sr-users November 2025

sr-users@lists.kamailio.org

22 participants
24 discussions

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

handling of REFER in Kamailio?
by Benoît Panizzon 28 Nov '25

28 Nov '25

Hi Gang While testing our interconnection signalling with a carrier it was noted that we pass on REFER from CPE to the origin of the call via an IC. We are supposed to handle them locally. I wonder if somebody has an example how to correctly do this in Kamailio. -- 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 2

Dispatcher: set_duid_state with "IX" continues sending OPTIONS probes
by Tiago Serra 25 Nov '25

25 Nov '25

Hello, I'm experiencing an issue with the dispatcher module where setting a destination state to "IX" (Inactive + No probing) using dispatcher.set_duid_state doesn't stop OPTIONS probing. Configuration: modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 60) Steps: - Destination is being probed normally - Execute: kamcmd dispatcher.set_duid_state ix 11 ma5q - kamcmd dispatcher.list shows FLAGS: "IX" - SIP OPTIONS continue being sent to this destination I was expecting that marking the destination with the X flag would stop the probing. Is this incorrect behavior or do I need to use other ds_probing_mode value? I'm using kamailio version 5.8.4 Thanks, Regards, Tiago Serra

3 3

Timer between 100 and 180/183
by avi＠avipc.net 25 Nov '25

25 Nov '25

Hey all I have a provider that will respond to an invite with a 100 and then hang, trying to find the right timer to use to combat this , any ideas? thanks

1 0

How to make RTPEngine transcode the sampling time from 10 milliseconds to 20 milliseconds
by 13715209697＠139.com 25 Nov '25

25 Nov '25

Hi, Team How to make RTPEngine transcode the sampling time from 10 milliseconds to 20 milliseconds? OS: Debian 12 Kamailio ver: 6.0.4 RTPEngine ver: mr13.5.2.0 UAC (ptime:40) --- Kamalio/RTPEngine ---- UAS（ptime:20） kamailio.cfg: route[NATMANAGE] { ... if(has_totag()) { rtpengine_manage("SIP-source-address replace-origin replace-session-connection"); } else { rtpengine_manage("SIP-source-address replace-origin replace-session-connection codec-mask-all codec-transcode-PCMA;ptime--20 always-transcode"); } } rtpengine.log: ``` [1764059397.148615] INFO: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [control] Received command 'offer' from 127.0.0.1:49176 [1764059397.148641] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [control] Dump for 'offer' from 127.0.0.1:49176: { "supports": [ "load limit" ], "sdp": "v=0 o=1001 1764055864 1764059397 IN IP4 192.168.118.35 s=Kapanga [1764055864] c=IN IP4 192.168.118.35 t=0 0 m=audio 5112 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=sendrecv a=maxptime:40 a=ptime:40 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,36 ", "flags": [ "SIP-source-address", "always-transcode" ], "replace": [ "origin", "session-connection" ], "codec": { "mask": [ "all" ], "transcode": [ "PCMA;ptime--20" ] }, "call-id": "01153738E7580CA64AF270E08C0026CA(a)192.168.118.57", "received-from": [ "IP4", "192.168.118.35" ], "from-tag": "9A4E71D6E3F7B60DB47A7F494D93454F", "command": "offer" } [1764059397.148677] INFO: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] replace-session-connection flag encountered, but not supported anymore. [1764059397.148694] NOTICE: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Creating new call [1764059397.148801] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] No matching media (index: 1) using subscription, just use an index. [1764059397.148824] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Subscribing media with monologue tag '' (index: 1) to media with monologue tag '9A4E71D6E3F7B60DB47A7F494D93454F' (index: 1) [1764059397.148838] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Subscribing media with monologue tag '9A4E71D6E3F7B60DB47A7F494D93454F' (index: 1) to media with monologue tag '' (index: 1) [1764059397.148850] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] setting other slot to 0, setting slot to 0 [1764059397.148857] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Updating codecs for offerer 9A4E71D6E3F7B60DB47A7F494D93454F #1 [1764059397.148877] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Adding codec pcma/8000/ (8) [1764059397.148886] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Adding codec telephone-event/8000/0-15,36 (101) [1764059397.148894] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Accepting codec pcma/8000/ (8) [1764059397.148900] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Accepting codec telephone-event/8000/0-15,36 (101) [1764059397.148906] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Updating offer codecs for answerer #1 [1764059397.148912] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Adding codec pcma/8000/ (8) [1764059397.148919] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Adding codec telephone-event/8000/0-15,36 (101) [1764059397.148929] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Stripping codec pcma/8000/ (8) due to strip=all or strip=full [1764059397.148944] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Stripping codec telephone-event/8000/0-15,36 (101) due to strip=all or strip=full [1764059397.148954] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Adding codec PCMA;ptime=20 for transcoding [1764059397.148962] WARNING: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Usage error: List of codecs empty. Restoring original list of codecs. Results may be unexpected. [1764059397.148985] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Adding codec pcma/8000/ (8) [1764059397.149001] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Adding codec telephone-event/8000/0-15,36 (101) [1764059397.149005] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Updating supplemental codecs for #1 [1764059397.149017] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Setting up codec handlers for #1 -> 9A4E71D6E3F7B60DB47A7F494D93454F #1 [1764059397.149023] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Default sink codec is pcma/8000/ (8) [1764059397.149026] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Checking receiver codec pcma/8000/1/ (8) [1764059397.149028] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Creating codec handler for pcma/8000/ (8) [1764059397.149036] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink codec for pcma/8000/ is pcma/8000/1/ (8) [1764059397.149040] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink supports codec pcma/8000/ (8) for passthrough (to 8) [1764059397.149046] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Shutting down codec handler for pcma/8000/1 [1764059397.149052] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Using passthrough handler for pcma/8000/ (8) with DTMF 101, CN -1 [1764059397.149067] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Checking receiver codec telephone-event/8000/1/0-15,36 (101) [1764059397.149073] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Creating codec handler for telephone-event/8000/0-15,36 (101) [1764059397.149078] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink codec for telephone-event/8000/0-15,36 is telephone-event/8000/1/0-15,36 (101) [1764059397.149090] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink supports codec telephone-event/8000/0-15,36 (101) for passthrough (to 101) [1764059397.149093] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Shutting down codec handler for telephone-event/8000/1 [1764059397.149103] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Using passthrough handler for telephone-event/8000/0-15,36 (101) with DTMF 101, CN -1 [1764059397.149115] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Updating supplemental codecs for #1 [1764059397.149130] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Setting up codec handlers for #1 -> 9A4E71D6E3F7B60DB47A7F494D93454F #1 [1764059397.149145] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Default sink codec is pcma/8000/ (8) [1764059397.149156] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Checking receiver codec pcma/8000/1/ (8) [1764059397.149167] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink codec for pcma/8000/ is pcma/8000/1/ (8) [1764059397.149176] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink supports codec pcma/8000/ (8) for passthrough (to 8) [1764059397.149190] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Shutting down codec handler for pcma/8000/1 [1764059397.149207] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Using passthrough handler for pcma/8000/ (8) with DTMF 101, CN -1 [1764059397.149228] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Checking receiver codec telephone-event/8000/1/0-15,36 (101) [1764059397.149236] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink codec for telephone-event/8000/0-15,36 is telephone-event/8000/1/0-15,36 (101) [1764059397.149246] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Sink supports codec telephone-event/8000/0-15,36 (101) for passthrough (to 101) [1764059397.149252] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Shutting down codec handler for telephone-event/8000/1 [1764059397.149256] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [codec] Using passthrough handler for telephone-event/8000/0-15,36 (101) with DTMF 101, CN -1 [1764059397.149286] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to find RTP/RTCP ports (allocation attempt = '0') [1764059397.149302] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to bind the socket for RTP/RTCP ports (allocation attempt = '1') [1764059397.149306] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to bind the socket for port = '10068' [1764059397.149439] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to bind the socket for port = '10069' [1764059397.149464] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Opened 2 socket(s) from port '10068' (on interface '192.168.118.57') for a media relay [1764059397.149533] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to find RTP/RTCP ports (allocation attempt = '0') [1764059397.149544] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to bind the socket for RTP/RTCP ports (allocation attempt = '1') [1764059397.149547] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to bind the socket for port = '10000' [1764059397.149561] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Trying to bind the socket for port = '10001' [1764059397.149570] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] Opened 2 socket(s) from port '10000' (on interface '192.168.118.57') for a media relay [1764059397.149594] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] set FILLED flag for stream, local 192.168.118.57:10000 remote 192.168.118.35:5112 [1764059397.149602] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [core] set FILLED flag for stream, local 192.168.118.57:10001 remote 192.168.118.35:5113 [1764059397.149637] INFO: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [control] Replying to 'offer' from 127.0.0.1:49176 (elapsed time 0.000980 sec) [1764059397.149653] DEBUG: [01153738E7580CA64AF270E08C0026CA(a)192.168.118.57]: [control] Response dump for 'offer' to 127.0.0.1:49176: { "sdp": "v=0 o=1001 1764055864 1764059397 IN IP4 192.168.118.57 s=Kapanga [1764055864] t=0 0 m=audio 10068 RTP/AVP 8 101 c=IN IP4 192.168.118.57 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,36 a=sendrecv a=rtcp:10069 a=ptime:40 a=maxptime:40 ", "result": "ok" } ``` codec-transcode-PCMA;ptime--20 doesn't seem to work. The response for 'offer' contains `ptime:40`. What I really want is to be able to pass 'ptime=20' in the flags, but the rtpengine module doesn't seem to support it. Thanks!

1 1

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

how to jsonrpc_exec() to remote server?
by Benoit Panizzon 24 Nov '25

24 Nov '25

Hi Gang We use a mysql database to store call forwarding information for customers. To reduce the amount of mysql queries from the config, we use an expiring htable to cache this information. This has the drawback that if a customer alters his call forwarding settings this might not be immediately active as long as old data is still cached in the htable. To avoid this I am looking for a way to remove cached data when a customer calls a service number. jsonrpc_exec() calling htable.delete on the key works. We have a pair of redundant servers so I would need to also delete the key on another server via RPC HTTP request. How can I perform an jsonrpc call via HTTP to a remote server? Mit freundlichen Grüssen -Benoît Panizzon- -- 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 2

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

How to access all branches in the failure_route
by 13715209697＠139.com 20 Nov '25

20 Nov '25

Hi team How to access all branches in the failure_route? ··· SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 220.231.180.131:12290;branch=z9hG4bK7699.12bab93a2131301bddba965165d7b8a4.0 Via: SIP/2.0/UDP 192.168.99.203:5090;received=192.168.99.203;rport=5090;branch=z9hG4bKeS4Hy8evHZa4N To: <sip:1001@192.168.99.203:12290>;tag=a8016513 From: <sip:9196@192.168.99.203:12290>;tag=rQSg2p11ptytp Call-ID: d25b05e3-3fb6-123f-0ba6-000c29943390 ··· failure_route[MANAGE_FAILURE] { xlog("$sel(via[1].branch) + $T_branch_idx\n"); } The obtained value is z9hG4bKeS4Hy8evHZa4N, which is the branch generated by UAC. But I also want to obtain the branch automatically generated by Kamailio. How can I do this? Thanks!

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sr-users November 2025