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37519 discussions

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

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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!

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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.

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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/

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