, addr16 = {3266, 49329, 0, 0, 6688, 62963, 32764, 0}, addr = "\302\f\261\300\000\000\000\000 \032\363\365\374\177\000"}} from = 0x0 local_addr = {s = {sa_family = 19200, sa_data = "\264\000\000\000\000\000\360\034\363\365\374\177\000"}, sin = {sin_family = 19200, sin_port = 180, sin_addr = {s_addr = 0}, sin_zero = "\360\034\363\365\374\177\000"}, sin6 = {sin6_family = 19200, sin6_port = 180, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "\360\034\363\365\374\177\000\000\000\000\272\001}\177\000", __u6_addr16 = {7408, 62963, 32764, 0, 0, 442, 32637, 0}, __u6_addr32 = {4126350576, 32764, 28966912, 32637}}}, sin6_scope_id = 0}, sas = {ss_family = 19200, __ss_padding = "\264\000\000\000\000\000\360\034\363\365\374\177\000\000\000\000\272\001}\177\000\000\000\000\000\000\000\000\000\000\360\030\363\365\374\177\000\000\310\320i\000\000\000\000\000\300}\271\001}\177\000\000\000\000\000\000\000\000\000\000\340\271A\000\004\000\000\000\360\362\200\000\000\000\000\000\377\377\377\377", '\000' <repeats 12 times>, " \031\363\365\374\177\000\000\200\200\025\251}\177\000\000\000\000\000\000\000\000\000", __ss_align = 140724434836720}} con = 0x7f7d03e4c500 wsev = {type = 11, buf = 0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., len = 301, id = 992} evp = {data = 0x7ffcf5f31870, obuf = {s = 0x0, len = 0}, rcv = 0x0, dst = 0x0, req = 0x0, rpl = 0x0, rplcode = 0, mode = 0} ret = 0 __FUNCTION__ = "wss_send" #6 0x00007f7d84a54c47 in ul_ka_send (kamsg=0x7ffcf5f31e90, kadst=0x7ffcf5f31c30) at ul_keepalive.c:259 __FUNCTION__ = "ul_ka_send" #7 0x00007f7d84a54b58 in ul_ka_urecord (ur=0x7f7d04fcb8f8) at ul_keepalive.c:239 uc = 0x7f7d04fcba10 kabuf = "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain >;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"... kabuf_len = 301 kamsg = { s = 0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., len = 301} vaddr = {s = 0x7f7d88e28a40 "tone.company.domain", len = 12} vport = {s = 0x7f7d88e28b30 "8089", len = 4} sdst = {s = 0x7f7d051d1318 "sip:194.12.177.192:50304;transport=ws", len = 37} sproto = {s = 0x81bb5d "WS", len = 2} duri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x7f7d051d131c "194.12.177.192:50304;transport=ws", len = 14}, port = {s = 0x7f7d051d132b "50304;transport=ws", len = 5}, params = {s = 0x7f7d051d1331 "transport=ws", len = 12}, sip_params = {s = 0x7f7d051d1331 "transport=ws", len = 12}, headers = {s = 0x0, len = 0}, port_no = 50304, proto = 5, type = SIP_URI_T, ---Type <return> to continue, or q <return> to quit--- flags = (unknown: 0), transport = {s = 0x7f7d051d1331 "transport=ws", len = 12}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x7f7d051d133b "ws", len = 2}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} dproto = 5 '\005' he = 0xb88fa0 <he.5288> ssock = 0x7f7d88e287d0 idst = {send_sock = 0x7f7d88e287d0, to = {s = {sa_family = 2, sa_data = "Ā\302\f\261\300\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 32964, sin_addr = {s_addr = 3232828610}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 32964, sin6_flowinfo = 3232828610, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = { 0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 2, __ss_padding = "Ā\302\f\261\300", '\000' <repeats 111 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 5 '\005', proto_pad0 = 0 '\000', proto_pad1 = 0} bcnt = 1 aortype = 0 i = 5 tv = {tv_sec = 1646903518, tv_usec = 435121} tnow = 1646903518 __FUNCTION__ = "ul_ka_urecord" #8 0x00007f7d84a1fb99 in mem_timer_udomain (_d=0x7f7d01990460, istart=0, istep=1) at udomain.c:1234 ptr = 0x7f7d04fcb8f8 t = 0x1ea00000000 i = 0 #9 0x00007f7d84a51d0e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:966 res = 0 ptr = 0x7f7d01990360 #10 0x00007f7d84a2ec5a in destroy () at usrloc_mod.c:498 __FUNCTION__ = "destroy" #11 0x00000000005934ee in destroy_modules () at core/sr_module.c:772 t = 0x7f7d88e37828 foo = 0x7f7d88e37338 __FUNCTION__ = "destroy_modules" #12 0x000000000041cda7 in cleanup (show_status=1) at main.c:569 memlog = 0 __FUNCTION__ = "cleanup" #13 0x000000000041e682 in shutdown_children (sig=15, show_status=1) at main.c:712 __FUNCTION__ = "shutdown_children" #14 0x000000000041f1f7 in handle_sigs () at main.c:743 chld = 0 chld_status = 0 any_chld_stopped = 0 memlog = 26855240 __FUNCTION__ = "handle_sigs" #15 0x000000000042babb in main_loop () at main.c:1845 i = 4 pid = 276058 si = 0x0 si_desc = "udp receiver child=3 sock=128.141.18.117:5060 (tone.company.domain:5060)\000\000\000\000\000\000\000\000\300N\001}\177\000\000\000\000\000\000\000\000\000\000h\002\231\001}\177\000\000\220.\363\365\374\177\000\000\330\002\231\001}\177\000\000B\327U\201}\177\000\000\300S\001\211}\177\000" nrprocs = 4 woneinit = 1 __FUNCTION__ = "main_loop" #16 0x000000000043423e in main (argc=9, argv=0x7ffcf5f33408) at main.c:2942 cfg_stream = 0xf06040 c = -1 r = 0 tmp = 0x7ffcf5f34eff "" tmp_len = 1472 ---Type <return> to continue, or q <return> to quit--- port = 960 proto = 32637 ahost = 0x0 aport = 0 options = 0x7d7ef8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 626051482 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffcf5f332c0 p = 0xf0b5ff "" st = {st_dev = 20, st_ino = 16553497, st_nlink = 2, st_mode = 16832, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1614937626, tv_nsec = 968330983}, st_mtim = {tv_sec = 1646814601, tv_nsec = 999994878}, st_ctim = {tv_sec = 1646814601, tv_nsec = 999994878}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 392 times>... option_index = 0 long_options = {{name = 0x7da22f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d56d4 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7da234 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7da23a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7da240 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7da249 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7da253 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7da25d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7da268 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7da271 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7da27c "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x7da282 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" Is it a right place to report or better to make an issue on github? Thanks in advance! -- Best regards, Igor

1 0

Kamailio v5.4.8 Released
by Daniel-Constantin Mierla 09 Mar '22

09 Mar '22

Hello, Kamailio SIP Server v5.4.8 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.8. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.4 being release out of it. For more details about version 5.4.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/03/kamailio-v5-4-8-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 Kamailio Advanced Training - Online March 28-31, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

Releasing Kamailio v.5.4.8
by Daniel-Constantin Mierla 09 Mar '22

09 Mar '22

Hello, I am planning to release a new version from branch 5.4, respectively 5.4.8, sometime next week, likely on Wednesday or Thursday (Mar 9/10, 2022). If anyone is aware of issues not yet reported to bug tracker or missing backports, report them in order to try to get the fixes in this release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online March 28-31, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

2 3

rtpengine - SRTP <> RTP missing a=crypto
by Rhys Hanrahan 09 Mar '22

09 Mar '22

Hi Everyone, I have Kamailio sitting between MS Teams and Asterisk, and using rtpengine to terminate SRTP on Kamailio so that all my internal traffic is unencrypted. My current config works fine for inbound calls where I initiate the INVITE and Teams responds, but if Teams sends the INVITE I am having an issue where SRTP cannot finish negotiating. Non SRTP calls work fine with RTPEngine as well, so it's just the RTP to SRTP I am struggling with. According to this I believe I must pass a=crypto in response to the INVITE which also has a=crypto: https://www.dialogic.com/-/media/1f8b54b43087407d9c2b38846c5c2cb5.ashx?h=40… You can see that in the initial invite from Teams, I get RTP/SAVP with a=crypto, but I do not send one in my OK response after 183 Session In Progress. As below - I am wondering if it's because not all audio channels seem to be getting swapped to SAVP? I'd like to do a generic SRTP <> RTP bridge config (I've tried below). However, I am not 100% sure on how to detect when to swap between AVP and SAVP, so I've also tried just doing rtpengine_manage() and relying on other code to swap between SAVP or AVP *only* when going to/from Teams to keep it simple. I also tried both with and without "replace-origin replace-session-connection ICE=remove" but I still get the same behaviour in all cases. Any advice appreciated, as this is my first time dealing with SRTP (and rtpengine). Feeling very stuck. Thanks! branch_route[MANAGE_BRANCH] { ... route(NATMANAGE); route(HANDLE_SRTP); } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } route(HANDLE_SRTP); } route[HANDLE_SRTP] { if (!has_body("application/sdp")) { return; } rtpengine_manage(); return; # As a test, just do rtpengine_manage() and set SAVP/AVP elsewhere. Same behaviour. # Handle bridging of RTP and SRTP # Inbound traffic to SBC should be converted from SRTP to RTP if (proto==TLS) { rtpengine_manage("RTP/AVP"); # Outbound traffic destined to a TLS destination should be converted from RTP to SRTP } else if ($ru =~ "transport=tls") { rtpengine_manage("RTP/SAVP"); } } # INVITE from teams rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP"); # INVITE to teams rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/SAVP"); INVITE sip:+614xxxx@rh.sbc-syd-01.teams.xxxx:5061;user=phone;transport=tls SIP/2.0^M ... v=0^M o=- 57931 0 IN IP4 127.0.0.1^M s=session^M c=IN IP4 52.113.76.53^M b=CT:10000000^M t=0 0^M m=audio 51398 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118^M c=IN IP4 52.113.76.53^M a=rtcp:51399^M a=ice-ufrag:C8ss^M a=ice-pwd:2bV9D6GcXF5f8m0px/wufQD/^M a=rtcp-mux^M a=candidate:1 1 UDP 2130706431 52.113.76.53 51398 typ srflx raddr 10.0.32.179 rport 51398^M a=candidate:1 2 UDP 2130705918 52.113.76.53 51399 typ srflx raddr 10.0.32.179 rport 51399^M a=candidate:2 1 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=candidate:2 2 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=label:main-audio^M a=mid:1^M a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:geUHLB1mshmnI5hN83bnO57Hbdm2i7dD14sDAnpA|2^31^M a=sendrecv^M a=rtpmap:104 SILK/16000^M a=rtpmap:9 G722/8000^M a=rtpmap:103 SILK/8000^M a=rtpmap:111 SIREN/16000^M a=fmtp:111 bitrate=16000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:97 RED/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=rtpmap:13 CN/8000^M a=rtpmap:118 CN/16000^M a=ptime:20^M I correctly convert to/from RTP/AVP and RTP/SAVP for the 183 Session in progress. It is RTP/SAVP before going to Teams: SIP/2.0 183 Session Progress^M ... v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M m=audio 37820 RTP/SAVP 9 8 0 101^M a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M But then when I send the OK after the 183, I am setting RTP/SAVP before sending to MS Teams, but not setting a=crypto: Also note that I can see there are _some_ channels still as RTP/AVP so maybe this is part of the issue. SIP/2.0 200 OK^M ... v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M m=audio 37820 RTP/SAVP 9 8 0 101^M a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M Rhys Hanrahan | Chief Information Officer e: rhys(a)nexusone.com.au<mailto:rhys@nexusone.com.au> [www.nexusone.com.au]<http://www.nexusone.com.au/> [signature_1116663581] <http://www.fusiontech.com.au/> NEXUS ONE | FUSION TECHNOLOGY SOLUTIONS p: 1800 NEXUS1 (1800 639 871) or 1800 565 845 | a: Suite 12.03 Level 12, 227 Elizabeth Street, Sydney NSW 2000 www.nexusone.com.au<http://www.nexusone.com.au/> | www.fusiontech.com.au<http://www.fusiontech.com.au/> The information in this email and any accompanying attachments may contain; a. Confidential information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; b. Legally privileged information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; and or c. Copyright material Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties. If you have received this email in error, please notify the sender immediately and delete this message. Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd does not accept any responsibility for loss or damage arising from the use or distribution of this email. Please consider the environment before printing this email.

4 13

Kamailio + KEMI + topos + change_reply_status
by Marrold 09 Mar '22

09 Mar '22

Hi, I'm using Kamailio 5.5.3 with a KEMI / Python based configuration on Debian 11. In my reply route I am trying to convert a 183 to a 180 with: KSR.textopsx.change_reply_status(180, "Ringing") I've noticed with TOPOS enabled, it doesn't work and I still see a 183 go out. But with it disabled, it works as expected and it's converted to a 180. I get the same behaviour with / without msg_apply_changes() Config looks like this: def ksr_onreply_manage(self, msg): KSR.info(f"ksr_onreply_manage\r\n") scode = KSR.pv.get("$rs") if KSR.permissions.allow_source_address_group()<0: if scode == 183: KSR.info(f"ksr_onreply_manage: Converting 183 to 180\r\n") KSR.textopsx.change_reply_status(180, "Ringing") KSR.textopsx.msg_apply_changes() if scode>100 and scode<299 : self.ksr_route_natmanage(msg) return 1 The debug log doesn't reveal anything too obvious, I can share if its of any use. Does anyone have any suggestions? Thanks Matthew

2 2

kamailio.cfg : behind the NAT (listen and advertise) understanding
by Youssef Boujraf 07 Mar '22

07 Mar '22

Hi Dear I am trying to understand how "listen" and "advertise" work for sip signaling purpose ( corebooks : listen <http://www.kamailio.org/wiki/cookbooks/5.4.x/core#listen> and advertise ) Flow : kamailio behind NAT using softphones on private and public networks. - kamailio : Network : IPv4_private 192.168.1.6 / IPv4_public kamailio.cfg : listen=udp: IPv4_private: 5060 advertise IPv4_public : 5060 - uac (softphone) : zoiper, microsip, cisco spa508g USE CASE : I 'd like to force all softphone on the same local network to reach kamailio through a public ip address. ??? QUESTION : Do I need to REGISTER on IPv4_private address of Kamailio OR REGISTER on IPv4_public address of kamailio or DEFINE another "listen" with another PORT WITHOUT "advertise" for softphone on same kamailio network ? listen=udp: IPv4_private: 5060 advertise IPv4_public : 5060 listen=udp: IPv4_private: 5066 Do I need to use "kamctl ps" to verify which interface is listening on port ? Thanks in advance Best Regards, Youssef BOUJRAF Email: yboujraf(a)gmail.com

3 5

simple GUI for Kamailio as a SBC - could be paid
by Krzysztof Drewicz 05 Mar '22

05 Mar '22

Hello, We searched for a simple GUI that could drive a kamailio - a bunch of trunks, some custom logic, tried to use open source Sipwise C5 - a way too big and heavy. Any hints ? links ? Mabe a paid subscription? Our goal is - to have a GUI that could - put a version number to configuration, and have a simple way to audit changes - only to log - what was done, when and from which management account. vi + svn/git - it's working but this is no GUI, ACLs in sqlite, also work, but - need to have a ssh console, keep files in some repository - well this is simply inconvenient when you have more than 5 trunks with registration etc... BR, -- *Krzysztof Drewicz* Senior Infrastructure Administrator CLUDO | ul. Grochowska 306/308, 03-840 Warszawa t+48221223977 kdrewicz(a)cludo.pl | www.cludo.pl <http://www.cludo.pl/>

5 7

OS-assigned TLS source Port#?
by Matthias Urlichs 02 Mar '22

02 Mar '22

Hi, My Kamailio installation originates TLS connections using source port 5061. Is there a way to teach it not to do that? Seting tcp_reuse_port=no doesn't have any effect. -- -- Matthias Urlichs

2 2

ndb_redis - redis-cluster error
by Maharaja Azhagiah 02 Mar '22

02 Mar '22

Hi All, I am new to Kamailio and testing location server using kamailio and redis-cluster. My setup is simple: All SIP REGISTER messages will be forwarded to this location-server. The server checks for AUTH and then needs to save the usrloc in redis-cluster. I have hiredis and hiredis-cluster libraries installed I am using ndb_redis to support redis-cluster with below configuration: modparam("ndb_redis", "server", "name=172.17.0.3:7000 ;addr=172.17.0.3;port=7000") modparam("ndb_redis", "server", "name=172.17.0.4:7001 ;addr=172.17.0.4;port=7001") modparam("ndb_redis", "server", "name=172.17.0.5:7002 ;addr=172.17.0.5;port=7002") modparam("ndb_redis", "server", "name=172.17.0.6:7003 ;addr=172.17.0.6;port=7003") modparam("ndb_redis", "server", "name=172.17.0.7:7004 ;addr=172.17.0.7;port=7004") modparam("ndb_redis", "server", "name=172.17.0.8:7005 ;addr=172.17.0.8;port=7005") modparam("ndb_redis", "cluster", 1) If SIP method is REGISTER, Kamailio sends redis_cmd to store as below: redis_cmd("172.17.0.3:7000", "sadd %s %s", "location:usrdom::437780082: mazhagiah.sip.zang.io", "location:entry::$xavp(ulrcd=>ruid)", "r"); However, redis-cluster replies with MOVED message, but kamailio / ndb_redis throughs error as below: 0(3915) DEBUG: ndb_redis [redis_client.c:391]: redisc_get_server(): Hash 2590462363 (172.17.0.3:7000) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2591109923 (172.17.0.8:7005) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590232843 (172.17.0.7:7004) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590175467 (172.17.0.6:7003) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590347603 (172.17.0.5:7002) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590290227 (172.17.0.4:7001) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590462363 (172.17.0.3:7000) 0(3915) DEBUG: ndb_redis [redis_client.c:896]: redisc_exec(): rsrv->ctxRedis = 0x558e4d739210 0(3915) DEBUG: ndb_redis [redis_client.c:903]: redisc_exec(): rsrv->ctxRedis = 0x558e4d739210 0(3915) DEBUG: ndb_redis [redis_client.c:957]: redisc_exec(): rpl->rplRedis->type:6 0(3915) ERROR: ndb_redis [redis_client.c:959]: redisc_exec(): Redis error:MOVED 10082 172.17.0.4:7001 0(3915) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-621e8438-f4b-b] (11 / 19) 0(3915) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-621e8438-f4b-b 0(3915) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-621e8438-f4b-c] (12 / 19) Please help if I am missing anything with my configuration or library. Thanks Raja

2 1

Problem with auth_ephemeral and parse_uri(): bad port in uri
by Vlasis Chatzistavrou 01 Mar '22

01 Mar '22

Hello, I have a problem with Kamailio 5.4.6 and auth_ephemeral. I have the following in the Kamailio configuration loadmodule "auth_ephemeral" modparam( "auth_ephemeral", "sha_algorithm", 3 ) modparam( "auth_ephemeral", "username_format", 0 ) modparam( "auth_ephemeral", "secret", 1234 ) as per https://kamailio.org/docs/modules/4.1.x/modules/auth_ephemeral.html#auth_ep… and registrations fail. In the logs we see: Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:777]: check_parse_uris(): looking up From header Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:817]: check_parse_uris(): parsing From URI Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} <core> [core/parser/parse_uri.c:1296]: parse_uri(): bad port in uri (error at char 5 in state 2) parsed: <sip:3518929:16411>(17) /<sip:3518929:1641150726@192.168.2.99> (35) Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: WARNING: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:820]: check_parse_uris(): failed to parse From uri Apparently Kamailio is confused by the timestamp following the username separated by the : character. The REGISTER message is below: REGISTER sip:192.168.2.99 SIP/2.0 Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK5452321 Max-Forwards: 70 To: "3518929" <sip:3518929:1641148397@192.168.2.99> From: "3518929" <sip:3518929:1641148397@192.168.2.99>;tag=ht76o8b2b6 Call-ID: phkj9mi2n3s3ju7uu3qq2f CSeq: 274 REGISTER Contact: <sip:edh7mmti@192.0.2.202;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:ca5e9372-dfa1-459a-b6ba-4398d23bd896>";expires=300 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, outbound User-Agent: Raspberry Phone (SipJS - 0.11.6) Content-Length: 0 and Kamailio parses it as sip:<username>:<port> instead of sip:<username>:<timestamp>. Is this a bug that should be reported or is there any setting that I am missing?

4 9

t_on_failure => what variables being reset to which values?
by Benoît Panizzon 01 Mar '22

01 Mar '22

Hi List We t_on_failure re-route calls to another destination in case one fails for example announcement servers... Before sending to the announcement server I do add some extra header to define which announcement in which language should be played: append_hf("IMP-annc: $avp(announcecode)\r\n"); append_hf("IMP-mandate: $avp(mandate)\r\n"); append_hf("IMP-language: $avp(callerlang)\r\n"); Also $rU is being set prior to sending the call to the announcement server. Now one of our announcement servers became unavailable and t_on_failure was triggered. The 2nd announcement servers was receiving the calls without any of the additional header and with the 'inbound' $ru without the altered $rU https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_on_failu… I'm a bit confused by that, as this states the URI is reset to the value it had on 'relaying'. Is this on 'inbound replaying' or on 'outbound relaying' when route(RELAY) was called to send the call TO the first annoucement server? Is there a bit more specific information on that behaviour? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 3

KEMI jsdt CommonJS-based module
by Ian Carlson 01 Mar '22

01 Mar '22

Greetings, I'm new to Kamailio and working to add Kamailio to an existing Asterisk solution. I'm attracted to the idea of routing via KEMI jsdt since our Asterisk ARI is using nodejs. Playing with KEMI jsdt I was trying to use require() to break up the monolithic routing script. It seems that Kamailio uses Ducktape 2 without module-node or module-ducktape, requiring all routing in a single file? I'd appreciate any suggested practices for managing Kamailio configurations. Cheers, Ian

2 4

RTP Engine early media issue one to many
by Filippo Graziola 28 Feb '22

28 Feb '22

Hi everyone, in my environment (kamailio - http API for retrieve destinations - rtpengine) it is used a lot parallel forking, the flow is as follows: A -> INVITE -> Kamailio -> http API/rtjson / t_relay -> Kamailio -> client B / C / D (only one accepting the call) Kamailio loops to itself to have a parallel forking, it involves rtpengine correctly. Both audio and video work during call. The issue comes with one of destinations sending 183 with SDP, the caller will send then a video preview as early media before call is established. I noticed that only the first one "asking" for early media is receiving this video preview (sometimes none of clients requesting it), all others are ignored by caller. This is a case of 1-to-many, I searched a lot about this but it seems at least not working or not supported, could you confirm that there is no configuration trick possible to make work this early media to many destinations? Another issue in above call flow, is that if client B is receiving correctly this early media video and client C answers the call (200OK) client C does not see any video, audio working and video is completely missing from rtp, only a reinvite makes possible to see video. Is this a client side only issue (so only solution is the reinvite) or there is something missing on my kamailio configuration? Thanks Filippo

1 0

ACK and DNS failover
by Julien Klingenmeyer 25 Feb '22

25 Feb '22

Hello, I use DNS failover feature for routing some calls, and I wonder about its limitations. I noticed that INVITE and ReINVITE requests are correctly routed based on SRV priority/weight in case of failure, but ACK requests do not use them. Let’s say I have Kamailio A relaying requests to a pool of Kamailios B1 and B2 in TLS. DNS records are the below ones. kamailiob.net. 60 NAPTR 20 100 S SIPS+D2T _sips._tcp.kamailiob.net. _sips._tcp.kamailiob.net. 60 SRV 1 10 5061 kamailiob-1.net. _sips._tcp.kamailiob.net. 60 SRV 2 10 5061 kamailiob-2.net. If B1 is down, I expect requests being relayed to B2 (because of priority 2 in the SRV record). * For initial INVITEs: R-URI is sip:kamailiob.net * If B1 is down, the request is retried to B2 => OK * For in-dialog requests: Route header in incoming request is sip:kamailiob.net so it is set as next destination with loose_route function * If B1 is down, the request is retried to B2 for ReINVITE => OK * But once the ACK comes in, with the same Route header (sip:kamailiob.net), Kamailio tries to send it to B1 only (no retries to B2, and even no retransmissions to B1 either). * When B1 is back UP again a few seconds later, Kamailio does not try to relay the ACK again. The ACK request is attempted to be retransmitted only once to B1. It fails and no more retries after that. Is it the expected behavior? Or is it something misconfigured on Kamailio A? Or is it a TLS connection issue? Here is the DNS configuration on Kamailio A (TM module is enabled): dns_try_naptr=yes use_dns_failover=yes use_dns_cache=yes dns_srv_lb=yes enable_tls=yes tcp_max_connections=4096 tls_max_connections=4096 tcp_reuse_port=yes tcp_connect_timeout=10 tcp_send_timeout=10 tcp_connection_lifetime=3600 I had a look into the DNS tutorial (https://github.com/kamailio/kamailio/blob/master/doc/tutorials/dns.txt) without finding any hint about this. If anyone has already played with SIP DNS failover in Kamailio, your help would be appreciated, thanks! Julien

3 4

kamailio and certificate renewal
by Aymeric Moizard 25 Feb '22

25 Feb '22

Hi all, I'm switching my kamailio to use Let's Encrypt certificates. Do I need to force a restart of kamailio when the certificate is renewed or will this be automatic? The doc indicates that reload is possible (but not advised) but it looks to be for a "configuration change" (tls.cfg file change) and not for certificate change. Maybe certificates are automatically reloaded? Regards Aymeric -- Antisip - http://www.antisip.com

2 1

Re-using an existing TLS connection
by Matthias Urlichs 25 Feb '22

25 Feb '22

Hello, My problem: our provider opens a TLS connection to us. They want us to use this channel for outgoing calls, instead of opening a new SIP connection to their server. Is there a way to teach Kamailio to do that? -- -- Matthias Urlichs

3 3

Info: SIP resources collection
by Daniel-Constantin Mierla 24 Feb '22

24 Feb '22

Hello, over the 20 years I used various SIP applications, libraries and tools but after a while I forgot many of them, therefore I tried to collect many in a list hosted on github at: * https://github.com/miconda/sip-resources Hoping it proves useful for some of you! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

4 6

kamailio : how to xlog the value of "myself" ?
by Youssef Boujraf 24 Feb '22

24 Feb '22

Dear I 'd like to xlog the value of the "myself" ? if(uri==myself) { log("the request is for local processing\n"); };Best Regards, -- Youssef BOUJRAF

3 3

dispatcher.reload
by David Villasmil 24 Feb '22

24 Feb '22

Hello guys, We're dispatching with hash over user, if i execute dispatcher.reload <http://www.kamailio.net/docs/modules/5.0.x/modules/dispatcher.html#dispatch…> does the dispatcher list gets calculated? i.e. user A is always dispatched to box3, if i reload, is the hash recalculated and will probably be dispatched differently? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 4

Dispatcher Load Balance with SIP Trunk
by warawich 24 Feb '22

24 Feb '22

Hello, I would like to do a load balance between Asterisk SIP trunks. You can see a diagram from this link: https://drive.google.com/file/d/1Qy66L5rQCfxysYQpSd2-ek_8-by0T8PR/view?usp=… SIP Packets capture log: https://drive.google.com/file/d/1CHGUOwoRDAC93MMBtyfa8gBISKIVC-ng/view?usp=… Details 1. Asterisk1 makes SIP trunk connection with Kamailio. 2. Kamailio makes SIP trunk connection with Asterisk2 and Asterisk3 3. Caller register SIP phone with Asterisk1 Caller extension = 8002 4. Asterisk2 makes a blind transfer to Kamailio. (Call to 8009 then transfer to ARI APP) exten = 8009,1,Transfer(SIP/3802@<kamailio public IP>) 5. ARI app extension is 3802 My problem is that Kamailio LB only works when I try to connect with Asterisk1 and Asterisk2 or Asterisk1 and Asterisk3. If I have two Asterisks in dispatcher.list, it doesn't work and it appears *SIP/2.0 401 Unauthorized* in sip packets capture log. Kamailio version: 5.5.2 Kamailio.cfg modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 3) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } dispatcher.list # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attrs(str,opt) 1 sip:10.148.0.31:5123 1 sip:10.148.0.44:5123 Thanks, Warawich

2 2

SQLOPS sql_query_async() not functioning in an async manner?
by Ben Kaufman 23 Feb '22

23 Feb '22

I'm not sure if this is by design or not. I have things I want to log to a SQL DB, but it's not nearly as important as call handling, so the sql_query_async() function looked ideal, but in testing, if the DB is not responsive, the call processing gets blocked. In the example below, I'm using just a short hostname which isn't resolvable, but if I have an IP address there that isn't really assigned (for example if the DB server is down), the same blocking behavior is exhibited, but the error is slightly different (can't connect instead of unknown host). Here's my config. This specific simple config was tested on Kamailio 5.5, but I observe similar behavior (in a more complicated config) in the dev branch as well. - - - - #!KAMAILIO async_workers=2 loadmodule "db_mysql" loadmodule "sqlops" loadmodule "sl" loadmodule "xlog" loadmodule "pv" force_rport=yes ## Host sql does not resolve. modparam("sqlops","sqlcon","cb=>mysql://kamailio:kamailiorw@sql/kamailio") ## But start anyways. modparam("sqlops", "connect_mode", 1) request_route { sl_send_reply("100", "Trying"); ## Prevent re-transmissions from the UAC xlog("L_NOTICE", "Time before sql_query_async(): [$TV(Sn)]\n"); sql_query_async("cb", "INSERT INTO table1 (col) VALUES ('val')"); xlog("L_NOTICE", "Time before sl_send_reply(): [$TV(Sn)]\n"); sl_send_reply("404", "Not found"); } Here's the log if I send a simple INVITE with SIPp. Note the timestamps 9(16) NOTICE: <script>: Time before sql_query_async(): [1645574694.546165] 9(16) ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'sql' (-3) 9(16) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 9(16) ERROR: sqlops [sql_api.c:190]: sql_reconnect(): failed to connect to the database [cb] 9(16) ERROR: sqlops [sqlops.c:232]: sql_check_connection(): failed to connect to database 9(16) ERROR: sqlops [sqlops.c:264]: sql_query_async(): invalid connection to database 9(16) NOTICE: <script>: Time before sl_send_reply(): [1645574699.552440] Ben Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

4 4

kamailio :?==?utf-8?q? ROUTE[AUTH] - IPAUTH - use case
by Youssef Boujraf 23 Feb '22

23 Feb '22

Dear, I enabled two softphone with ADDRESS. - codec1 : ip 192.168.30.241/32 5060 - codec2 : ip 192.168.30.242/32 5060 I enable one softphone with SUBSCRIBER - codec3 : is registered (login/passwd) codec4 is not part of the domain, is not subriber and address. Kamailio proxy - kamailio : ip 192.168.30.240/32 5060 In the kamailio.cfg ROUTE[AUTH]# IP authorization and user authenticationroute[AUTH] {#!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return;} Regarding the condition #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif USE CASE : If codec4 (not registered or same domain) tries to INVITE codec3 (in subscriber) the codec, an INVITE MESSAGE is sent and the codec3 could ACK. - But I don't want to allow the INVITE, I 'd like to not allow the codec4 to reach codec3. - But if codec1 or codec2 tries to reach codec3, kamailio will allow the INVITE. Best Regards, -- Youssef BOUJRAF

1 0

Issue with Authenticating outbound Call
by Victor ku 22 Feb '22

22 Feb '22

Hi, I'm new to sip and kamailio, I have recently setup a docker container based on kamailio/kamailio:5.5.0-stretch to check things out. While I was trying to authenticate an outbound call to a 3rd party sip provider from my kamailio container, I noticed there are times that the 2nd invite with the authorization header that's supposed to be sent from the failure route doesn't get sent out. There's no error recorded in the log nor does wireshark picking up any reply to sip phone or message to 3rd party sip provider. Anyone know what might have caused this? Some other things to note: Kamailio does sent out the INVITE with authorization header and authenticated successfully when my sip phone sent the sip request with FROM ip set to the 3rd party sip server's ip. When I change the sip phone to send INVITE with FROM Domain using kamailio container's ip, I'm able to see header is being built from the trace but no INVITE with authorization header is being sent. Changing the FROM uri using $fu or uac_replace_from() does help either. ---------------------------------------------------------------------------------------------------------------- > > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} uac [auth_hdr.c:408]: > build_authorization_hdr(): hdr is <Authorization: Digest > username="xxxxxxxx", realm="xxx.xxx.xxx.xxx", nonce="07f4a5779568", > uri="sip:+0123456789@xxx.xxx.xxx.xxx", qop=auth, nc=00000001, > cnonce="3782993246", response="d5ebd0ef00d44db2292c7b8508fa23bc", > algorithm=MD5 > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1026 a=16 > n=if > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1021 a=24 > n=t_relay > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} <core> [core/msg_translator.c:1797]: > check_boundaries(): no multi-part body > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1030 a=2 > n=exit > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483650 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1564]: > t_should_relay_response(): rps discarded - uas status: 100 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1924]: relay_reply(): > reply status=2 branch=0, save=0, relay=-1 icode=0 msg status=401 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:596]: > receive_msg(): reply-route executed in: 1753 usec > kamailio-kamailio-1 | 14(22) DEBUG: dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483652 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:606]: > receive_msg(): cleaning up ---------------------------------------------- package capture ---------------------------- 192.168.0.165 xxx.xxx.xxx.xxx SIP/SDP 1383 Request: INVITE > sip:+0123456789@ xxx.xxx.xxx.xxx | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 585 Status: 100 Trying | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 820 Status: 401 Unauthorized | > 192.168.0.165 xxx.xxx.xxx.xxx SIP 396 Request: ACK sip:+0123456789 > @ xxx.xxx.xxx.xxx | ------------------------------------------nothing more till sip phone cancel ----------------- # ----------------- below is failure route config --------------------- > failure_route[MANAGE_FAILURE] { > xlog("~~~~~ MANAGE_FAILURE ($rm) ($rs-$rr) (from $fu) (to $tu) (req :$ru) > (dest $du)\r\n"); if (t_is_canceled()) { exit; } xlog("Checking status code\r\n"); #!ifdef WITH_UAC if(t_check_status("401|407")) { if (isflagset(7)) { t_reply("503","Authentication failed"); exit; } $avp(auser) = "xxxxxxxxxx"; $avp(apass) = "xxxxxxxxxx; if(uac_auth()) { setflag(7); xlog("after uac_auth $rm (from $fu)(to $tu)(req $ru)(dest $du)\r\n"); if(!t_relay()) { xlog("Failed to relay ($rm) ($rs-$rr) (from $fu)(to $tu)(req $ru)\r\n"); t_reply("500","Error"); } } else { xlog("uac_auth() not available\r\n"); t_reply("503","Authentication invalid"); } exit; } #!endif } > # ----------- end failure route config --------------------

2 2

reinvite with 2 video lines in SDP
by Filippo Graziola 22 Feb '22

22 Feb '22

Hi all, I'm facing a strange behaviour from a client (iOS application based on pjsip library) which is doing as follow when receiving a call (on phone locked): - accept the call rejecting video (m=video 0...) in SDP - reinvite to negotiate again video media Specifically the reinvite passing through kamailio (+rtpengine) has this SDP body after rtpengine offer (the client use TLS so I can't see the original message): v=0 o=- 3854443139 3854443142 IN IP4 PUBLIC_RTP_ENGINE_IP s=pjmedia b=AS:386 t=0 0 a=X-nat:0 m=audio 46734 RTP/AVP 8 101 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:96000 a=ssrc:1456418416 cname:441b90ac6db72f22 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:46735 m=video 0 RTP/AVP 96 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 m=video 46776 RTP/AVP 98 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=42e01e; packetization-mode=1 a=sendrecv a=rtcp:46777 from this I can understand that the client is doing something strange, because there is still the rejection line (I know that this question should go to the pjsip guys) but I would also like to know if it is possible to "clean" this sdp by kamailio. A few clients receiving this sdp are answering with only audio. Reading the docs, sdpops/textops modules allow modifying lines by prefix or media, but till now I have no luck to exclude only the line "m=video 0 RTP/AVP 96" and his attributes without touching the correct video request. Could someone please give me some hints on it?

2 2

Enable switch redirect to other SIP server source
by bayu p 21 Feb '22

21 Feb '22

Dear Kamailio Expert, Could you help me switching redirect sip server to another sip server ? As described below architecure. I have some uac_redirect module to implement , but kamailio cannot perform running states. Pleas help to reditect sip server to another sip server. Thanks & Best regards,

2 1

kamailio success story
by Valentin Christoph 21 Feb '22

21 Feb '22

Hi all, We had to connect to an SBC, which was not compliant to RFC 3261. It was urgent, because management was waiting for first call. Within minutes, I could implement a workaround in kamailio routing configuration: ## CV experimental 2022-02-17: fake Supported HF if (is_present_hf("Require") && !is_present_hf("Supported")) { append_hf("Supported: $hdr(Require)\r\n", "Require"); msg_apply_changes(); } Thank you, kamailio! All the best Christoph

2 1

SIP server port service cannot be changed from default value (5060)
by bayu p 18 Feb '22

18 Feb '22

Hello Experts, Please help me to change port service from default 5060 port into other port due to security reason. Since I have change port in kamailio.cfg file, but services cannot fly normally that shown client unable login online (status not online). [image: image] <https://user-images.githubusercontent.com/76675592/154027407-24ab423b-302b-…> Besides, I need to run the kamailio services after install default kamailio, I have not found where procedure found in this installation process on github.com/kamailio. Could you please share if there another steps of compiling Kamailio source, install, deploy process, and running command for this specific installation process? Thanks Gb

4 9

Kamailio v5.5.4 Released
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, Kamailio SIP Server v5.5.4 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.4. For more details about version 5.5.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/02/kamailio-v5-5-4-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

2 2

Releasing Kamailio v5.5.4
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, I am considering to release Kamailio v5.5.4 later this week, likely on Thursday, Feb 17, 2022. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

Info: sipexer v1.0.0 - sip cli tool
by Daniel-Constantin Mierla 16 Feb '22

16 Feb '22

Hello, I want to announce the availability of sipexer v1.0.0 - a sip cli tool that can facilitate testing and monitoring of SIP signalling systems. It tries to have a modern approach, with a flexible templating system, supporting both IPv4 and IPv6 with all the common transport layers, respectively UDP, TCP, TLS and WebSocket (for WebRTC). The project can be found at: * https://github.com/miconda/sipexer It is written in Go language for better portability, binaries for Linux, MacOS and Windows are made available for download in the release page: * https://github.com/miconda/sipexer/releases/tag/v1.0.0 Among its features: * send OPTIONS request (quick SIP ping to check if server is alive) * do registration and un-registration with customized expires value and contact URI * authentication with plain or HA1 passwords * set custom SIP headers * template system for building SIP requests * fields in the templates can be set via command line parameters or a JSON file * variables for setting field values (e.g., random number, data, time, environment variables, uuid, random string, …) * simulate SIP calls at signalling layer (INVITE-wait-BYE) * respond to requests coming during SIP calls (e.g., OPTIONS keepalives) * send instant messages with SIP MESSAGE requests * color output mode for easier troubleshooting * support for many transport layers: IPv4 and IPv6, UDP, TCP, TLS and WebSocket (for WebRTC) * send SIP requests of any type (e.g., INFO, SUBSCRIBE, NOTIFY, …) One usage example that could ease the testing of Kamailio is initiating registrations or simulating calls over WebSocket without the need of having a JavaScript soft phone application running in a web browser. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

10 12

allow_source_address_group in failure route
by Marrold 16 Feb '22

16 Feb '22

Hello, I'm using Kamailio as a proxy between several voice gateways and if the call fails for whatever reason (far end is busy etc) I am forwarding to an Asterisk box to play an announcement. Within the failure route I am using allow_source_address_group() to ascertain if the failure response was from a Voice Gateway or Asterisk itself, however it appears to return the group for the first branch when the reply is from the second branch to Asterisk. To handle the failure I am adding some headers, and setting a new destination indirectly with ds_select_dst before t_relay. I am using 5.5.3 on Debian 11, using KEMI / Python. Does anyone have any ideas? Thanks Matthew

2 1

Missing contact in 180 ringing causes wrong routing
by Xuo Guoto 15 Feb '22

15 Feb '22

Hello all, I have a kamailio ( kamailio 5.5.3 (x86_64/linux)) with asterisk (Asterisk 16.2.1~dfsg-2ubuntu1) with following architecture ┌─────────────┐ ┌─────────────┐ ┌─────────────────────┐ │ Asterisk ├───────┤ Kamailio │ 188.xxx.59.171 ───────── 103.xxx.xx.134 │ Linphone │ └─────────────┘ └─────────────┘ └─────────────────────┘ 10.10.1.13 10.10.1.9 In case the diagram is not displayed correctly its pasted at https://pastebin.com/b0UtE4Lj Registrations are handled by Kamailio and calls from linphone are routed to Asterisk. Similarly all calls originating in asterisk are routed over to Kamailio which in turn looks up AOR and routes the calls over to Linphone. Along with Linphone I also have a WebRTC endpoint at Kamailio. When calls are sent to WebRTC from asterisk, it gets correctly routed over to Asterisk, but if the client is Linphone the ACK of 200OK is not routed correctly to Linphone. I get this error in the logs: [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced tls:188.xxx.59.171:5061, to udp:188.xxx.59.171:5060) Digging into the difference between WebRTC and Linphone, I could see that in 1800 Ringing packet from WebRTC has a Contact headers while its missing in the packet from Linphone This is the 180 ringing from WebRTC: tag: rcv pid: 695615 process: 17 time: 1644860126.209155 date: Mon Feb 14 23:05:26 2022 proto: wss ipv4 srcip: 103.xxx.xx.134 srcport: 52508 dstip: 188.xxx.59.171 dstport: 443 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/WSS 188.xxx.59.171:443;branch=z9hG4bK730a.2cc7461b72f0c895e7cce5a0cedce4f7.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Content-Length: 0 While this is from Linphone tag: rcv pid: 695614 process: 16 time: 1644860097.970935 date: Mon Feb 14 23:04:57 2022 proto: tls ipv4 srcip: 103.xxx.xx.134 srcport: 25843 dstip: 188.xxx.59.171 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/TLS 188.xxx.59.171:5061;branch=z9hG4bKb5b1.d3231367383d07b1063098dcb35cc2b0.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPjd16f1379-1be7-448e-894a-c42d96f63146 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=cf7a7770-529d-4450-8bed-880382dce1dd To: <sip:user_jbjp@10.10.1.9>;tag=2ooYDia Call-ID: 8a025999-b096-4fe0-a623-9f6e6a69b8ae CSeq: 27838 INVITE User-Agent: Linphone Desktop/4.3.2 (Ubuntu 20.04.3 LTS, Qt 5.12.12) LinphoneCore/5.0.49 Supported: replaces, outbound, gruu Record-route: <sip:188.xxx.59.171:5061;transport=tls;r2=on;lr=on;did=3b9.b1b2> Record-route: <sip:10.10.1.9;r2=on;lr=on;did=3b9.b1b2> Content-Length: 0 When the 180 ringing is passed on to the Astrerisk it has the contact header modified with alias. tag: snd pid: 695615 process: 17 time: 1644860126.210189 date: Mon Feb 14 23:05:26 2022 proto: udp ipv4 srcip: 10.10.1.9 srcport: 5060 dstip: 10.10.1.13 dstport: 7000 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;alias=103.xxx.xx.134~52508~6;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Content-Length: 0 Now, I am not sure if this is the reason for routing the packets to wrong destination, but this is all I could figure out. Any clue to get this resolved will be much appreciated. My Kamailio config is pasted at https://pastebin.com/qyRS0usw with regards, X. Sent with ProtonMail Secure Email.

1 0

get kamailio sip processes id
by Patrick Karton 14 Feb '22

14 Feb '22

hello, as we are all aware kamailio create a lot of processes for many tasks. im using kamailio in kubernetes and i would like to monitor only sip processes with top command. i only need a way to get only kamailio sip processes id (udp receiver that handle sip request) . but i dont knwo how to get it. top -c -p id1,id2,id3,id4 Thanks

3 5

SEMS license with kamailio and rtpengine
by Patrick Karton 14 Feb '22

14 Feb '22

Hello im using kamailio and rtpengine to build a custom SBC. i was using freeswitch as my b2bua but i noticed a lot of issues when cps and simultaneous increased so i tried sems and i have better results. But i do not really understand sems license so i want to know if i can use it. i read from sems github "It is licensed under dual license terms, the GPL (v2+) and proprietary license. This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed. For a license to use SEMS under non-GPL terms, please contact FRAFOS GmbH at info(a)frafos.com" so my question is if i want to use sems as b2bua with kamailio and rtpengine and make profit of it do i need to purchase a license each time i want to sell it to a customer ? Thanks.

7 39

Kamailio server behind NAT
by Tihomir Culjaga 13 Feb '22

13 Feb '22

Hi all I'm wondering what is best practice to configure kamailio for a setup shown below. Carriers only support interconnection over a private network so I need to terminate every separate network on kamailio machine (and on RTPEngine) . That will turn very quickly into an administration nightmare as we add more and more carriers. My plan is to terminate carrier private networks on a router and NAT the traffic to kamailio where it leads me to a kamailio behind NAT setup. Well, in theory set_advertised_address and set_advertised_port per message can do the trick... at least I hope. Also, some carriers require a SIP registration to activate SIP trunks. I do that using UAC module (remote registration) but here again I got a problem with contact address as you can set only one contact address (modparam("uac", "reg_contact_addr", "10.1.1.11:5060")). /--RTP 10000-19999-- RTPEngine1 -----RTP------+-- IPPBX1 carrier1 --- NAT ---| | | | | | carrier2 --- NAT ---+-----SIP 5060------ kamailio ----SIP---- Dispatcher | | | carrier3 --- NAT ---| | | \--RTP 20000-29999-- RTPEngine2 -----RTP------+-- IPPBX2 So bottom line, I'm wondering what would be the best practice for such a scenario ? That would spare me from hitting many pitfalls. Would really appreciate a hint. Regards, Tihomir.

3 3

Vo5G with open5gs
by Hornbuckle, Mark A 11 Feb '22

11 Feb '22

Hello, I have been trying to find documentation on Kamailio IMS and Vo5G support for open5gs. I wanted to ask you directly if Kamailio IMS does support Vo5G with open5gs? Thank you

1 0

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