October 2025 - sr-users

Postgres connections optimization #4342
by Stephane Savonitto 17 Jun '26

17 Jun '26

Hello, On P-CSCF when running we have 200+ permanent connection to postgres db, even if there is no traffic. Is there a way to manage connections dynamically using a pool like using PgBouncer ? Regards, Stéphane

6 10

Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
by Ben Fitzgerald 17 Apr '26

17 Apr '26

Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: <sip:40081@kamailio:5060> Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER ========================= sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

4 8

problems using kamailio
by Peter Walber 17 Apr '26

17 Apr '26

Hello, I try to use kamailio as webrtc-sip-gateway between my openhab smart home system and my doorbird doorbell with phone and camera. I try to use the product of "https://github.com/florian-h05/webrtc-sip-gw". Which integrates kamailio together with rtpengine to the gateway. I want to use openhab as SIP Client and have a Telekom Speedport Smart 4 Router on the other side. The webrtc-sip-gw from florian-h05 shall be the software between Openhab and the speedport smart 4. On the speedport smart router the SIP Server is enabled and configured. I can sucessfully call with zoiper extern phone numbers. The doorbird als is configured correctly. I can call the doorbird with my zioper softphone using the SIP interface of the doorbird. The runtime system is a docker container on debian 12. I want to call the doorbird by SIP from my openhab SIP client. I can initiate the call, but it doesn't establish Here is the log from the docker container. --------------------------------------------------------------------------------------------------- 2025-08-07T07:46:57.384687165Z 9(52) INFO: <script>: HTTP Request Received 2025-08-07T07:46:57.391479050Z 9(52) INFO: <script>: START: REGISTER from sip:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:46:57.391516534Z 9(52) INFO: <script>: Current Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:243e461d-f0cb-4c09-9a63-f57c9bba8d6b>";expires=600 2025-08-07T07:46:57.391526535Z 9(52) INFO: <script>: Setting new Contact header: <sip:**73@192.168.2.209:5060> 2025-08-07T07:47:00.195739350Z 9(52) INFO: <script>: START: INVITE from sip:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:47:00.195800424Z 9(52) INFO: <script>: Current Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;transport=ws;ob> 2025-08-07T07:47:00.195811886Z 9(52) INFO: <script>: Setting new Contact header: <sip:v0fds0hl@fe8ojuo7pnkl.invalid;transport=ws;ob;alias=192.168.2.185~39510~6;alias=192.168.2.185~39510~6> 2025-08-07T07:47:00.196163528Z 9(52) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:**72@speedport.ip 2025-08-07T07:47:00.196181228Z 9(52) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:**72@speedport.ip, method:INVITE, status:<null>, extra_id: z9hG4bK44082210, rtpengine_manage: replace-origin replace-session-connection via-branch=extra rtcp-mux-demux SDES-off ICE=remove RTP/AVP 2025-08-07T07:47:00.196608235Z INFO: [penvq8f06ehafutk9ivf]: [control] Received command 'offer' from 127.0.0.1:50790 2025-08-07T07:47:00.196625247Z NOTICE: [penvq8f06ehafutk9ivf]: [core] Creating new call 2025-08-07T07:47:00.198519843Z INFO: [penvq8f06ehafutk9ivf]: [control] Replying to 'offer' from 127.0.0.1:50790 (elapsed time 0.001979 sec) 2025-08-07T07:47:01.927528447Z 9(52) INFO: <script>: START: CANCEL from sip:**73@speedport.ip (IP:192.168.2.185:39510) 2025-08-07T07:47:01.927577111Z 9(52) INFO: <script>: Current Contact header: <null> 2025-08-07T07:47:01.927586439Z 9(52) INFO: <script>: Setting new Contact header: <null> 2025-08-07T07:47:09.169489716Z 7(50) INFO: <script>: WebSocket connection from 192.168.2.185:39510 has closed *2025-08-07T07:47:27.355185671Z 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found* 2025-08-07T07:47:27.914769340Z 9(52) INFO: <script>: HTTP Request Received 2025-08-07T07:47:27.951608347Z 9(52) INFO: <script>: START: REGISTER from sip:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:27.951663073Z 9(52) INFO: <script>: Current Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:39eac0ca-0aea-4526-a5c7-0516bc8a2d71>";expires=600 2025-08-07T07:47:27.951915520Z 9(52) INFO: <script>: Setting new Contact header: <sip:**73@192.168.2.209:5060> 2025-08-07T07:47:30.168546446Z 2(45) INFO: <script>: BRANCH FAILED: z9hG4bK4408221 + 0INFO: [penvq8f06ehafutk9ivf]: [control] Received command 'delete' from 127.0.0.1:36296 2025-08-07T07:47:30.168587880Z INFO: [penvq8f06ehafutk9ivf]: [core] Deleting call branch '' (via-branch 'z9hG4bK44082210') 2025-08-07T07:47:30.168597011Z INFO: [penvq8f06ehafutk9ivf]: [core] Call branch '' (via-branch 'z9hG4bK44082210') deleted, no more branches remaining 2025-08-07T07:47:30.168605215Z INFO: [penvq8f06ehafutk9ivf]: [core] Deleting entire call 2025-08-07T07:47:30.168612883Z INFO: [penvq8f06ehafutk9ivf]: [core] Final packet stats: 2025-08-07T07:47:30.168620604Z INFO: [penvq8f06ehafutk9ivf]: [core] --- Tag 'utenlull77', created 0:30 ago for branch '' 2025-08-07T07:47:30.168628359Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscribed to '' 2025-08-07T07:47:30.168636445Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscription for '' 2025-08-07T07:47:30.168645446Z INFO: [penvq8f06ehafutk9ivf]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec 2025-08-07T07:47:30.168653691Z INFO: [penvq8f06ehafutk9ivf]: [core] --------- Port 192.168.2.209:23484 <> 192.168.2.185:60650, SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:47:30.168661939Z INFO: [penvq8f06ehafutk9ivf]: [core] --- Tag '', created 0:30 ago for branch 'z9hG4bK44082210' 2025-08-07T07:47:30.168669844Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscribed to 'utenlull77' 2025-08-07T07:47:30.168677713Z INFO: [penvq8f06ehafutk9ivf]: [core] --- subscription for 'utenlull77' 2025-08-07T07:47:30.168685661Z INFO: [penvq8f06ehafutk9ivf]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec 2025-08-07T07:47:30.168693705Z INFO: [penvq8f06ehafutk9ivf]: [core] --------- Port 192.168.2.209:23472 <> :0 , SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:47:30.168718608Z INFO: [penvq8f06ehafutk9ivf]: [core] --------- Port 192.168.2.209:23473 <> :0 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:47:30.168728065Z INFO: [penvq8f06ehafutk9ivf]: [control] Replying to 'delete' from 127.0.0.1:36296 (elapsed time 0.000505 sec) *2025-08-07T07:47:30.169042099Z 2(45) INFO: <script>: Failure: <null> 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found* 2025-08-07T07:47:30.265075785Z 9(52) INFO: <script>: START: INVITE from sip:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:30.265125134Z 9(52) INFO: <script>: Current Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;transport=ws;ob> 2025-08-07T07:47:30.265135557Z 9(52) INFO: <script>: Setting new Contact header: <sip:lebth4eq@fe8ojuo7pnkl.invalid;transport=ws;ob;alias=192.168.2.185~44816~6;alias=192.168.2.185~44816~6> 2025-08-07T07:47:30.265484502Z 9(52) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:**72@speedport.ip 2025-08-07T07:47:30.265664116Z 9(52) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:**72@speedport.ip, method:INVITE, status:<null>, extra_id: z9hG4bK16860370, rtpengine_manage: replace-origin replace-session-connection via-branch=extra rtcp-mux-demux SDES-off ICE=remove RTP/AVP 2025-08-07T07:47:30.265819591Z INFO: [412n0o8dedgair81t70p]: [control] Received command 'offer' from 127.0.0.1:50790 2025-08-07T07:47:30.265979425Z NOTICE: [412n0o8dedgair81t70p]: [core] Creating new call 2025-08-07T07:47:30.267497852Z INFO: [412n0o8dedgair81t70p]: [control] Replying to 'offer' from 127.0.0.1:50790 (elapsed time 0.001599 sec) 2025-08-07T07:47:30.469565385Z 9(52) INFO: <script>: START: CANCEL from sip:**73@speedport.ip (IP:192.168.2.185:44816) 2025-08-07T07:47:30.469629711Z 9(52) INFO: <script>: Current Contact header: <null> 2025-08-07T07:47:30.469645116Z 9(52) INFO: <script>: Setting new Contact header: <null> 2025-08-07T07:47:41.207672567Z 7(50) INFO: <script>: WebSocket connection from 192.168.2.185:44816 has closed 2025-08-07T07:47:57.917754216Z 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found 2025-08-07T07:48:00.230412646Z 2(45) INFO: <script>: BRANCH FAILED: z9hG4bK1686037 + 0INFO: [412n0o8dedgair81t70p]: [control] Received command 'delete' from 127.0.0.1:36296 2025-08-07T07:48:00.230457143Z INFO: [412n0o8dedgair81t70p]: [core] Deleting call branch '' (via-branch 'z9hG4bK16860370') 2025-08-07T07:48:00.230466718Z INFO: [412n0o8dedgair81t70p]: [core] Call branch '' (via-branch 'z9hG4bK16860370') deleted, no more branches remaining 2025-08-07T07:48:00.230475640Z INFO: [412n0o8dedgair81t70p]: [core] Deleting entire call 2025-08-07T07:48:00.230483968Z INFO: [412n0o8dedgair81t70p]: [core] Final packet stats: 2025-08-07T07:48:00.230491909Z INFO: [412n0o8dedgair81t70p]: [core] --- Tag 'hc7h5imu96', created 0:30 ago for branch '' 2025-08-07T07:48:00.230518917Z INFO: [412n0o8dedgair81t70p]: [core] --- subscribed to '' 2025-08-07T07:48:00.230528302Z INFO: [412n0o8dedgair81t70p]: [core] --- subscription for '' 2025-08-07T07:48:00.230537387Z INFO: [412n0o8dedgair81t70p]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec 2025-08-07T07:48:00.230545867Z INFO: [412n0o8dedgair81t70p]: [core] --------- Port 192.168.2.209:23400 <> 192.168.2.185:46431, SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:48:00.230554949Z INFO: [412n0o8dedgair81t70p]: [core] --- Tag '', created 0:30 ago for branch 'z9hG4bK16860370' 2025-08-07T07:48:00.230563124Z INFO: [412n0o8dedgair81t70p]: [core] --- subscribed to 'hc7h5imu96' 2025-08-07T07:48:00.230570851Z INFO: [412n0o8dedgair81t70p]: [core] --- subscription for 'hc7h5imu96' 2025-08-07T07:48:00.230578827Z INFO: [412n0o8dedgair81t70p]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec 2025-08-07T07:48:00.230587091Z INFO: [412n0o8dedgair81t70p]: [core] --------- Port 192.168.2.209:23494 <> :0 , SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:48:00.230595631Z INFO: [412n0o8dedgair81t70p]: [core] --------- Port 192.168.2.209:23495 <> :0 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 30 ts 2025-08-07T07:48:00.230603868Z INFO: [412n0o8dedgair81t70p]: [control] Replying to 'delete' from 127.0.0.1:36296 (elapsed time 0.000253 sec) *2025-08-07T07:48:00.230753116Z 2(45) INFO: <script>: Failure: <null> 2(45) WARNING: tm [../../core/forward.h:203]: msg_send_buffer(): TCP/TLS connection for WebSocket could not be found* --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- especially the bold lines confuse me. Why is the TCP/TLS closed and can not be found ? Could that be the problem ?* * * * Thanks for some help and best regards Peter* *

7 15

topoh breaks SIPS with PJSIP if PJMEDIA_SRTP_MANDATORY is set
by Mathias Schneuwly 01 Dec '25

01 Dec '25

Hi I have a question related to topoh. We are running Kamailio in version 6.0.2 with topoh enabled. The PJSIP based softphone is unable to establish a call with SIPS and SRTP if we use "PJMEDIA_SRTP_MANDATORY" in PJSIP. PJSIP complains about "Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session". The other side is sending "Contact: <sips:620150@10.165.44.117:59686;transport=tls>", but topoh modifies the contact header to "Contact: <sip:127.0.0.8;line=sr-avdFaQDvhUxGrqtxhqxThq-ZnUy6nUzGrQDZmqa1rUf6a0JTaLt4aoyV2EGQ>" without sips scheme and without transport=tls. PJSIP then refuses to establish the call and sends immediately a BYE. If topoh is disabled, the call is successfully established. Is it somehow possible to tell topoh to keep the scheme or transport=tls if the other side provided it? Thanks for your help Best regards Mathias

3 3

Kamailio Developers Meeting, Dec 8-9, 2025, in Dusseldorf
by Daniel-Constantin Mierla 24 Nov '25

24 Nov '25

Hello, Kamailio SIP Server project is organizing another meeting of its developers and community members during December 8-9, 2025 (Mon-Tue), hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and contributors in order to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join, developer or user interested in helping the project. Please note we have a limited capacity of seats in the meeting room, the main policy for accepting participants being first come first server. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

RTPEngine installation on RHEL 8.x or Rocky Linux 8.x
by Chandramouli P 21 Nov '25

21 Nov '25

Hello, I tried a lot to install RTPEngine on RHEL 8.x and as well as in Rocky Linux 8.x also using different sources. But, I am facing a lot of issues like libraries issues, dependencies issues, conflict issues etc. Anyone successfully installed RTPEngine on RHEL 8.x or Rocky Linux 8.x (Not on CentOS)? If so, could you please share the documentation step-by-step or web URL? Thanks in advance. Best regards, Chandramouli.

5 18

Unable to start Kamailio with TLS configuration
by Chandramouli P 21 Nov '25

21 Nov '25

Hello, Please find my server environment below: Operating System: RockyLinux 8.x and RHEL 8.x Kamailio version: 6.0.2 IP address: 10.122.0.4 I have generated SSL certificates using OpenSSL. After configuring Kamailio, I am unable to start Kamailio. Please find the steps that I used to generate certificates along with configuration at the below link: *https://pastebin.com/Veu8z9Pr <https://pastebin.com/Veu8z9Pr>* Any help would be appreciated and thanks in advance. Best Regards, Chandramouli.

3 11

seting sdp bundle with rtpengine
by Samuel Moya Tinoco 02 Nov '25

02 Nov '25

Hi everyone, I want to configure my kamailio+rtpengine to use the option bundle in SDP. I have a dedicated route to configure the specific rtpengine_manage options for each case (it's based on a route I saw in github) route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); if(sdp_with_transport("RTP/SAVPF")) { rtpengine_manage("replace-origin replace-session-connection DTLS=passive SDES-off ICE=force SRTP AVPF rtcp-mux-reject bundle=accept"); t_on_reply("REPLY_WS_TO_WS"); return; } rtpengine_manage("replace-origin replace-session-connection DTLS=passive SDES-off ICE=force UDP/TLS/RTP/SAVPF bundle=accept bundle=accept"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); rtpengine_manage("replace-origin replace-session-connection DTLS SDES-off ICE=force SRTP AVPF bundle=accept"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); if (isflagset(RTPENGINE_NO_SRTP)){ xlog("L_INFO", "Llamada de Telealarm a la IPS (no srtp)"); rtpengine_manage("replace-origin replace-session-connection DTLS=off SDES-off RTP/AVP"); t_on_reply("REPLY_NO_SRTP"); return; } if ($tU=~"^ips") { rtpengine_manage("replace-origin replace-session-connection DTLS=passive SDES-off UDP/TLS/RTP/SAVPF"); t_on_reply("REPLY_NO_SRTP_ICE"); return; } if (isflagset(RTPENGINE_NO_ICE)) { rtpengine_manage("replace-origin replace-session-connection DTLS=passive SDES-off UDP/TLS/RTP/SAVPF ICE=remove"); t_on_reply("MANAGE_CLASSIC_REPLY"); return; } rtpengine_manage("replace-origin replace-session-connection DTLS=passive SDES-off UDP/TLS/RTP/SAVPF ICE=force"); t_on_reply("MANAGE_CLASSIC_REPLY_NO_ICE"); return; } } Reading the rtpengine and kamailio docs I see that rtpengine shows more options than kamailio rtpengine module. In rtpengine options it shows bundle option but in kamailio it doesn't appear. And I've tried to set this option but it doesn't work. Kamailio recives the invite with bundle but when it sends the invite to my asterisk the invite doesn't have the bundle option enabled Does anyone know what could be happening? Maybe I'm doing something wrong Thanks in advance for your time and help Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> ViveLibre C/ La Orotava 4 28660 Boadilla del Monte Madrid www.vivelibre.es<http://www.vivelibre.es/> [cid:image001.png@01DC49C0.02D76010] Soluciones inteligentes para la autonomía personal "Tanto este mensaje como todos los posibles documentos adjuntos al mismo son confidenciales y están dirigidos exclusivamente a los destinatarios de los mismos. Por favor, si Usted no es uno de dichos destinatarios, notifíquenos este hecho y elimine el mensaje de su sistema. Queda prohibida la copia, difusión o revelación de su contenido a terceros sin el previo consentimiento por escrito de VIVELIBRE AUTONOMÍA PERSONAL S.L.U. (VIVELIBRE). En caso contrario, vulnerará la legislación vigente. De conformidad con lo establecido en el Reglamento (UE) 2016/679, General de Protección de Datos, le informamos de que sus datos son objeto de tratamiento por VIVELIBRE, en calidad de Responsable del Tratamiento. VIVELIBRE tratará sus datos con la finalidad de mantener la relación contractual, gestionar su solicitud, así como remitirle comunicaciones de carácter comercial relacionadas con su ámbito de actividad y los servicios prestados. Si desea ejercitar sus derechos de acceso, rectificación, supresión, limitación, oposición o portabilidad, puede dirigirse a la dirección postal Calle de la Orotava 4, 28660, Boadilla del Monte, (Madrid) o a la dirección de correo electrónico protecciondedatos(a)vivelibre.es. Para obtener más información sobre cómo tratamos sus datos, consulta nuestra Política de Privacidad en Política de Privacidad - Vivelibre."

2 2

How to get count of contacts/branches after t_next_contacts()
by Benoît Panizzon 31 Oct '25

31 Oct '25

Hi Gang I use: lookup_to_dset(), t_load_contacts(), t_next_contacts() to gather contacts for parallel forking (like for a hunt group). After that I use $cnt($xavp(tm_contacts)) to find the number of branches which would be created. But $cnt($xavp(tm_contacts)) seems to always be 1 no matter how many branches I get in route(RELAY). What am I doing wrong? Is there another PV I could use? Background: When a 100rel is supported by the originating device, the destination devices might all reply 180 RINGING with 100required, same rseq but different ToTags. The originating device would require to reply PRACK to each of those ringing and it looks like MANY originating devices (especially commercial SBC) do NOT consider the ToTag but ignore any subsequent 180 RINGING with same CallID and rseq causing such forked calls fail. So if I have > 1 branches, I would like to exclude_hf_value("Allow", "PRACK"); exclude_hf_value("Supported", "100rel"); -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

how to add users without db_mysql
by disa_369＠msn.com 31 Oct '25

31 Oct '25

Hello everyone. Is there a way to add users (sip) to kamailio without db_mysql? Thank you

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