- sr-users - kamailio.org

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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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!

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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

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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

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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

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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

1 0

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

2 3

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 } }

1 0

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

3 4

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.

3 2

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

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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

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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

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"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

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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

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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

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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

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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.

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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.

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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,

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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

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