- sr-users - kamailio.org

REG_MAR_REPLY never get executed
by stinktohighheaven＠gmail.com 26 Feb '25

26 Feb '25

Using Kamailio v5.3, based on the example for scscf found here: https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/scscf/ka… after calling ims_www_challenge("REG_MAR_REPLY", "$var(realm)", "$var(alg)"); the control never goes to REG_MAR_REPLY route block, but it goes to the start of the requet_route block. here is the kamailio.cfg script: request_route { ....... /* Handle Registrations */ if (is_method("REGISTER")) { route(REGISTER); exit; } ...... } /** * Handling of REGISTER requests */ route[REGISTER] { ...... #!ifdef WITH_AUTH if (!ims_www_authenticate("$var(realm)")) { #!else if ($var(alg) == "3GPP-Digest" && !ims_www_authenticate("$var(realm)")) { #!endif if ($? == -2) { send_reply("403", "Authentication Failed"); exit; } else if ($? == -3) { send_reply("400", "Bad Request"); exit; } else if ($? == -9) { xdbg("Authentication re-sync requested\n"); ims_www_resync_auth("REGISTER_RESYNC_REPLY", "$var(realm)"); exit; } else { /* user has not been authenticated. Lets send a challenge via 401 Unauthorized */ xdbg("About to challenge! auth_ims\n"); ims_www_challenge("REG_MAR_REPLY", "$var(realm)", "$var(alg)"); exit; } } else { xdbg("Auth succeeded\n"); /* We need to check if this user is registered or not */ if (!impu_registered("location")) { save("PRE_REGISTER_SAR_REPLY", "location"); exit; } else { isc_match_filter_reg("1", "location"); save("REGISTER_SAR_REPLY", "location"); exit; } } } } /** * Handling of REGISTER MAR replies */ route[REG_MAR_REPLY] { xlog("L_DBG","maa_return code is $avp(s:maa_return_code)\n"); /* this is async so to know status we have to check the reply avp */ switch ($avp(s:maa_return_code)) { case 1: /* success */ xdbg("MAR success - 401/407 response sent from module\n"); break; case -1: /* failure */ xerr("MAR failure - error response sent from module\n"); break; case -2: /* error */ xerr("MAR error - sending error response now\n"); send_reply("503", "MAR failed"); break; default: xerr("Unknown return code from MAR, value is [$avp(s:maa_return_code)]\n"); send_reply("503", "Unknown response code from MAR"); break; } exit; } and here's the log: 12(18) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/usr/local/etc/kamailio/kamailio.cfg] l=757 a=27 n=ims_www_challenge 12(18) DEBUG: ims_auth [authorize.c:293]: challenge(): Looking for route block [REG_MAR_REPLY] 12(18) INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [-2] 12(18) DEBUG: ims_auth [authorize.c:317]: challenge(): Need to challenge for realm [ims.mnc071.mcc432.3gppnetwork.org] 12(18) DEBUG: ims_auth [authorize.c:324]: challenge(): Checking if REGISTER is authorized for realm [ims.mnc071.mcc432.3gppnetwork.org]... 12(18) DEBUG: ims_auth [authorize.c:1449]: get_auth_userdata(): Searching auth_userdata for IMPU sip:432710000010044@ims.mnc071.mcc432.3gppnetwork.org (Hash 425) 12(18) DEBUG: ims_auth [authorize.c:460]: challenge(): Suspending SIP TM transaction 12(18) DEBUG: ims_auth [authorize.c:1529]: multimedia_auth_request(): Sending MAR 12(18) DEBUG: ims_auth [cxdx_mar.c:574]: cxdx_send_mar(): Successfully sent async diameter 70(76) INFO: ims_auth [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp (avp_code = 297, vendor_id = 0) 70(76) DEBUG: ims_auth [authorize.c:1303]: new_auth_vector(): new auth-vector with ck [d2deb48b5b3b614ba9112396e3c036fe] with status 0 70(76) DEBUG: ims_auth [authorize.c:1560]: pack_challenge(): setting QOP str used is [, qop="auth"] 70(76) DEBUG: ims_auth [authorize.c:1562]: pack_challenge(): QOP str used is [, qop="auth"] 70(76) DEBUG: ims_auth [authorize.c:1449]: get_auth_userdata(): Searching auth_userdata for IMPU sip:432710000010044@ims.mnc071.mcc432.3gppnetwork.org (Hash 425) 70(76) DEBUG: ims_auth [authorize.c:1458]: get_auth_userdata(): Found auth_userdata 70(76) DEBUG: ims_auth [authorize.c:1682]: add_auth_vector(): Adding auth_vector (status 1) for IMPU sip:432710000010044@ims.mnc071.mcc432.3gppnetwork.org / IMPI 432710000010044(a)ims.mnc071.mcc432.3gppnetwork.org (Hash 425) 70(76) DEBUG: ims_auth [cxdx_mar.c:486]: async_cdp_callback(): DBG:UAR Async CDP callback: ... Done resuming transaction 70(76) INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [1] 70(76) ERROR: tm [t_suspend.c:197]: t_continue_helper(): active transaction not found 70(76) DEBUG: ims_auth [cxdx_mar.c:87]: free_saved_transaction_data(): Freeing saved transaction data: async 11(17) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=380 a=5 n=route

4 7

PRACK Retransmission and 408 Timeout Issue in Kamailio
by 高秉寬 26 Feb '25

26 Feb '25

Dear Kamailio Developers, I am using Kamailio as a SIP server to facilitate a phone call, but I am encountering an issue where the call cannot be established due to Kamailio sending a 408 Request Timeout to the caller. Issue Details: - A (Caller) sends an initial INVITE to Kamailio, which then forwards it to B (Callee). The INVITE contains the following Supported header: Supported: timer,100rel,replaces,precondition,from-change,histinfo,tdialog I suspect the *precondition* option may be relevant to the issue. - *B* responds with a *183 Session Progress* message, which includes: Require: 100rel,precondition Kamailio correctly forwards this response to *A.* - *A* then sends a *PRACK* request to Kamailio. However, this request contains: Require: sec-agree Supported: timer Notably, *precondition* is absent from both the *Require* and *Supported* headers. - Kamailio forwards this *PRACK* to *B*, and *B* responds with *200 OK*, including: Supported: 100rel,from-change,tdialog - However, Kamailio does not seem to acknowledge *B's 200 OK* and instead *keeps retransmitting the PRACK* to *B*. B responds with *200 OK* again, but the cycle repeats. - After three retransmissions of *PRACK*, Kamailio eventually sends *408 Request Timeout* to *A*, leading *A* to send a *CANCEL* request. Device Used: - *Caller (A):* Asus ROG Phone 5 - *Callee (B):* Asus ROG Phone 6 Both are commercially available mobile phones. Possible Cause: According to *3GPP TS 24.229, Subclause 5.1.3.1*, If the received response with an SDP body includes a Require header with > the "precondition" option-tag, the originating UE shall include a Require > header with the "precondition" option-tag in responses with an SDP body to > subsequent requests that include an SDP body and include the "precondition" > option-tag in either the Supported or Require header field received > in-dialog. In this case: - The *183 Session Progress* from *B* contains Require: *precondition*. - The subsequent *PRACK* from *A* does not include *precondition* in either *Require* or *Supported*. Could this mismatch be the reason Kamailio keeps retransmitting *PRACK* and ignoring *B's 200 OK*? Or could there be another factor causing Kamailio to behave this way? If additional details are needed, please let me know. Best regards, Ping Kuan Kao

3 3

Best practices for proxy protocol and TLS offloading
by Alex Balashov 25 Feb '25

25 Feb '25

Hello, I am converting a Kamailio WSS/TLS gateway to pure TCP, with an external HAProxy handling TLS termination and emitting the proxied connection as plain TCP. I am using the PROXYv2 protocol, with `tcp_accept_haproxy=yes`, to convey the upstream network and transport-layer reachability info into Kamailio. I am trying to figure out best practices for mapping the contacts on those connections to the proxied connections themselves. For registrations, tcp_force_alias() (with `tcp_accept_aliases=yes`) works well, and inbound calls go to the right place. However, I'm not quite sure what to do with other flows, such as, for example, in-dialog requests on inbound calls going to the TLS endpoints. There are obviously a lot of possibilities, all or most of which I've tinkered with. These generally involve either {s.replace}-ing `;transport=tls` with `;transport=tcp` in the contacts received from the client, or using the traditional `nathelper` contact alias / RURI alias bag of tricks. However, I don't like the former solution because it leads to a non-compliant R-URI going to the endpoint (it's not the ;transport it sent in its contact), and I don't like the latter because it seems like there are too many moving parts. When tcp_force_alias() works so well for registrations, there must be some small linchpin I'm missing for normal request-reply flows. What is it? Thanks in advance! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

1 1

tm stats on prometheus
by Alexis Fidalgo 25 Feb '25

25 Feb '25

Hello all, i found that tm stats (kamcmd tm.stats) are not present in the response of xhttp_prom (prometheus), so in new configurations that we applied, we have no information/stats on reply codes. Is there a way to add this information to the xhttp_prom response? thanks in advance. regards

2 4

debug memory usage by AVP
by Sergey Safarov 25 Feb '25

25 Feb '25

In the "kamcmd mod.stats all shm" output i see to big value for "create_avp" metric Module: core { tcpconn_new(1205): 35104 counters_prefork_init(211): 61952 create_avp(178): 20737248 cfg_clone_str(132): 304 cfg_shmize(221): 1248 main_loop(1381): 16 Is possible to print AVP list and APV content without kamailio config change and restart?

1 0

Drop RTP Flow and move to another point
by Social Boh 24 Feb '25

24 Feb '25

Hello List, before spending much time on solving this scenario, I'd like to know if this kind of operation can be done using some Kamailio function (maybe dlg_req_within?) or other method. The scenario is: a B2B-1 server call two different destinations, one is a SIP trunk and another is another B2B-2. Both calls go through Kamailio which uses RTPEngine for the media flow. The idea is to drop the first B2B from RTP Flow (leave for signaling) when both calls are answered and leave everything to Kamailio-RTPEngine, like: LegA: Kamailio-RTPEngine -> B2B-2 LegB: Kamailio-RTPEngine -> SIP trunk Thank you in Advance Regards --- I'm SoCIaL, MayBe

1 0

Kamailio uses an ephemeral IP port instead of the port defined in $fs
by Leonid Fainshtein 24 Feb '25

24 Feb '25

Hi, I am using Kamailio v.5.8.4 on Ubuntu 24.04, kernel 6.8.0-51-generic In the configuration script I explicitly set $fs="udp:adrr:port". In normal situations, it works properly and I see that the UDP SIP requests are sent from the requested IP and port. Due to some reasons, free space on the server root filesystem was exhausted. I cleaned up the file system and everything looked good except the fact that Kamailio started to send messages from ephemeral IP ports instead of the port defined in $fs. Restart of Kamailio didn't solve the problem. But the server reboot fixed the issue. I don't understand how the lack of free disk space can cause such a problem and I'm afraid that there is another reason that I am missing. Has anybody seen such a problem? Best regards, Leonid

3 3

Issue with registrations
by Anthony Ford 24 Feb '25

24 Feb '25

Hi Version = 5.6.0-dev0 We are seeking help to resolve an issue with registering devices to 3CX (All 3CX servers, have tried 8 different ones) . all the auth info including passwords etc are identical on the 3cx servers. We use Kamailio to register mobile sim cards to 3rd party sip servers i.e. 3CX. We send the normal Username, server addresses and auth details but are experiencing failures with 3CX rejecting the registration and return Proxy Auth failure 407. We have noticed in the pcaps from both sides that the expire value is changing from 180 as an example to 0. As an example In the uac reg_dump the values are : l_uuid: +447529135850<tel:+447529135850> l_username: +447529135850<tel:+447529135850> l_domain: sbc-mt.asim.apalo.io<http://sbc-mt.asim.apalo.io/>;transport=udp r_username: 1001 r_domain: egte.3cx.uk<http://egte.3cx.uk/> realm: <null string> auth_username: FdlHDC1IvA auth_password: extracted auth_ha1: none auth_proxy: sip:egte.3cx.uk:5060 expires: 60 flags: 0 diff_expires: -1738853736 timer_expires: 0 reg_init: 1738850836 reg_delay: 0 contact_addr: k01.asim.apalo.io<http://k01.asim.apalo.io/>;transport=tcp socket: <null string> [cid:image001.png@01DB86AB.2701F190] [cid:image002.png@01DB86AB.2701F190] [cid:image003.png@01DB86AB.2701F190] Looking forward to hearing from you. Many thanks Tony Ford

1 0

Outdated location information on deleted registrations
by Mathias Schneuwly 19 Feb '25

19 Feb '25

Hi I'm using Kamailio 5.8.4 and the python KEMI module. When I move the registration from one device to another, I can see for a while two entries with 'kamctl ul show' where the Expires parameter of the "old" registration is set to "deleted" and the "new" registration shows the seconds until the registration expires. "Address": "sips:200101@10.0.1.130:42323 ;transport=tls", "Expires": "deleted", -- "Address": "sips:200101@10.0.1.129:56374 ;transport=tls", "Expires": 3600, So far so good. When I try to fetch the actual registration address using "registrar.reg_fetch_contacts" or "registrar.lookup_xavp", I'm always getting the "old" deleted registration address instead of the new address as long as the deleted record is not removed. Feb 18 10:05:51 ttel /usr/sbin/kamailio[2356327]: WARNING: {1 511 INVITE 952600514-5060-52(a)BA.A.C.BCI} <core> [core/kemi.c:157]: sr_kemi_core_log(): reg_fetch_contacts: sips:200101@10.0.1.130:42323;transport=tls Feb 18 10:05:51 ttel /usr/sbin/kamailio[2356327]: WARNING: {1 511 INVITE 952600514-5060-52(a)BA.A.C.BCI} <core> [core/kemi.c:157]: sr_kemi_core_log(): lookup_xavp: aor=200101, uri=sips:200101@10.0.1.130:42323;transport=tls, socket=tls:10.0.1.1:5061, dsturi=None I also see that the database is only updated after the deleted entry is removed. Until then, the database still contains the old address. Btw, I'm using "modparam("usrloc", "db_mode", 2)", which would explain the delayed persisting into the database. 62|uloc-67b44aae-23f48e-2|200101||sips:200101@10.0.1.130:42323 ;transport=tls|||2460724.91858218|-1.0|482845299-5060-1(a)BA.A.B.BDA|2002|2460724.87691551|0|0|Grandstream GXV3350 1.0.3.52|tls:10.0.1.1:5061 |7135|<urn:uuid:00000000-0000-1000-8000-C074AD188038>|1|0|52|0|16 I'm asking myself whether the behavior is correct that "registrar.reg_fetch_contacts" and "registrar.lookup_xavp" are returning the deleted information instead of the "new" registration information? If yes, is there another way how I could get the "new" registration information? I'm not really interested in the old outdated information... Best regards Mathias

2 2

Info: maintenance work for kamailio.org server on Tuesday, Feb 18, 2025
by Daniel-Constantin Mierla 18 Feb '25

18 Feb '25

Hello, there is a planned mainenance work on the infrastructure for the server hosting kamailio.org services (among them: the main website, wiki, mailing lists) on next week, Tuesday, February 18, 2025, to start about 10:00am Berlin time (CET). Expect some intervals of unavailability, we hope they will be short and most of them not noticeable for the majority accessing these services during the day. Once the work is finished, an announcement will be sent out. The apt repository, github project and the matrix chat channels should not be affected, they can be accessed as usual and can be used a backup communication channels if anyone observes something that has to be addressed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 2

Asterisk ConfBridge behind Kamailio+Rtpengine
by Rahul mm 18 Feb '25

18 Feb '25

Dear all, Thanks for the amazing product ! I’m trying my hands on a scenario involving Asterisk for video_sfu and a bunch of webrtc and sip clients. The condition is that every call must essentially be a conference call for the participants to add more dynamically. I front ended asterisk with Kamailio and RTPEngine for ssl offloading, load balancing, and registrations. So when one user is present in the conference (his invite SDP containing one audio and one video line) i get local video and MoH back, but when another user joins in and a re-invite is send from asterisk containing two video lines - one for the sendrecv and other for send only, rtpengine doesn’t process the sendonly video line - keeps it as is RTP/AVP instead of making it *UDP/TLS/RTP/SAVPF.* *Is there a way to achieve that? or I’m on the wrong path ?* BR, RM

1 0

Kamailio - asterisk registration issue & transparent proxy
by Enzo Damato 17 Feb '25

17 Feb '25

Hi. I'm working on setting up kamailio as a load balancer for some asterisk instances. I've build a configuration that is working for the most part, but I am encountering issues with calls from the asterisk instances to users who's registrations were processed through the kamailio instance. When these users register, asterisk saves thier AOR as the address of the kamailio instance, and then subsequently tires to send invites to kamailio, instead of to the UAC. Since kamailio just proxied the registration, without actually registering the user, it doesn't know what to do with the invite, and the call fails. I would like to configure kamailio to rewrite the sip requests so that all traffic appears to be coming from the end user, rather than from the kamailio instance, essentially making the proxy transparent. Is this possible? Thank you, Enzo Damato

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Prioritize specific codecs
by Temilolu Fayomi 17 Feb '25

17 Feb '25

Hello, I am working with inbound calls on kamailio (specifically fax), and was wondering if there was a way to re-prioritize the codecs found in the sdp of an invite. I've looked at using the sdpops module, but haven't been able to re-arrange the priority of the codecs listed. I've tried removing codecs to help incoming faxes switch over to t38 per my upstream providers instructions (calls will do a t38 re-invite if the g729a is the highest priority). I am just looking for which direction to go from here or if I misunderstand the nature of faxing and kamailio. Thank you in advance, Temilolu Fayomi

3 3

Constant crashes across versions
by Jon Bonilla (Manwe) 17 Feb '25

17 Feb '25

Hi all I have a setup of several proxies behind a load balancer and all of them have several coredumps every day. I've tried versions 5.3, 5.5 and 5.8 and all of them crash I didn't have dbgsym packages installed but now I've built a 5.8.4 version in one of them and can gdb the coredumps. Can't see anything particular and I'm wondering if I'm doing anything wrong. I'm descompressing the coredump with lz4cat. Sometimes I see that the system generates 3 coredumps at the same time. I've tried running gdb in one of them and exec "bt full" I usually see "can't access memory" in some of them but inb others I don't see anything relevant. I'm attaching one coredump. I don't even know if I'm doing it properly. Could you please guide me to how to debug what's going on? thanks

2 4

Info: sipexer v1.2.0 released
by Daniel-Constantin Mierla 11 Feb '25

11 Feb '25

Hello, a short note to announce that that sipexer v1.2.0 was released: - https://github.com/miconda/sipexer/releases/tag/v1.2.0 It is a modern and flexible SIP testing cli tool that can be used to send SIP signalling messages over UDP/TCP/TLS/WebSocket. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

DNS Unlinked Warnings
by John Larsen 10 Feb '25

10 Feb '25

Hi Everyone, We recently attempted an upgrade from kamailio 5.5.7 to 5.7.6. We immediately started to receive a log of warning logs similar to `dns_hash_put_shm_unsafe(): unlinked item 0x7f08799dcd90` The code shows that this warning log was added to the DNS cache in september 2023 - but its unclear what it means to have an unlinked item exactly? Is anyone able to shed any light on how DNS entries might become unlinked, and what the implications of it are? thanks John Larsen John Larsen Tech Lead EJohn.Larsen(a)symbio.global W symbio.global This communication is intended only for the person to whom it is addressed and may contain confidential material. If you received this communication in error, please inform the sender immediately and delete all copies. Please think of the environment before printing this email.

2 1

switch between "sips:" and "sip:" schema
by James Lipski 10 Feb '25

10 Feb '25

Hello, I have a kamailio installation where it's basically doing transport protocol translation. To go into more detail, I have a UA communicating over TLS towards kamailio/ and kamailio forwards the request to a freeswitch installation via UDP, so call-flow is as follows: UA <-- (TLS) --> Kamailio <-- (UDP) --> Freeswitch For the most part, this does work/ it's only when I use a UA that uses "sips:" as the schema instead of "sip:" where issues occur -- in my current topology, freeswitch has some weird behavior where it doesn't disconnect the call properly. At least based on my testing, I believe the issue is relating to the schema -- as the UA in question does have the ability to switch between schemas/ when using "sip:" instead of "sips:" over TLS, no issues with the same call-flow. In the event that I do come across a UA that doesn't have that ability, I was wondering if kamailio has the ability to switch schemas when it sends traffic over the UDP connection. I'm able to replace most aspects of the packet with "sip:" however the only header which contains the "sips:" schema is the contact header. With this alone, it propagates to the B-leg side (UDP side) of the call; and will cause freeswitch to not disconnect calls properly. I'm using the topos module to obfuscate my topology, not sure if that matters. -------- For example, on the A-leg side of the call, kamailio receives this: INVITE sips:12125551000@dev-sip.server.com:5061 SIP/2.0 Via: SIP/2.0/TLS 172.16.1.194:5061;branch=z9hG4bK236557042;rport;alias From: <sips:100000000@dev-sip.server.com>;tag=444048970 To: <sips:12125551000@dev-sip.server.com> Call-ID: 1326419916-5061-4(a)BHC.BG.B.BJE CSeq: 21 INVITE Contact: <sips:100000000@172.16.1.194:5061;transport=tls> Proxy-Authorization: Digest username="100000000", realm="dev-sip.server.com", nonce="Z6Ig0WeiH6Vdxtvej0I3XGCTUAEfh+0/", [uri="sips:12125551000@dev-sip.server.com](mailto:uri=)", response="43c1487d077db36a70fffdd1e25b3b52", algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream HT812 1.0.57.1 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 1034 v=0 o=100000000 8000 8000 IN IP4 172.16.1.194 s=SIP Call c=IN IP4 172.16.1.194 t=0 0 m=audio 5004 RTP/SAVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=crypto:1 AEAD_AES_256_GCM inline:UlX3PAjNBP/aQwtwyf5ymsfJciB+VdFxMElO5zLVooUqmcEzZ8YyQQk9stI=|2^32 ------------------------------------------------------ On the B-leg side of the call, kamailio forwards the following to freeswitch: INVITE sip:12125551000@fs.server.com SIP/2.0 Via: SIP/2.0/UDP 10.64.52.113:5080;branch=z9hG4bK1552.89ec9666294034a5611f908f2a6a427e.0;i=f From: "Test" <sip:16461234567@dev-sip.server.com>;tag=444048970 To: <sip:12125551000@fs.server.com> Call-ID: 1326419916-5061-4(a)BHC.BG.B.BJE CSeq: 21 INVITE Max-Forwards: 15 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 316 Contact: <sips:btpsh-67a21ca2-2dbe86-1@10.64.52.113:5080> v=0 o=100000000 8000 8000 IN IP4 10.64.52.113 s=SIP Call c=IN IP4 10.64.52.113 t=0 0 m=audio 13060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=ptime:20 -------------------------------------------------- As with the above, I was wondering if there is a way within kamailio to switch schemas on the b-leg side of this call example; specifically is there a way to update the contact header so it doesn't use the "sips:" schema when it traverses the UDP connection. Thank you for you time.

4 5

Auth_ephemeral module setup
by Anoop L 04 Feb '25

04 Feb '25

Hi, I'm trying to understand and setup Auth_ephemeral for authentication. My goal is to authenticate users dynamically using an web service for that I bumped into auth_ephemeral module. In the module how is a request made to the web service? Are username and password fetched from Pseudo-Variables variable? Is this module to be worked with xhttp module for http requests? Any help is much appreciated.

1 0

Auth_ephemeral Module Setup
by anoop96115＠outlook.com 04 Feb '25

04 Feb '25

Hi, I'm trying to understand and setup Auth_ephemeral for authentication. My goal is to authenticate users dynamically using an web service for that I bumped into auth_ephemeral module. In the module how is a request made to the web service? Are username and password fetched from Pseudo-Variables variable? Is this module to be worked with xhttp module for http requests? Any help is much appreciated.

1 0

Testing Kamailio IMS Modules with IPSec as P-CSCF
by Saeed Mogharian 03 Feb '25

03 Feb '25

Hi, I’m looking for suggestions on how to test Kamailio IMS modules, specifically for the IMS registration process over IPSec when Kamailio is configured as the P-CSCF node. Are there any scripts or IMS clients available that can help with this? Thanks in advance!

2 1

Kamailio 6.0 presentation at FOSDEM 2025
by Henning Westerholt 31 Jan '25

31 Jan '25

Hello, just a quick announcement, that tomorrow afternoon will be a presentation about Kamailio 6.0 at the FOSDEM conference in Brussels, Belgium [1]. Usually all the talks in this room are streamed online as well. I am happy also to meet during the conference to chat about Kamailio or other topics, just send me a (direct) message. Best regards, Henning [1] https://fosdem.org/2025/schedule/event/fosdem-2025-5978-kamailio-6-0-develo… -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Strange memory stats
by Barry Flanagan 30 Jan '25

30 Jan '25

Hi, I have a 5.8.4 runing on Bookworm that after a while reports very strange memory stats like this: shmem.fragments: '729' shmem.free_size: '18446744009887078560' shmem.max_used_size: '67043934736' shmem.real_used_size: '67043698528' shmem.total_size: '3221225472' shmem.used_size: '67034971568' The settings are: SHM_MEMORY=3072 PKG_MEMORY=256 The shmem.free_size and max_used/max_size seem quite wrong. The output of "mod.stats all shm" shows the below: - 'Module: jsonrpcs' - Total: 0 - 'Module: core' - Total: 7891296 - 'Module: rr' - Total: 0 - 'Module: dialog' - Total: 146144 - 'Module: permissions' - Total: 130672 - 'Module: auth' - Total: 0 - 'Module: topoh' - Total: 0 - 'Module: sl' - Total: 14048 - 'Module: siptrace' - Total: 79184 - 'Module: sqlops' - Total: 0 - 'Module: acc' - Total: 0 - 'Module: xlog' - Total: 0 - 'Module: dmq' - Total: 1520 - 'Module: avpops' - Total: 0 - 'Module: rtpengine' - Total: 69328 - 'Module: db_mysql' - Total: 0 - 'Module: geoip2' - Total: 32 - 'Module: pv' - Total: 0 - 'Module: registrar' - Total: 0 - 'Module: textops' - Total: 0 - 'Module: htable' - Total: 6375584 - 'Module: tsilo' - Total: 74048 - 'Module: usrloc' - Total: 554240 - 'Module: tm' - Total: 6812224 - 'Module: websocket' - Total: 2551536 - 'Module: pike' - Total: 6880 - 'Module: tls' - Total: 5462960 - 'Module: tmx' - Total: 21968 - 'Module: kex' - Total: 3536 - 'Module: cfgutils' - Total: 128 - 'Module: sipdump' - Total: 32 - 'Module: domain' - Total: 5920 - 'Module: dispatcher' - Total: 20144 Thanks! -Barry

2 8

Kamailio v6.0.0 Released - new major version is out
by Daniel-Constantin Mierla 29 Jan '25

29 Jan '25

Kamailio v6.0.0 is out – it comes with 5 new modules and important updates to the core components and existing modules. The increment of the first version number reflects several important updates, among them: multi-threaded option for UDP receiving, option for specifying a range of ports to listen to, support for expressions with variables for most of the module functions parameters, internal libraries are directly inked to the core, support to use cmake to compile and install Kamailio. You can read a bit more detailed release notes at: * https://www.kamailio.org/w/kamailio-v6.0-0-release-notes/ Many thanks to all developers and community members that made possible this release. v6.0.0 brings more flexibility and optimizations across many existing components and modules, new parameters, functions, variables and transformations. Enjoy Kamailio v6.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

2 2

Releasing v5.8.5
by Daniel-Constantin Mierla 29 Jan '25

29 Jan '25

Hello, I am considering to release Kamailio v5.8.5 (out of branch 5.8) on Thursday, Jan 23, 2025. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 2

MS Teams BYE routing
by Barry Flanagan 28 Jan '25

28 Jan '25

Hi all, I have a Teams gw working apart from one issue., In a call originated in Teams,. if the called party hangs up, the BYE is sent to api-du-b-euwe.pstnhub.microsoft.com:443 which does not work. This is taken from the Contact header of the ACK I presume. Here is the ACK to the 200 OK: 28/01/2025 17:08:26.560 +00:00: 52.114.75.24:53728 -> 100.100.198.241:5061 ACK sip:9000@teams.domain.com:5061;transport=tls;tps=atpsh-67990f01-8ed8-1 SIP/2.0 FROM: "Teams User" <sip:+35314375362@sip.pstnhub.microsoft.com:5061;user=phone>;tag=c29cb7a79ddf40f9ba909f3f1034a770 TO: <sip:9000@teams.domain.com:5061>;user=phone;tag=as6e287620 CSEQ: 1 ACK CALL-ID: e49f7df631ee576286ecf159e81952fe MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK46d28f33 CONTACT: <sip:api-du-b-euwe.pstnhub.microsoft.com:443;x-i=0781e8a1-91e6-4f6a-bc4a-73d6890827d4;x-c=e49f7df631ee576286ecf159e81952fe/d/8/703117d8337e4975a8216a54f59035c2> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2025.1.20.11 i.EUWE.7 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY ..and here is the BYE. 28/01/2025 17:08:36.119 +00:00: 100.100.198.241:5060 -> 52.123.134.13:443 BYE sip:api-du-b-euwe.pstnhub.microsoft.com:443;x-i=0781e8a1-91e6-4f6a-bc4a-73d6890827d4;x-c=e49f7df631ee576286ecf159e81952fe/d/8/703117d8337e4975a8216a54f59035c2 SIP/2.0 Record-Route: <sip:teams.domain.com:5061;transport=tls;ftag=as6e287620;lr;r2=on> Record-Route: <sip:100.100.198.241:5060;ftag=as6e287620;lr;r2=on> Via: SIP/2.0/UDP 100.100.198.241;branch=z9hG4bKc89c.0c1cf04a771c30293432e5f1f59331c8.0 Via: SIP/2.0/UDP 102.102.139.182:5060;rport=5060;branch=z9hG4bK3c42c163 Max-Forwards: 69 From: <sip:9000@teams.domain.com:5061>;user=phone;tag=as6e287620 To: "Teams User"<sip:+35314375362@sip.pstnhub.microsoft.com:5061;user=phone>;tag=c29cb7a79ddf40f9ba909f3f1034a770 Call-ID: e49f7df631ee576286ecf159e81952fe CSeq: 102 BYE Content-Length: 0 How can I get around this? Using a pretty standard WITHINDLG route. I am using topos but not manipulating the Contact header. Thanks! -Barry

4 8

RTPengine module - pause/resume recording
by Who AmI 28 Jan '25

28 Jan '25

Hi list, I have been looking at the recording daemon in RTPengine with Kamailio controlling this via rtpengine_manage() flags: "record-call=on" in the initial Invite, or 200OK based on direction. I tested and this works brilliantly for the full calls, but when looking at pause/resume the module documentation is bare and I am a bit confused. It looks like the ngcp docs and Opensips module docs (for RTPengine) describe the pause/resume as just another start/stop flag. Use case is simple, all calls are recorded, but we want the option to pause, then resume once any private or personal information exchange has completed. My question is also simple - Is this the recommended way to pause and then resume recordings or have I missed something? The Kamailio documentation doesn't explicitly say this ( https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html#rtpengi…) so just looking for advice if this is the best route to go. Thanks, John.

2 2

Re: [sr-dev] Re: Branching 6.0 series
by Daniel-Constantin Mierla 28 Jan '25

28 Jan '25

Hello, On 20.01.25 08:25, Daniel-Constantin Mierla via sr-dev wrote: > Hello, > > I think we should branch it on Wednesday, Jan 22, 2025, no matter on > what stage we are with cmake support. The old-Makefiles should be kept > anyhow (as they are or in a special folder with an easy way to recover > them), because there are many bits and pieces that can be discovered > later when more users. > > Because there were no many bugs reported to the C code specific to 6.0, > we can aim releasing on January 29, 2025, with old-Makefile still to be > used for tasks that are not yet covered by cmake. shall we still aim for releasing v6.0.0 on this upcoming Wednesday, January 29, 2025? We the old-Makefiles kept, it is doable, the wiki pages needs to be created for this release series, which I can do tomorrow, as I waited to see if how it should be the recommended method for compilation/installation. If the feeling is that we should wait, then probably the target date has to be shifted somewhere in the week starting on February 10, 2025, because Fosdem follows this weekend and then some countries have a week holiday break. If no other opinions and change of decision, the release will be done on Wednesday, with content using the old-Makefiles (to reuse existing tutorials) and reference to cmake variant when it is the case. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

2 1

defining configuration variables
by Sergio Charrua 27 Jan '25

27 Jan '25

Hello, just wondering if there is any way to define 2 constants as : # sets quarantine period the URL is set, in secs, after number of HTTP_RETRIES retries #!define SOME_VAR_1 60 # sets the configuration values for HTable module parameter #!define SOME_VAR_2 "some_configration_value" + SOME_VAR_1 I couldn't find anything in docs, and made several different ways without success. Any suggestions? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

2 2

UDP worker lock investigation
by Володимир Іванець 26 Jan '25

26 Jan '25

Hello everyone! I was wondering if it is possible to get information about a packet that is being possessed by the worker without increasing log level? I have an installation with Kamailio that is configured with two interfaces. Once in a while Kamailio stops accepting requests on one of the interfaces. It works fine for requests from other one. I think that worker processes might be locked while processing some requests. Enabling debug log is not an option because the traffic is quite high and the log file will not just be enormous, some lines will be missing. It looks like syslog will skip some of the log entries. So is it possible to check what packet worker process is currently processing? Or do I need to add some log entries with lower log level to trace requests manually? Thank you!

3 4

How to access registration contact user agent string in branch route after location lookup?
by Benoit Panizzon 24 Jan '25

24 Jan '25

Hi List I have some CPE which require special handling like removing 100rel and update from the supported methods because they misbehave. Previously, I was using reg_fetch_contacts and then appending the branches manually and stacking the UA in an AVP. With the branch index this allowed me to get the AVP containing the UA to which the call was branched in the branch route and perform filtering of methods based on UA, but it was complicated. Now I use the intended function lookup_to_dset() but now I miss how to access the register contact UA from within a branch route. Is there an intended way to access that information? 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 7

Use local_identity or custom data to set callerID
by matu lopi 23 Jan '25

23 Jan '25

Hi team, Is it possible to use either local_identity or custom data properties to define the CallerID during the RTCSession lifecycle? Thank you.

2 1

Issue with ACK on INVITE
by Barry Flanagan 23 Jan '25

23 Jan '25

Hi folks, Trying to hook up a new endpoint but am having an issue. Kamailio is in front of an Asterisk box. They send an INVITE, we send 100, then 200 OK. However, when they send their ACK, the RURI is not set to the Contact of the 200, instead it is<number>@<proxy ip>. This causes the ACK to get routed to the proxy itself and the call fails. 82.160.190.100:5060 = Kamailio IP Port 198.201.240.242:5080 = Asterisk IP/Port 182.33.174.10:5060 = Provider endpoint The far endpoint say they cannot fix the RURI - should I be able to handle this ACK below? My understanding is the ACK's RURI should be the Contact of the 200 OK. 200 OK sent from us to the Provider (Contact shows correct URI) ========================================================= 2025/01/15 17:14:49.470634 82.160.190.100:5060 -> 182.33.174.10:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.0 Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK5bbf9591;rport=5060 Record-Route: <sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Record-Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Record-Route: <sip:182.33.174.10;lr;ftag=as0b42eef3> From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3 To: <sip:1800715080@82.160.190.100>;tag=as4133584e Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060 CSeq: 102 INVITE Server: Asterisk PBX 18.9.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces *Contact: <sip:1800715080@198.201.240.242:5080>* Content-Type: application/sdp Content-Length: 314 , ACK Reply from Provider to Us (RURI points to Kamailio ip:port now =========================================================== 2025/01/15 17:14:49.478119 182.33.174.10:5060 -> 82.160.190.100:5060 *ACK sip:1800715080@82.160.190.100:5060 *SIP/2.0 Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.2 Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK55656469;rport=5060 Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes>,sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Max-Forwards: 69 From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3 To: <sip:1800715080@82.160.190.100>;tag=as4133584e Contact: <sip:0737965510@182.33.174.39:5060> Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060 CSeq: 102 ACK Thanks -Barry

5 5

Kamailio v5.8.5 Released
by Daniel-Constantin Mierla 23 Jan '25

23 Jan '25

Hello, Kamailio SIP Server v5.8.5 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.5. For more details about version 5.8.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/01/kamailio-v5-8-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Getting headers from xhttp/websocket
by Barry Flanagan 23 Jan '25

23 Jan '25

Hi, I need to be able to pull in a header provided by a webrtc endpoint so I can use it in my request_route. I understand that I will need to use shm/htable to make it visible but I am unable to get the xhttp header using $header(x-p-push), or any header by name. In my event_route[xhttp:request] I have the following: if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET") { .... xlog("L_INFO", "BF x-p-push header is $hdr(x-p-push) Contact is $hdr(Contact)\n"); } The log shows both x-p-push and Contact header as null. sipdump shows the headers present in the request. Am I missing something? Running Kamailio 5.8.4 Thanks -Barry

3 3

Development is open for future release of v6.1.x series
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, the branch 6.0 was created, therefore the master branch is open for adding new features, to be part of future release series v6.1.x. Any bug fix committed to master that applies to 6.0.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 6.0 or 5.8. Expect that v6.0.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v6.1.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime during the last part of 2025 or the beginning of 2026. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Branch 6.0 has been created
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, the branch 6.0 has been created, to be used for releasing v6.0.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-6.0 cd kamailio-6.0 git checkout -b 6.0 origin/6.0 Pushing commits in this branch: git push origin 6.0:6.0 Note that 6.0 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 6.0 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 6.0 will be out, respectively Kamailio v6.0.0. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Not able to get the $TV(un) inside xhttp module
by kaseenadh111＠gmail.com 22 Jan '25

22 Jan '25

Hi, I am running Kamailio 5.7.* RHEL I need to log the latency of each XHTTP request (milli or microseconds precision) so I tried using $TV(un) which always gives a 0 value. Can we access $TV(un) under XHTTP routes? event_route[xhttp:request] { $var(tv_http_req_start_time) = $TV(un); xlog("L_INFO", "$$var(tv_http_req_start_time)= $var(tv_http_req_start_time) and actual $$TV(un)= $TV(un)"); .... } Output: Jan 22 16:18:20 VM-123************ kamailio[346535]: INFO: <script>: $var(tv_http_req_start_time)= 0 and actual $TV(un)= 0 $TS - I have tried this, it gives the Unix in seconds precision but I want milliseconds precision. Thanks Kasee

1 0

Re: [sr-dev] Re: Branching 6.0 series
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, later today, in a few hours, I am going to create the branch 6.0. More details will be provided once it has been created. Cheers, Daniel On 20.01.25 08:25, Daniel-Constantin Mierla via sr-dev wrote: > Hello, > > I think we should branch it on Wednesday, Jan 22, 2025, no matter on > what stage we are with cmake support. The old-Makefiles should be kept > anyhow (as they are or in a special folder with an easy way to recover > them), because there are many bits and pieces that can be discovered > later when more users. > > Because there were no many bugs reported to the C code specific to 6.0, > we can aim releasing on January 29, 2025, with old-Makefile still to be > used for tasks that are not yet covered by cmake. > > Cheers, > Daniel > > On 15.01.25 11:16, Henning Westerholt via sr-dev wrote: >> Hello, >> >> yes, Xenofon is also waiting for a final reply from Sergey for the RPM packaging if now all is fine (was discussed as part of PR #4085). >> >> Cheers, >> >> Henning >> >>> -----Original Message----- >>> From: Victor Seva via sr-dev <sr-dev(a)lists.kamailio.org> >>> Sent: Mittwoch, 15. Januar 2025 10:59 >>> To: sr-dev(a)lists.kamailio.org >>> Cc: Victor Seva <linuxmaniac(a)torreviejawireless.org> >>> Subject: [sr-dev] Re: Branching 6.0 series >>> >>> Hi, >>> >>> please postpone. I didn't have time finalize the test of cmake builds on debian >>> packaging yet. >>> >>> On 15/1/25 7:18, Daniel-Constantin Mierla via sr-dev wrote: >>>> Hello, >>>> >>>> creating git branch 6.0 was planned for today and I wonder if it shall >>>> be done latter during the evening or it would be preferred to be >>>> postponed for two days or so, till end of the week/Friday, as I >>>> noticed yesterday some new work being done to cmake files for FreeBSD >>>> and maybe new commits are still coming. >>>> >>>> Of course, there is also the option to create the branch and then the >>>> new commits backported, but maybe postponing makes everything more >>>> convenient. >>>> >>>> In short, if anyone feels that postponing a bit creating the branch >>>> 6.0 simplifies their current work, let us know. If nobody ask for >>>> postponing, the branch will be created later towards the end of today. >>>> >>>> Cheers, >>>> Daniel >>> -- >>> ----------------------------------------------------------------- >>> | ⢀⣴⠾⠻⢶⣦⠀ Victor Seva | >>> | ⣾⠁⢠⠒⠀⣿⡁ linuxmaniac(a)torreviejawireless.org | >>> | ⢿⡄⠘⠷⠚⠋⠀PGP: 8F19 CADC D42A 42D4 5563 730C 51A0 9B18 CF5A >>> 5068 | >>> | ⠈⠳⣄⠀⠀⠀ Debian Developer | >>> ----------------------------------------------------------------- >> _______________________________________________ >> Kamailio - Development Mailing List -- sr-dev(a)lists.kamailio.org >> To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to the sender! > > -- > Daniel-Constantin Mierla (@ asipto.com) > twitter.com/miconda -- linkedin.com/in/miconda > Kamailio Consultancy, Training and Development Services -- asipto.com > > _______________________________________________ > Kamailio - Development Mailing List -- sr-dev(a)lists.kamailio.org > To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender! -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 1

Multi-line Via: rport added incorrectly
by James Browne 22 Jan '25

22 Jan '25

Hi I see that when kamailio adds rport to the Via header field of a request that has two Via values on the same line (comma-separated, of course), it adds the rport (and received) to the wrong value. I have this test kamailio.cfg for demonstration. children=1 loadmodule "sl" loadmodule "textops" loadmodule "nathelper" request_route { force_rport(); sl_send_reply(200, "OK"); } I send this OPTIONS (truncated) and get this response (truncated). The rport should be on the first Via value, not the second. OPTIONS sip:server SIP/2.0 Via: SIP/2.0/UDP 3.9.4.2:5060;branch=z9hG4bK503983089,SIP/2.0/UDP 1.3.4.1:5060;branch=z9hG4bK53805983059 From: <sip:client>;tag=LeonhardEuler To: <sip:server> SIP/2.0 200 OK Via: SIP/2.0/UDP 3.9.4.2:5060;branch=z9hG4bK503983089,SIP/2.0/UDP 1.3.4.1:5060;branch=z9hG4bK53805983059;rport=53805;received=127.0.0.1 From: <sip:client>;tag=LeonhardEuler To: <sip:server>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.49678e65 Server: kamailio (6.0.0-pre0 (x86_64/linux)) I've tested this with the latest commit (c994fb8) from kamailio. I can't think how this can be anything other than a bug. Should I log a bug report for this? James

2 4

Error when sending NOTIFYs to disconnected websocket clients
by Flavio Grossi 20 Jan '25

20 Jan '25

Hi, i have some SIP websocket clients (built using the SIP.js library) SUBSCRIBEd to other clients via kamailio v5.8.2. If one of the watchers suddenly disappears (e.g. the web page is reloaded), another connection is made and a new watcher appears in kamailio without the old one disappearing. For example: # kamctl rpc presence.watcher_list basic sip:1001@example.voismart.com { "jsonrpc": "2.0", "result": [ { "pres_uri": "sip:1001@example.com", "to_user": "1001", "watcher_user": "1000", "contact": "sip:1000@mfp0h2dhhmq2.invalid;transport=ws;alias=192.168.56.10~34166~5", "callid": "1o9b499kj6l65if0p30t", "user_agent": "sip.js/1.12.1", "expires": 1736422494, ... }, { "pres_uri": "sip:1001@example.com", "to_user": "1001", "watcher_user": "1000", "contact": "sip:1000@0j7qoabnv827.invalid;transport=ws;alias=192.168.56.10~34156~5", "callid": "8mjhrqifgm39up9t2qq3", "user_agent": "sip.js/1.12.1", "expires": 1736422492, ... } ], "id": 11753 } you can see the duplicated watcher in which the old connection is now invalid because the websocket client disappeared. In this situation, issuing a pres_refresh_watchers() call will fail to send any notify (even to the valid connection) because of this error: WARNING: <core> [core/msg_translator.c:3007]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: tm [t_msgbuilder.c:1423]: assemble_via(): via building failed ERROR: tm [t_msgbuilder.c:1614]: build_uac_req(): error while assembling Via ERROR: tm [uac.c:552]: t_uac_prepare(): Error while building message ERROR: presence [notify.c:1734]: send_notify_request(): in function tm t_request_within() ERROR: presence [notify.c:1829]: notify(): sending Notify not successful ERROR: presence [notify.c:1585]: query_db_notify(): Could not send notify for [event]=dialog ERROR: presence [presence.c:739]: pres_refresh_watchers(): sending Notify requests in which the building of the notify message fails because it cannot find a valid connection when building the via header. Is this a wanted behaviour? shouldn't it just fail the sending of the single message? Is there anything i can do to fix this? Trying to reproduce the same problem with a regular tcp client, it seems that the other notifys are correctly sent and only a timeout error is sent for the single message: ERROR: <core> [core/tcp_main.c:4622]: handle_tcpconn_ev(): connect 192.168.56.10:21994 failed which i think is the correct behaviour. Related question: is it possible to remove active watchers when the connection goes down? Thanks for the help on this.

2 5

Performance issues with KEMI
by Daniel W. Graham 20 Jan '25

20 Jan '25

Testing out KEMI functionality and running into performance issues. If I exceed 150 calls per second the network receive queue grows and Kamailio is unable to keep up with requests and they begin dropping. KEMI script for testing is just doing a stateless reply to invites. Using python3s module. I've played with Kamailio child processes and memory allocations, but there is no impact. I've also attempted some buffer / memory tweaking at the OS level, again with no impact. Increasing CPU cores and even running the test on bare metal results in the same. Example of receive queue at 150 calls per second - Netid State Recv-Q Send-Q Local Address:Port udp UNCONN 337280 0 x.x.x.x:5060 Just wondering if anyone has experienced similar issues or has an example of the performance they are seeing before I continue down this path. Thanks, - dan

5 9

Need help with serial branching using: t_load_contacts(1)
by Benoît Panizzon 20 Jan '25

20 Jan '25

Hi Gang I guess, I don't completely understand who to properly perform serial branching. I did try to follow the examples from: https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_next_con… This is, stripped down, the relevant config used. modparam("tm", "contacts_avp", "tm_contacts") modparam("tm", "contact_flows_avp", "tm_contact_flows") route[LOCATION] { lookup_to_dset("location", "$var(lookupuri)"); if (!t_load_contacts(1)) { xlog("L_ERR", "$cfg(route): ######### load_contacts failed\n"); sl_send_reply("500", "Server Internal Error - Cannot load contacts"); exit; } xlog("L_INFO", "$cfg(route): Contacts loaded: $cnt($xavp(tm_contacts))\n"); => Confirms, there is more than one contact loaded. if (!t_next_contacts()) { send_reply("480", "Not registered"); } t_set_fr(5000,1500); # Set 5 second timeout for LAB testing to quickly try the next contact. t_on_branch("BR_TO_CUST"); t_on_branch_failure("BR_TO_CUST_FAILURE"); exit; } event_route[tm:branch-failure:BR_TO_CUST_FAILURE] { xlog("BRANCH FAILED $T_reply_code to $rm message\n"); t_on_branch("BR_TO_CUST"); t_on_branch_failure("BR_TO_CUST_FAILURE"); if (t_next_contact_flow()) { xlog("BRANCH FAILED, Try Next\n"); t_relay(); } else { xlog("L_INFO", "No more flows\n"); t_reply("408", "Branch Timeout"); exit; } } What happens is: INVITE is sent to the first contact, who is replying RINGING. After 5 seconds the timeout is reached and the branch-failure route engaged. Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: ERROR: <script>: BRANCH FAILED 408 to INVITE message Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: INFO: <script>: No more flows Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: CRITICAL: tm [tm.c:1554]: ki_t_reply(): w_t_reply entered in unsupported mode To my understanding, t_next_contact_flow() should have loaded the next ds, but this does not seem to happen. What am I missing? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 3

Kamailio IMS, used with IPv6?
by Valentin Christoph 16 Jan '25

16 Jan '25

Anybody experiences with Kamailio IMS modules (ims_registrar_scscf, ims_usrloc_scscf, ims_isc, ........) and the use of IPv6 addresses? Any pitfalls? DNS issues? Thanks, Christoph

2 2

Asynchttp failover
by Sergio Charrua 15 Jan '25

15 Jan '25

Hello all! just wondering if anyone knows a good way to failover with the HTTP Async module. The standard HTTP Client module (synchronous) does implement some nice functionalities for failover between URL addresses, but the HTTP Async doesn't have any failover functionality... I have implemented some functions (within kamailio.cfg) for failover upon HTTP requests, but despite it working fine, it would be better to have that functionality on the HTTP Async module, just like HTTP Client has... Any technical reason for HTTP Async module not having failover functionalities? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

2 2

Module snmpstats reporting inconsistent call success metrics on intense peaks of failed calls
by Igor Potjevlesch 15 Jan '25

15 Jan '25

Hello! We are plotting the call success rate on each of our proxies. Using the Kamailio snmpstats module, we try to obtain three counters to plot them on a graph: Total Calls Active Calls Error Calls Most of the time, everything works well, and we obtain sensible values: snmpwalk -c mycommunity -v 2c W.X.Y.Z 1.3.6.1.4.1.34352.3.1.3.1.3.2 -O n .1.3.6.1.4.1.34352.3.1.3.1.3.2.1.0 = Gauge32: 309 -> Total Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 = Gauge32: 95 -> Active Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.3.0 = Gauge32: 214 -> Error calls However, when our proxies are facing intense peaks of failed calls in a short time span, sometimes, the "Error Calls" counter returns a value higher than "Total Calls". As a result, "Active Calls" suddenly reaches a value close to the maximum of a Gauge32 (because T o t a l C a l l s − E r r o r C a l l s < 0 ). snmpwalk -c mycommunity -v 2c W.X.Y.Z 1.3.6.1.4.1.34352.3.1.3.1.3.2 -O n .1.3.6.1.4.1.34352.3.1.3.1.3.2.1.0 = Gauge32: 153 -> Total Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 = Gauge32: 4294967226 -> Active Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.3.0 = Gauge32: 223 -> Error calls As a result, our graphs make no more sense. We are currently running kamailio 5.6.4. Any idea if it's a bug or a configuration issue? Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

3 7

Branching 6.0 series
by Daniel-Constantin Mierla 15 Jan '25

15 Jan '25

Hello, I am trying to figure out what would be the best date to branch 6.0 in the git repo. To allow a bit of time after returning from the winter holidays to tune more the cmake and packaging, I would think that January 15, 2025 could be a good choice. If nobody comes with other opinions/suggestions, then we will go for it. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

kamcmd tls.reload issue
by Yuriy Nasida 14 Jan '25

14 Jan '25

Hello, I am using letsencrypt cert and key and do not want to restart kamailio every 3 months to load new ones. I know that there is: kamcmd tls.reload method but it has an error for me. error: 500 - Error while fixing TLS configuration (consult server log) I am checking the logs and see: kamailio[3865480]: INFO: tls [tls_domain.c:345]: ksr_tls_fill_missing(): TLSs<default>: tls_method=3 kamailio[3865480]: INFO: tls [tls_domain.c:357]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/certs/my_cert.crt' kamailio[3865480]: INFO: tls [tls_domain.c:364]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:371]: ksr_tls_fill_missing(): TLSs<default>: ca_path='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:378]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:382]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 kamailio[3865480]: INFO: tls [tls_domain.c:390]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:397]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/certs/private.key' kamailio[3865480]: INFO: tls [tls_domain.c:401]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 kamailio[3865480]: INFO: tls [tls_domain.c:406]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 kamailio[3865480]: INFO: tls [tls_domain.c:410]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 kamailio[3865480]: NOTICE: tls [tls_domain.c:1168]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... kamailio[3865480]: ERROR: tls [tls_domain.c:590]: load_cert(): TLSs<default>: Unable to load certificate file '/etc/kamailio/certs/my_cert.crt' kamailio[3865480]: ERROR: tls [tls_util.h:49]: tls_err_ret(): load_cert:error:03000072:digital envelope routines::decode error (sni: unknown) kamailio[3865480]: ERROR: tls [tls_util.h:49]: tls_err_ret(): load_cert:error:0A00018F:SSL routines::ee key too small (sni: unknown) Any advice ? It's interesting that there are not any errors in case I restart kamailio. I can make TLS calls without problems. deb 12.5 version: kamailio 5.7.4 (x86_64/linux)

4 5

FOSDEM VoIP Dinner
by Torrey Searle 14 Jan '25

14 Jan '25

Hello all, As in previous years, I'm organising the VoIP dinner for FOSDEM (1 Feb @ 19:00) However unlike previous years (since the restaurant got upset about splitting the bill 30 ways). I will ask everyone to order their meal and pay in advance. (and I will pay the bill to the restaurant with my card) If you are wanting to attend the dinner please fill out the following form and follow the instructions for payment. https://form.jotform.com/250012543310033 Contact details are also in the form if you want to reach out to me with any further questions. Kind regards Torrey

5 4

ims_diameter_server not freeing shared memory
by marpett365＠gmail.com 09 Jan '25

09 Jan '25

Hi, I have run in to an issue when sending diameter requests using the ims_diameter_server module. Shared memory keeps increasing and are not being freed until restart of kamailio. This leads to all shared memory is being used and no more are able to be allocated. This only happens when the request is sent synchronously. When sent asynchronously shared memory is being freed as far as I can see. have anyone ran into this issue and found a remedy or can give some pointers how to proceed? Following functions have been used for sending requests. diameter_request([peer], appid, commandcode, message) diameter_request_async([peer], appid, commandcode, message) route[CHECK_LOCATION] { diameter_request("hss.ims.test.local", "16777217", "306", "[ " + "{ \"avpCode\":260, \"vendorId\":0, \"Flags\":64, \"list\":[ { \"avpCode\":266, \"vendorId\":0, \"Flags\":64, \"int32\":10415 }, { \"avpCode\":258, \"vendorId\":0, \"Flags\":64, \"int32\":16777217 } ] }, " + "{ \"avpCode\":277, \"vendorId\":0, \"Flags\":64, \"int32\":1 }, " + "{ \"avpCode\":283, \"vendorId\":0, \"Flags\":64, \"string\":\"ims.test.local\"}, " + "{ \"avpCode\":706, \"vendorId\":10415, \"Flags\":192, \"int32\":0 }, " + "{ \"avpCode\":700, \"vendorId\":10415, \"Flags\":192, \"list\": [ { \"avpCode\":601, \"vendorId\":10415, \"Flags\":192, \"string\":\"tel:" + $fU + "\" }]}, " + "{ \"avpCode\":707, \"vendorId\":10415, \"Flags\":192, \"int32\":0 }, " + "{ \"avpCode\":703, \"vendorId\":10415, \"Flags\":192, \"int32\":14 } ]"); xlog("L_INFO", "Sent Diameter request"); } Shared memory after restart kamcmd mod.stats cdp shm Module: cdp { save_peer_applications(779): 144 make_send_pipe(126): 320 dp_add_pid(92): 216 AAAAddRequestHandler(69): 40 cb_add(183): 8 cb_add(177): 32 cdp_sessions_init(178): 8 cdp_sessions_init(173): 8 cdp_sessions_init(165): 1024 cdp_sessions_init(157): 3072 cdp_sessions_init(149): 8 cdp_trans_init(67): 8 cdp_trans_init(60): 24 add_timer(150): 24 add_timer(145): 144 new_peer(77): 16 new_peer(75): 16 new_peer(72): 16 new_peer(69): 80 new_peer(66): 96 new_peer(60): 384 peer_manager_init(84): 8 peer_manager_init(83): 8 peer_manager_init(82): 8 peer_manager_init(79): 8 peer_manager_init(76): 16 worker_init(101): 16 worker_init(94): 128 worker_init(89): 32 worker_init(87): 32 worker_init(84): 8 worker_init(82): 48 timer_cdp_init(182): 8 timer_cdp_init(179): 16 diameter_peer_init_real(137): 8 diameter_peer_init_real(131): 16 diameter_peer_init_real(122): 16 diameter_peer_init_real(115): 8 diameter_peer_init_real(108): 8 diameter_peer_init_real(101): 8 diameter_peer_init_real(94): 8 new_routing_entry(88): 64 parse_dp_config(447): 16 parse_dp_config(348): 8 parse_dp_config(340): 16 parse_dp_config(333): 40 parse_dp_config(326): 144 quote_trim_dup(96): 456 new_dp_config(52): 184 Total: 7024 } Shared memory after 10 diameter requests Module: cdp { save_peer_applications(779): 144 AAACreateAVP(137): 5072 AAATranslateMessage(500): 1320 do_receive(600): 7392 make_send_pipe(126): 320 dp_add_pid(92): 216 AAAAddRequestHandler(69): 40 cb_add(183): 8 cb_add(177): 32 cdp_sessions_init(178): 8 cdp_sessions_init(173): 8 cdp_sessions_init(165): 1024 cdp_sessions_init(157): 3072 cdp_sessions_init(149): 8 cdp_trans_init(67): 8 cdp_trans_init(60): 24 add_timer(150): 24 add_timer(145): 144 new_peer(77): 16 new_peer(75): 16 new_peer(72): 16 new_peer(69): 80 new_peer(66): 96 new_peer(60): 384 peer_manager_init(84): 8 peer_manager_init(83): 8 peer_manager_init(82): 8 peer_manager_init(79): 8 peer_manager_init(76): 16 worker_init(101): 16 worker_init(94): 128 worker_init(89): 32 worker_init(87): 32 worker_init(84): 8 worker_init(82): 48 timer_cdp_init(182): 8 timer_cdp_init(179): 16 diameter_peer_init_real(137): 8 diameter_peer_init_real(131): 16 diameter_peer_init_real(122): 16 diameter_peer_init_real(115): 8 diameter_peer_init_real(108): 8 diameter_peer_init_real(101): 8 diameter_peer_init_real(94): 8 new_routing_entry(88): 64 parse_dp_config(447): 16 parse_dp_config(348): 8 parse_dp_config(340): 16 parse_dp_config(333): 40 parse_dp_config(326): 144 quote_trim_dup(96): 456 new_dp_config(52): 184 Total: 20808 } Shared memory after 170k requests do_receive(600): 131363752 AAATranslateMessage(500): 32326656 AAACreateAVP(137): 110067152 Total: 273764840 Br Martin

2 5

evapi not processing received messages
by Sergio Charrua 09 Jan '25

09 Jan '25

Hi all! I have implemented evapi module on Kamailio 5.8.4, calling a python script. Python script is listening to a socket. It receives messages (JSON) from Kamailio, makes an HTTP request somewhere, and sends the response back to Kamailio. The Python script is working fine: it connects to Kamailio, sends HTTP requests, and forwards a response back to Kamailio: I have confirmed using TCPDump that, in fact, the data is received and forwarded back. However, despite showing messages that the Python client script is correctly connected to Kamailio, the event_route[evapi:message-received] is never executed. Probably there is some detail that I am missing, but the docs and online info for evapi are very rare, and they all show very similar configuration. And as I wrote, I have confirmed using TCPDump that the data is received from Python client script to Kamailio node. What am I missing? This is my Kamailio cfg: [loads modules] [sets modparams] modparam("evapi","bind_addr","10.20.20.1:8888") debug=2 children=16 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " disable_sctp = yes force_rport = yes rundir="/tmp" request_route { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); } exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } route(TOEVAPI); exit; } event_route[evapi:connection-new] { xlog("LINFO","new connection from [$evapi(srcaddr):$evapi(srcport)]\n"); if($evapi(srcaddr)!="10.20.0.1") { evapi_close(); exit; } } event_route[evapi:connection-closed] { xlog("LINFO","connection closed by $evapi(srcaddr):$evapi(srcport)\n"); } event_route[evapi:message-received] { xlog("LINFO","received [$evapi(msg)] from $evapi(srcaddr):$evapi(srcport)\n"); jansson_get("t-index", "$evapi(msg)", "$var(t-index)"); jansson_get("t-label", "$evapi(msg)", "$var(t-label)"); $var(evmsg) = $evapi(msg); xlog("L_INFO", "preparing to resume transaction for processing: $var(tindex) / $var(tlabel)\n"); t_continue("$var(tindex)", "$var(tlabel)", "EVAPIRESPONSE"); } route[EVAPIRESPONSE] { xlog("L_INFO", "resumed transaction for processing: $T(id_index) / $T(id_label)\n"); [do stuff] } route[TOEVAPI]{ xlog("L_INFO", "suspended transaction: $T(id_index) / $T(id_label)\n"); evapi_async_relay( [some JSON string] ); exit; } Thanks in advance! Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 6

Kamailio and Faxing
by Temilolu Fayomi 09 Jan '25

09 Jan '25

Hello, I am having some issues ensuring t38 faxes are being received reliably. I am currently using kamailio as an inbound sbc with freepbx as the endpoint. I understand that potentially stripping other codecs except for g711a/u makes inbound faxes reliable, but beyond that I am having trouble determining how to ensure reliable faxing. Is this something kamailio can handle (such as an error correction mode or pass through setting or module?) I would appreciate any insight as I have hit a wall when it comes to this issue. Thank you, Temi

3 3

Re: WS and rtpengine_manage
by Knserbrave 08 Jan '25

08 Jan '25

test

2 4

deb-archive.kamailio.org unreachable
by Floimair Florian 08 Jan '25

08 Jan '25

Hi! It seems that deb-archive.kamailio.org is currently down. It resolves to IP 76.9.245.162 on my system Can someone with access to that please have a look. Thank you FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2459749227] commend.com LG Salzburg / FN 178618z

2 2

Kamailio crash
by Mitesh Thakkar 07 Jan '25

07 Jan '25

Hi, I have recently migrated from Kamailio 5.2.5 to Kamailio 5.5.7. After a migration, the Kamailio crashed. Find the details below. Thanks in advance. Back trace: ========= Core was generated by `/usr/local/sbin/kamailio -DD -E -m 1941 -M 970 -f /etc/kamailio/kamailio.cfg --'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8e27a2be15 in OPENSSL_cleanse () from /lib/x86_64-linux-gnu/libcrypto.so.3 (gdb) bt #0 0x00007f8e27a2be15 in OPENSSL_cleanse () from /lib/x86_64-linux-gnu/libcrypto.so.3 #1 0x00007f8e278faa8a in CRYPTO_clear_free () from /lib/x86_64-linux-gnu/libcrypto.so.3 #2 0x00007f8e27838175 in BUF_MEM_free () from /lib/x86_64-linux-gnu/libcrypto.so.3 #3 0x00007f8e27bc0c98 in SSL_free () from /lib/x86_64-linux-gnu/libssl.so.3 #4 0x00007f8e27c8cd6e in tls_h_tcpconn_clean_f (c=0x7f8db0fa1e90) at tls_server.c:701 #5 0x000055f7f42cc6e1 in _tcpconn_free (c=0x7f8db0fa1e90) at core/tcp_main.c:1541 #6 0x000055f7f42e2f3c in tcpconn_put_destroy (tcpconn=0x7f8db0fa1e90) at core/tcp_main.c:3285 #7 0x000055f7f42e8759 in handle_tcp_child (tcp_c=0x7f8e27d88078, fd_i=-1) at core/tcp_main.c:3722 #8 0x000055f7f42f506f in handle_io (fm=0x7f8e27db1578, ev=1, idx=-1) at core/tcp_main.c:4539 #9 0x000055f7f42b6e49 in io_wait_loop_epoll (h=0x55f7f45ae240 <io_h>, t=5, repeat=0) at core/io_wait.h:1070 #10 0x000055f7f42f8bc3 in tcp_main_loop () at core/tcp_main.c:4824 #11 0x000055f7f4057fc2 in main_loop () at main.c:1856 #12 0x000055f7f4062670 in main (argc=10, argv=0x7ffc6d583528) at main.c:3052 OS: Ubuntu 22.04.5 LTS \n \l # kamailio --version version: kamailio 5.5.7 (x86_64/linux) 272fdd 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: 272fdd compiled on 14:21:35 Oct 21 2024 with gcc 11.4.0 # dpkg -l |grep openssl ii openssl 3.0.2-0ubuntu1.18 amd64 Secure Sockets Layer toolkit - cryptographic utility Regards.

3 3

children monitoring
by Sergey Safarov 06 Jan '25

06 Jan '25

If TCP children are busy, Please correct me if I am wrong if the remote IP is not responding via TCP protocol, this leads to a timeout and this child is busy. If many remote endpoints are not reachable via TCP this can lead to the case when half or more children are by timeout operations. I think required to monitor how many children are busy. Is it possible to get such info using kamcmd utility, statistics module, or xprom module. Sergey

4 7

Crash on dialog_publish.
by satyaprakash ch 06 Jan '25

06 Jan '25

Hi, The kamailio version we are using is "5.7.4", We are facing a kamailio crash recently and it has generated the core-dump. Below is the backtrace for the coredump. Could you please assist with this crash. *Bachtrace:* #0 0x00007fa6b051748a in dialog_publish (state=0x7fa6b0519505 "terminated", ruri=0x7069734030783532, entity=0x7ffcdfb16510, peer=0x7ffcdfb16520, callid=0x7fa5b0527cb8, initiator=1, lifetime=10, localtag=0x0, remotetag=0x0, localtarget=0x7fa5b0527ce8, remotetarget=0x7ffcdfb16500, do_pubruri_localcheck=0, uuid=0x7fa5b0527d18) at dialog_publish.c:302 body = 0x0 uri = {s = 0x0, len = 0} publ = 0x0 size = 0 content_type = {s = 0x0, len = -1288318368} ruri_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = (unknown: 0), transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} __func__ = "dialog_publish" #1 0x00007fa6b0518f89 in dialog_publish_multi (state=0x7fa6b0519505 "terminated", ruris=0x7069734030783532, entity=0x7ffcdfb16510, peer=0x7ffcdfb16520, callid=0x7fa5b0527cb8, initiator=1, lifetime=10, localtag=0x0, remotetag=0x0, localtarget=0x7fa5b0527ce8, remotetarget=0x7ffcdfb16500, do_pubruri_localcheck=0, uuid=0x7fa5b0527d18) at dialog_publish.c:412 __func__ = "dialog_publish_multi" #2 0x00007fa6b0506944 in __dialog_sendpublish (dlg=0x7fa5b046a850, type=64, _params=0x7fa6b0437620 <params>) at pua_dialoginfo.c:402 tag = {s = 0x0, len = 0} uri = { s = 0x7fa5b0527d41 "sip:+1248XXXXXXX@10.XXX.X.Xm6ka0542f9v0lmjjc4npkamailio954739148rdb2847sip :XXXXXX.com:5060sip:521XXXx0;tgrp=51;trunk-context=198.X.XXX.XXX@10.XXX.XX.XX:5060padi-6761ad46-154e-1560padi\300\300\300\300", len = 29} identity_local = { s = 0x7fa5b0527d28 "sip:413XXXx0@10.XXX.X.XXsip :+1248XXXXXXXX@10.XXX.X.XXm6ka0542f9v0lmjjc4npkamailio954739148rdb2847sip :XXXXXX.com:5060sip:415XXXx0;tgrp=51;trunk-context=198.X.XXX.XXX@10.XXX.XX.XX:5060padi-6761ad46-154e-"..., len = 25} target = {s = 0x0, len = 0} dlginfo = 0x7fa5b0527c90 request = 0x7fa6b3361b58 __func__ = "__dialog_sendpublish" Thanks.

1 2

record_route() adding wrong sips URI when transport is missing (defaulting to UDP)
by Benoit Panizzon 06 Jan '25

06 Jan '25

Hi Gang A quick search via google did not reveal any useful hits. Kamailio 5.7.6 CPE (TLS) => SBC (UDP) => Registrar (UDP) => Core CPE uses a sips contact URI to register, no transport attribute! sips:315996608@172.29.255.11:5061 SBC uses Path functionality. Situation: Call to CPE I see the registrar adding this Record-Route: Record-Route: <sips:registrar-ip;lr;ftag=6159862378> The SBC is adding a r2=on Record-Route to indicate the change between UDP and TLS. The Route-Set received by the CPE then is: INVITE sips:315996608@172.29.255.11:5061 SIP/2.0 Record-Route: <sips:sbc-ip:5061;transport=tls;r2=on;lr;ftag=6159862378> Record-Route: <sips:sbc-ip;r2=on;lr;ftag=6159862378> Record-Route: <sips:registrar-ip;lr;ftag=6159862378> Record-Route: <sip:core-ip;lr;ftag=6159862378> 180 Ringing req 100rel is making it back as the Via header look right. But when the remote side is sending a PRACK, is has to follow the Routeset which if I am not mistaking, is not making it back because because the CORE does not know how to send this via TLS to the Registrar. Core logging: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) IMHO already the 2nd Record-Route added by the Registrar should not contain a sips: URI but a sip: URI (reflecting the transport attribute not being present, defaulting to UDP) Is this a known issue? 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 4

Guidance Required for Kamailio Load Balancer Behavior in Attended Transfer Scenarios
by Kotappa Gandudi 03 Jan '25

03 Jan '25

Dear Kamailio Team, I hope this email finds you well. I am currently working on a project where Kamailio is being utilized as a load balancer to distribute calls across multiple FreeSWITCH instances. I am using the round-robin technique for load distribution, and it is working perfectly for regular call scenarios. However, I am encountering an issue when handling attended transfer calls. Specifically, for a normal call, Kamailio assigns one FreeSWITCH instance, but during the attended transfer call, it assigns another FreeSWITCH instance. This behavior disrupts the continuity of the call handling. My requirement is for Kamailio to use the same FreeSWITCH instance for the entire call, including any attended transfers. I believe maintaining the same FreeSWITCH instance for the entire call flow would ensure consistency and avoid potential issues. Could you please guide me on how to configure Kamailio to achieve this behavior? I would greatly appreciate any suggestions, configurations, or modules that can help resolve this issue. Thank you for your time and assistance. Looking forward to your response. Best regards, Kotappa Y Gandudi

4 4

how to call loose_route() after adding route header without using msg_apply_changes?
by Benoît Panizzon 31 Dec '24

31 Dec '24

Hi List To handle CPE which can only handle a limited number of route header (PacketCable SIP specification), I need to store and restore the route header (limited topology hiding only affecting Route Header) to present the CPE only one Route or RR header. I pass the ID of the stored route-set via RR param. So when I get a message back from the CPE is: * Take ID from RR param and pull stored route set from cache. * remove_hf("Route"); * insert_hf("Route: $avp(stored-routeset)\r\n"); * msg_apply_changes(); * loose_route(); I found loose_route() only to be able to correctly determine the next hop, if I first did apply the changes. But after adding rtpengine to the mix, I got back into the known situation that calling rtpengine (on the branch route) after calling msg_apply_changes() results in duplicate SDP. Is there a way to get loose_route() to operate on freshly added header without calling msg_apply_changes? -- 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 ______________________________________________________

8 28

Async HTTP request is sent but response is incorrectly processed
by Sergio Charrua 30 Dec '24

30 Dec '24

Hi all! still struggling with Async HTTP Client ... My current configuration is very simple: #!define HTTP_API_ROUTING_ENDPOINT_MAIN " http://MY_HTTP_REST_API:18085/Endpoint" include_file "listen.cfg" loadmodule "pv" loadmodule "tm" loadmodule "http_async_client" loadmodule "xlog" loadmodule "tmx" loadmodule "sl" loadmodule "textops" loadmodule "cfgutils" loadmodule "siputils" loadmodule "rr" loadmodule "ctl" loadmodule "jsonrpcs.so" loadmodule "jansson.so" modparam("pv", "shvset", "http_async=i:8"); modparam("http_async_client", "workers", 1); modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "curl_verbose", 1) # for debug purpose #modparam("tm", "fr_timer", 750) #modparam("tm", "fr_inv_timer", 20000) #modparam("tm", "remap_503_500", 0) modparam("tm", "failure_reply_mode", 3) modparam("tm", "max_inv_lifetime",5000) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") debug=2 children=16 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " disable_sctp = yes force_rport = yes rundir="/tmp" request_route { $avp(REQUEST_URL) = HTTP_API_ROUTING_ENDPOINT_MAIN; if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); } exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } jansson_set("string","from" , $hdr(From) , "$var(req)"); jansson_set("string","to" , $hdr(To) , "$var(req)"); jansson_set("string","r-uri" , $ru , "$var(req)"); jansson_set("string","contact" , $hdr(Contact) , "$var(req)"); jansson_set("string","call-id" , $ci , "$var(req)"); xinfo("MAIN-ROUTE - Setting From=$hdr(From) To=$hdr(To) Contact=$hdr(Contact) ci=$ci \n"); if ( is_present_hf("Identity") ) jansson_set("string","identity" , $hdr(Identity) , "$var(req)"); if ( is_present_hf("P-Identity-Bypass") ) jansson_set("string","p-identity-bypass" , $hdr(P-Identity-Bypass) , "$var(req)"); if ( is_present_hf("P-Asserted-Identity") ) jansson_set("string","p-asserted-identity" , $hdr(P-Asserted-Identity) , "$var(req)"); if ( is_present_hf("P-STSH-UC") ) jansson_set("string","p-stsh-uc" , $hdr(P-STSH-UC) , "$var(req)"); jansson_set("string", "request-time" , $avp(requestTime), "$var(req)"); ## Create transaction before http_async_query; if (is_method("INVITE")){ t_newtran(); xinfo("MAIN-ROUTE - Sending request [$var(req)]\n"); route(HTTP_ASYNC_REQUEST); } exit; } route[HTTP_ASYNC_REQUEST] { xinfo("HTTP_ASYNC_REQUEST - Sending request [$var(req)]\n"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = 1500; # 1 second timeout $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; $http_req(body) = $var(req); if ( !http_async_query($avp(REQUEST_URL), "HTTP_REPLY") ) { xinfo("HTTP_ASYNC_REQUEST - http_async_query $ci FAILED. Return is [$rc]\n"); }else{ send_reply("503", "Server Error"); exit; } } route[HTTP_REPLY] { xinfo("HTTP_REPLY - Xphere Response Code $http_rs - $ci\n"); if ( $http_ok && $http_rs==200) { xinfo("HTTP_REPLY - $ci reply ok\n"); jansson_xdecode($http_rb, "json"); xinfo("HTTP_REPLY - HTTP JSON Response: $http_rb \n"); if ( $xavp(json=>sip-response-code) == 300 ){ remove_hf("Contact"); append_to_reply("Contact: $xavp(json=>contact)\r\n"); ## "touch" the ruri to add the contact header for 3xx: $ru = $ru; send_reply("300", "Multiple Choice"); exit; }else{ xinfo("HTTP_REPLY - $ci Error Replied $xavp(json=>sip-response-code) - $xavp(json=>sip-response-text) \n"); send_reply("$xavp(json=>sip-response-code)","$xavp(json=>sip-response-text)"); exit; } } else { xinfo("HTTP_REPLY - HTTP Response Code $ci - $http_rs and error $http_err \n"); if (!$http_ok){ sl_reply("503", "No Connection to HTTP"); exit; }else if ($http_rs>=500){ sl_reply("503", "HTTP Processing Error"); exit; } else{ sl_reply("503", "Cannot Connect to HTTP"); exit; } } exit; } The configuration is based on Ben Kaufman's kamailio_http_async <https://github.com/whosgonna/kamailio_http_async> example, but I have extended it a bit. Currently all the requests are being sent to the HTTP REST API. I can't find queue/buffering issues... What I am experiencing is that , using SNGREP, I get this flow: 10.240.12.209:5060 10.242.17.146:5060 ----------*---------- ----------*---------- 17:33:46.689852 x INVITE (SDP) x +0.000530 x --------------------------> x 17:33:46.690382 x 100 trying -- your call is x +1.499822 x <-------------------------- x 17:33:48.190204 x 503 No Connection to HTTP x +0.000029 x <-------------------------- x 17:33:48.190233 x 500 I'm terribly sorry, se x +0.002130 x <-------------------------- x 17:33:48.192363 x ACK x +0.485836 x --------------------------> x 17:33:48.678199 x 500 I'm terribly sorry, se x +0.004881 x <<<------------------------ x 17:33:48.683080 x ACK x +0.973295 x ------------------------>>> x 17:33:49.656375 x 500 I'm terribly sorry, se x +0.004105 x <<<------------------------ x 17:33:49.660480 x ACK x +1.995420 x ------------------------>>> x 17:33:51.655900 x 500 I'm terribly sorry, se x +0.005361 x <<<-------------------------x 17:33:51.661261 x ACK x x ------------------------>>> x I suspected issues with the HTTP REST service, but when using tcpdump, I can find the Request from Kamailio and a reply back from the REST API service, so doesn't seem to be a problem on the HTTP REST service So my questions are: - why is Kamailio sending a "SIP 500 I'm terribly sorry..." error after a 503 No Connection to HTTP is sent and an "exit" command is executed? - how can I get rid of these messages? - Why Kamailio keeps sending responses back to SBC after the ACK when it is set to just "exit"? - Why is Kamailio not processing the response from HTTP REST API when it is correctly received? Logs are also weird: - I find a lot of these warnings: "Dec 29 17:38:34 ire-nprod-sip1 /usr/local/sbin/kamailio[590864]: WARNING: http_async_client [http_async_client_mod.c:529]: ah_get_status(): an async variable was read after http error, use $http_ok to check the request's status" and "Dec 29 17:38:34 ire-nprod-sip1 /usr/local/sbin/kamailio[590864]: ERROR: http_async_client [http_multi.c:621]: check_multi_info(): handle 0x7f375f786b30 returned error 7:" (I am still not using EVAPI module, that will be my next step, after the holiday season) (I suspect, however, that the REST API is not being able to handle the load in this test environment (between 400 to 600 CAPS), but that is another matter.) In attach, TCPDUMP, if anyone wishes to analyse. Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 5

http sync / async final tests
by Alexis Fidalgo 26 Dec '24

26 Dec '24

Just to share, and thanks for your valuable help. My conclusion after tests (on my scenario) is http backend: is a golang http server with a 250ms delay to answer, and answering back a json, running in 10 replicas inside a K8 with 4 nodes. test (from kamailio perspective): receive INVITE send 100 call the http, get the result, get a key from the result and set it as the value of Contact header send 302 receive ACK (discard on sync scenario because it’s stateless, use it in async to end transaction) Kamailio host: 1vcpu kamailio 3.8.4 bookworm running inside containerd (with podman) 16GB RAM sipp: run from the K8 nodes hosts (directly from OS) (everything is LAN connected on 2 different physical servers, hypervisor is VMWare i guess 7.0.3) using http_client (sync with stateless): 8 children -M8 -m64 no more than 60 calls/sec (15cps on each sipp instance) until sipp start to retransmit the INVITES using async_http_client: 8 children -M8 -m2048 it’s been ~36 hours now at 250cps each sipp instance with 0 errors 0 retransmission and the kamailio host is userlab@sblb1:~$ vmstat 1 procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 4 0 0 10285460 364684 5444240 0 0 0 22 36 68 0 0 100 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5284 6822 27 14 58 0 0 4 0 0 10285460 364684 5444240 0 0 0 0 5306 6752 29 15 57 0 0 3 0 0 10285460 364684 5444240 0 0 0 0 5304 6823 26 16 57 0 0 2 0 0 10285460 364684 5444240 0 0 0 12 5287 6761 26 16 57 0 0 2 0 0 10285460 364684 5444240 0 0 0 0 5337 6838 27 14 58 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5101 6724 25 17 59 0 0 3 0 0 10285460 364684 5444240 0 0 0 0 5066 6650 26 14 60 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5306 6829 23 17 59 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5219 6725 23 19 58 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5274 6770 26 13 61 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5295 6756 26 18 56 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5235 6786 22 16 62 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5315 6833 21 20 59 0 0 at least for my scenario, it worths to pay the price of the memory tradeoff. again, Ben, Alex, Sergio, all, thanks for your help. regards

1 0

help on how to get ACK
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello all, moving just a bit aside of the http and async_http. After all the real useful and interesting thread on that topic what helped me, im facing a problem i cant deal with and need a hint at least. Scenario INVITE -> Kamailio on request_route ... if(is_method("INVITE")){ t_newtran(); http_async_query("http://nuc:8080", "HTTP_REPLY"); } … Kamailio -> 100 - Trying then route[HTTP_REPLY] { if ($http_ok) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); t_reply(302,"Redirect"); } else { xlog("L_INFO", "route[HTTP_REPLY]: error $http_err)\n"); } } Kamailio -> 302 Redirect ACK -> Kamailio This last ACK, how can i read it and use it to terminate the transaction? because Kamailio keeps transmitting the 302 message 3 more times until the transaction is finished by a timer 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [timer.c:642]: wait_handler(): finished transaction: 0x7f4c3fcd35a0 (p:0x7f4c3fad85d0/n:0x7f4c3fad85d0) 42(44) DEBUG: tm [h_table.c:133]: free_cell_helper(): freeing transaction 0x7f4c3fcd35a0 from timer.c:651 in request_route i have if(is_method("ACK")){ if(!t_check_trans()){ t_release(); } exit; }

4 26

append_branch
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello all, how can avoid to append_branch do this call to call SIP/2.0 302 Redirect Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-65665-5-0 From: sipp <sip:sipp@192.168.86.250:5060>;tag=65665SIPpTag005 To: 123456 <sip:123456@192.168.86.128:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-deb8da2e Call-ID: 5-65665(a)192.168.86.250 CSeq: 1 INVITE Contact: <sip:4093823670@otherdomain.com>, <sip:8888576523@otherdomain.com>, <sip:9719533197@otherdomain.com>, <sip:5302579529@otherdomain.com>, <sip:4972010218@otherdomain.com> Server: kamailio (5.8.4 (x86_64/linux)) Content-Length: 0 What i mean with “avoid”, call to call, contact should have only 1 branch, no a stack of previous. thanks

2 1

Transactions help
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello, after the discussion regarding http and http_async, I started some labs to make a comparison and see what’s the trade off and the associated cost. Main problem (to me) is im not clear on how transactions behave. Im running some fast tests now before I start to deploy the final scenarios and 1. A new transaction is created before the http_async call (as the module example shows for http_async_query function 2. The http_async_query call is executed, receives a response and the HTTP_REPLY route for http_async_query is executed. 3. As we need to redirect the call, im adding Contact information with append_to_reply and then calling t_reply(302,”Redirect) Up to here INVITE is received, a 100 is answered and then the 302 with the expected Contact information (collected and built from the http_async_query response), problem is that the 302 is retransmitted 3 more times. So, im not even sure that im doing the correct actions to process the call this way. Is there a document/book/reference I can read to understand transactions better? By now I can not handle the call to be processed correctly (without the retransmissions) and finish the transaction the correct way. Any help, link, guide, will be appreciated. Regards.

6 17

Kamailio not receiving packets on high CPS
by Sergio Charrua 23 Dec '24

23 Dec '24

Hello! for this ST/SH project, we are using Kamailio 5.8.4 in stateless mode, making HTTP requests to a REST API (Java Spring) that will reply with a JSON object, and then Kamailio replaces the Contact header and sends a SIP 300 Multiple Choice reply to the SBC that sent the initial INVITE. Kamailio is running on a VM with 4vCPUs and 4GB RAM and 2 NICs. On NIC ens224 there is a Virtual IP managed by Keepalived. Kamailio is listening on port 5060 on ens224 and set with 32 children process (tried with 8, 16, 24 and also 64, but with this value it was a lot worst). We are doing load performance tests on our PreProd environment. With the SBC we are sending 450 to 600 CAPS to Kamailio, and what we noticed is that above 450 CAPS, and after less than 1 minute, Kamailio only replies SIP 100 to some INVITEs. We also could figure out that Kamailio is not receiving all the INVITES, despite having proved that ALL invites are sent to the server. It seems to me that, for some reason, the OS is somehow not able to deliver *all* SIP packets to Kamailio, because: - sngrep does capture all the SIP packets and shows the flows - though the flow shows SIP invite coming from SBC to Kamailio Server, there is no SIP 100 replied back to SBC, even though this is the 1st thing kamailio does, so, for me, Kamailio did not received the SIP Invite - picking a call with reinvites, which after some seconds, SBC cancels, if I copy the Call-ID value and grep on kamailio's logs, I just do not find any content! I have read some "network tuning" articles, including Linux Tune Network Stack (Buffers Size) To Increase Networking Performance - nixCraft <https://www.cyberciti.biz/faq/linux-tcp-tuning/> and Tuning Kamailio for high throughput and performance | Evariste Systems Blog <https://blog.evaristesys.com/2016/02/15/tuning-kamailio-for-high-throughput…> and when running the following command I get: $ ss -4 -n -l | grep 5060 udp UNCONN 0 0 10.242.17.125:5060 0.0.0.0:* udp UNCONN 324224 0 10.242.17.146:5060 0.0.0.0:* According to the article, the 3rd column should be as near 0 as possible, ideally 0. While I am waiting for some sysadmin with root permissions to be available and modify some of the network parameters on this VM, I wonder if anyone has some tips to share on how to solve this behaviour. I'm not sure, either, what parameters to change or if they should be changed! The goal is to have a minimum 750 CAPS. Lastly, some parts of the code: Kamailio.cfg: > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=1 > log_stderror=no > rundir="/tmp" > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > # configure the prefix for all log messages > log_prefix_mode = 1 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=32 # ------------- ALREADY TRIED WITH 8,16,24... no results..... > /* uncomment the next line to disable TCP (default on) */ > disable_tcp=no > /* uncomment the next line to disable the auto discovery of local aliases > * based on reverse DNS on IPs (default on) */ > # auto_aliases=no > /* add local domain aliases */ > # alias="sip.mydomain.com" > ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ ####### Modules Section ######## /* set paths to location of modules */ loadmodule "db_mysql.so" loadmodule "db_cluster.so" loadmodule "http_client.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "jansson.so" loadmodule "snmpstats.so" loadmodule "file_out.so" loadmodule "ctl.so" loadmodule "permissions.so" loadmodule "xhttp.so" loadmodule "xhttp_rpc.so" ####### Other Args and Env Vars ######### /* listen addresses */ include_file "listen.cfg" include_file "db_conn.cfg" # ----------------- setting module-specific parameters -------------- # --- DB Cluster params --- # minimum requirement is to have DBURL1 defined #!ifdef DBURL1 modparam("db_cluster", "connection" , DBURL1) #!endif #!ifdef DBURL2 modparam("db_cluster", "connection" , DBURL2) #!endif #!ifdef DBURL3 modparam("db_cluster", "connection" , DBURL3) #!endif #!ifdef DBURL4 modparam("db_cluster", "connection" , DBURL4) #!endif #!ifdef DBURL5 modparam("db_cluster", "connection" , DBURL5) #!endif modparam("db_cluster", "cluster", DBCLUSTER) # --- Permissions params --- modparam("permissions", "db_url", "cluster://k1") modparam("permissions", "db_mode", 1) #modparam("permissions", "reload_delta", 30) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") modparam("jsonrpcs", "transport", 0) # ----- ctl params ----- /* set the path to RPC unix socket control file */ modparam("ctl", "binrpc", "unix:/tmp/kamailio_ctl") # ----- http_async_client params ----- modparam("http_client", "query_result", 0) modparam("http_client", "keep_connections", 0) modparam("http_client", "connection_timeout",2) modparam("http_client", "timeout_mode",2) modparam("http_client", "config_file", "/usr/local/etc/kamailio/http_client.cfg") # ---- SNMP Stats params ---- modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "sipEntityType", "redirectServer") modparam("snmpstats", "sipEntityType", "other") modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "MsgQueueMinorThreshold", 20) modparam("snmpstats", "MsgQueueMajorThreshold", 100) modparam("snmpstats", "dlg_minor_threshold", 20) modparam("snmpstats", "dlg_major_threshold", 100) modparam("snmpstats", "snmpCommunity", "kamailio") # ---- File_Out params ---- modparam("file_out", "base_folder", "/data/sabire002/log/kamailio/") modparam("file_out", "file", "name=stsh_requests;interval=1440;extension=.log") modparam("file_out", "file", "name=cdr;interval=1440;extension=.log") modparam("file_out", "file", "name=http;interval=1440;extension=.log") ####### Routing Logic ######## include_file "includes/reqinit.cfg" include_file "includes/handle_options.cfg" include_file "includes/handle_cancel.cfg" include_file "includes/handle_stir_shaken_stateless.cfg" include_file "includes/handle_http_rpc.cfg" request_route { $avp(START_TIME)=$utimef(%Y-%m-%d %H:%M:%S); $avp(GROUPID) = allow_address_group($si, $sp); if ( $avp(GROUPID) == 100 || !allow_address_group($si, $sp) ) { xlog("L_INFO", "INIT - $si:$sp is not in the allowed ACL Group ID !\n"); #sl_reply("401", "Address not allowed"); exit; }; if (is_method("ACK") ) { #&& t_check_trans() ){ exit; } if (is_method("INVITE")) { file_out("cdr","$rm|$ft|$tt|$ci|$rs|$rr|$Ts|$avp(START_TIME)|$fU|$fd|$si|$tU|$rU|$rd|$utimef(%Y-%m-%d %H:%M:%S)"); send_reply("100","Trying"); } route(HANDLE_OPTIONS); route(REQINIT); xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","=================================================== \n"); xlog("L_INFO","New SIP message $rm with call-ID $ci received $pr request $rm $ou source $si:$sp from $fu to $tu \n"); xlog("L_INFO","=================================================== \n"); route(HANDLE_STIRSHAKEN); route(HANDLE_CANCEL); if (method == "INVITE"){ route(RELAY); } } route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","RELAY - send reply \n"); file_out("cdr","RELAY|$ft|$tt|$ci|$rs|$rr|$Ts|$avp(START_TIME)|$fU|$fd|$si|$tU|$rU|$rd|$utimef(%Y-%m-%d %H:%M:%S)"); send_reply("300", "Multiple Choices"); exit; } As for the HANDLE_STIRSHAKEN route, where the main process is: route[HANDLE_STIRSHAKEN] { if (!is_method("INVITE")) { return; } $xavp(requestTime) = $utimef(%s); $var(post) = $null; // resets and makes sure the $var(post) variable is null before usage [.... get some header values ...] $var(res) = http_connect("api1", "/stsh", "application/json", "$var(post)", "$var(result)"); jansson_xdecode($var(result), "json"); if ( $xavp(json=>sip-response-code) == 300 ) { remove_hf("Contact"); append_to_reply("Contact: $xavp(json=>contact)\r\n"); } else { # add error message to reply sl_reply("$xavp(json=>sip-response-code)","$xavp(json=>sip-response-text)"); exit; } } Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

8 73

How to use @via[idx] ?
by Benoit Panizzon 23 Dec '24

23 Dec '24

Hi https://www.kamailio.org/wikidocs/cookbooks/5.7.x/pseudovariables/#selname-… https://www.kamailio.org/wikidocs/cookbooks/5.7.x/selects/#viaihost I'm trying to list all via hosts: $var(dooh) = @via[1].host; xlog("L_INFO","******** VIA HOST 1: $var(dooh) \n"); This above works as expected I get the 2nd entry. $var(i) = 0; while ($var(i) < $hflc(Via)) { $var(dooh) = @via[$var(i)].host; xlog("L_INFO","******** VIA HOST: $var(dooh) \n"); $var(i) = $var(i) + 1; } This fails with syntax error. How do I specify the index of the via header I want to extract the host from? I also attempted $sel(via[$var(i)].host) or via[$var(i)] or $sel(hfl(via[$var(i)])).host but I found no variant not throwing an error. PS: Is there maybe another nice function to figure out if we have already seen (and how many times we have) this same message? 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 5

KAMI python3 and python3s not working / reloading
by trle.fuad＠gmail.com 20 Dec '24

20 Dec '24

Hello, I've started testing/learning KEMI and decide to use python3(s). I'm using default config with kamailio-basic-kemi-python3s.py (just deleted a couple of routes for simplicity). First thing I noticed is that reloading is not working. I've changed simple log line and did kamcmd app_python3s.reload. I'm compiling from source (inside Docker container, Debian bookworm as base), and have noticed it stopped working after thees commits for python3s: 62b4ee4a0d0b62b35c8bdf67e5daf9cbe9a28499 app_python3s: refactor GIL and thread state handling c81b2b3383c5900aba83672da024c812d8e6d89d app_python3s: initial support for free-threading Python And for python3 0ffe157bc13e7759ae1cee63a584fad4ac9eb38f app_python3: refactor GIL and thread state handling Am I missing some new parameter for module or compile flag? What I've noticed it's not only that it not reload, it also does not go through complete config file. While reloading, I see marking for reload. But nothing after that: 22(774) DEBUG: ctl [../../core/io_wait.h:368]: io_watch_add(): processing io_watch_add(0x748d1dfb83c0, 11, 3, 0x5f0adde454d0) - fd_no=1 22(774) DEBUG: ctl [io_listener.c:438]: handle_new_connect(): new connection (1) on /var/run/kamailio//kamailio_ctl 22(774) DEBUG: ctl [io_listener.c:499]: handle_stream_read(): bytes read: 29 22(774) INFO: app_python3s [apy3s_kemi.c:868]: app_python3s_rpc_reload(): marking for reload Python script file: /etc/kamailio/kamailio-basic-kemi-python3s.py (0 => 1) 22(774) DEBUG: ctl [io_listener.c:520]: handle_stream_read(): bytes processed: 29 22(774) DEBUG: ctl [io_listener.c:496]: handle_stream_read(): handle_stream read: eof on /var/run/kamailio//kamailio_ctl 22(774) DEBUG: ctl [../../core/io_wait.h:599]: io_watch_del(): DBG: io_watch_del (0x748d1dfb83c0, 11, -1, 0x10) fd_no=2 called

1 0

nethelper module - need clarification on how to handle uri alias on location contacts.
by Benoît Panizzon 20 Dec '24

20 Dec '24

Hi Gang When a CPE behind nat registers, I have the alias added to the AOR contact. It has format ip~port~transport Let's say: bob@1.2.3.4:5060;alias=5.6.7.8~7777~1 On an invite to that CPE behind nat, the contact is retrieved from the location record and I assumed I could call handle_ruri_alias in the branch route to that CPE and this would sent the call to that ip port via transport in the alias. I observe that the port is not taken into account. The INVITE to bob is sent to ip 5.6.7.8 but to port 5060 instead of 7777 In the example config, handle_ruri_alias() is only used for in dialog requests (has_totag). Have I misunderstood how CPE behind NAT should be registered and that registration resolved? -- 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 ______________________________________________________

1 0

Request for Feedback on an Installation Script for Kamailio (from Source)
by BROU STEPHANE FABIEN 19 Dec '24

19 Dec '24

Dear Kamailio Community, I hope this message finds you well. I have recently developed an install.sh script aimed at simplifying the process of installing Kamailio directly from source. My goal is to streamline the installation process while adhering to best practices for Kamailio deployment. I would greatly appreciate your feedback, suggestions, or any corrections to help improve the script’s robustness and ensure it aligns with the community’s standards. You can find the script in my GitHub repository: https://github.com/AL-CALL-LLC/KamailioBuilder.git Feel free to test it, share your thoughts, or open issues on GitHub for any recommendations or areas that need refinement. Thank you in advance for your time and valuable expertise. I am more than happy to provide further details or engage in discussions if needed. Best regards, Fabien Brou Developer

1 0

[print_dset(): no r-uri or branches] when working with async worker
by elhar.mohamed＠gmail.com 18 Dec '24

18 Dec '24

Hello, I'm getting this warning WARNING: <core> [core/dset.c:690]: print_dset(): no r-uri or branches when working with async worker. here is below the code used: request_route { if (is_method("INVITE")) { append_to_reply("Contact: <sip:+1234;myparam=test@$si:$sp>\r\n"); send_reply("302", "Moved Temporarily"); exit; } } Can you please help?

2 2

Multiple Branches and 3xx Status Codes
by Christian Berger 17 Dec '24

17 Dec '24

Hi, we have the following situation: Multiple UAs are subscribed to the proxy and get INVITEd via parallel forking. Now one of the UAs sends us a 302 status, we would like to handle this like a failure. So we'd like to CANCEL all the other branches, and forward the 302 status to the original INVITE. We have tried t_reply("302", "$T(reply_reason)") in the response, which almost solves this, however we cannot pass the required headers form the 302 response in it. BTW, could there have been a change in the default behavior of Kamailio? We think it used to "just work" in the past. Thank you very much Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

3 3

Development for v6.0.x is frozen
by Daniel-Constantin Mierla 17 Dec '24

17 Dec '24

Hello, the formal notification that the development for the next major version 6.0.0 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 6.0 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 6.0. 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 6.0. Once the branch 6.0 is created, new features can be pushed again to master branch as usual. From that moment, the v6.0.0 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 6.0 is going to be built at: * https://www.kamailio.org/wikidocs/features/new-in-6.0.x/ Upgrade guidelines will be collected at: * https://www.kamailio.org/wikidocs/install/upgrade/5.8.x-to-6.0.0/ 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 v5.8.x to 6.0.x. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Freezing development for 6.0.x series
by Daniel-Constantin Mierla 16 Dec '24

16 Dec '24

Hello, I propose to aim freezing the development for 6.0.x series at the end of the 16th of December 2024 (Monday). New features that one wants to get in this release series have to be pushed to git repository or pull requests made for them. Afterwards usually follows a 4-6 weeks of testing till the first release 6.0.0. Unfreezing will happen earlier, after the first weeks of testing when the 6.0 branch will be created. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

ims_diameter_server: $diameter_response PV is null in the event_route[diameter:response]
by Grotti, Daniel 15 Dec '24

15 Dec '24

Hi list, I'm using kamailio 5.8 and playing with the ims_diameter_server module, by sending an async diameter request and trying to catch the reponse. Following the documentation, I'm sending the request with: route[SEND_ACR] { route(PREPARE_ACR); diameter_request_async("3", "271", "$var(json_acr)"); } route[PREPARE_ACR] { # here I build the AVP list in JSON $var(json_acr) = "[{"avpCode":277,"vendorId":0,"Flags":64,"int32":0},{...},{...}]"; } event_route[diameter:response] { xlog("Reply to diameter request is: '$diameter_response'\n"); } The ACR request is successfully sent and the the server is replying with ACA. The list of AVP are properly parsed, but when I enter the event_route to print the response, I get the $diameter_response = 'null': 2024-12-11T10:52:39.192951872+00:00 stderr F 11(19) DEBUG: cdp [worker.c:374]: worker_process(): worker_process(): [2] got task Q(2/2) 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP((nil) < 0x7ff827633810 >0x7ff8276338c0);code=263,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=1;VendorID=0;DataLen=52; 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827633810 < 0x7ff8276338c0 >0x7ff827633970);code=268,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=3;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff8276338c0 < 0x7ff827633970 >0x7ff827633a20);code=264,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=1;VendorID=0;DataLen=37; 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827633970 < 0x7ff827633a20 >0x7ff8276340b0);code=296,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=1;VendorID=0;DataLen=33; 2024-12-11T10:52:39.193124627+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827633a20 < 0x7ff8276340b0 >0x7ff827634160);code=259,flags=40; 2024-12-11T10:52:39.193124627+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193286132+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff8276340b0 < 0x7ff827634160 >0x7ff827634210);code=480,flags=40; 2024-12-11T10:52:39.193286132+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193286132+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827634160 < 0x7ff827634210 >0x7ff8276342c0);code=485,flags=40; 2024-12-11T10:52:39.193286132+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193286132+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827634210 < 0x7ff8276342c0 >(nil));code=85,flags=40; 2024-12-11T10:52:39.193286132+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193612877+00:00 stderr F 11(19) INFO: <script>: [event_route]: Reply to diameter request is: '<null>' Is there something I miss to set/configure? Thanks, Daniel Daniel Grotti Development Team Leader, Private 5G daniel.grotti(a)hpe.com<mailto:daniel.grotti@hpe.com> Hewlett Packard Enterprise HPE.com [cid:803b85f7-01fe-4c9c-a069-74d14a9ea122]

3 9

How to remove rr param on a reply?
by Benoît Panizzon 13 Dec '24

13 Dec '24

Hi Is there a way to remove an param from the top most rr header? I add a rr param towards the B CPE to store some information which is sent back on the rr header of the reply from B. So far this is what is intended and I also get this param in subsequent route headers from messages created by B and as the route header are remove towards A, this works fine. Unfortunately, in a case of a reply, the parameter is passed to A which is then sending back the param in the route header which would make it somehow complicated to figure out which side sent that parameter. I'm only interested if sent from B. So I wonder, is there a way to delete a rr param on replies to make sure I don't get it back in the ACK to the reply? -- 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 4

slot hongkong terbaik
by Womkevintanuwijaya＠gmail.com 13 Dec '24

13 Dec '24

<a href="https://slothongkong.pro/">slot hongkong</a> The best information from the world's largest number one blog!

1 0

registrar / usrloc / tm => 'load balanced' serial forking?
by Benoit Panizzon 13 Dec '24

13 Dec '24

Hi List With the dispatcher module, you can use different ways of load balanced and weighted serial or parallel forking. Is there a corresponding way to achieve load balanced serial forking to a destination set loaded from registered locations? Use case: Customer with a high availability set-up has two redundant PBX registered to the same AoR. Calls can not be sent in parallel as this would generate two concurrent calls to the end device but call should be distributed evenly between the two contacts registered with that AoR. 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 8

HTTP_CLIENT timeout
by Sergio Charrua 13 Dec '24

13 Dec '24

Hello all! For a STSH project, kamailio is making HTTP requests to a middleware (REST API) using HTTP_CLIENT module. I have set the parameter "timeout" to 2 second, but this increases PDD to too high levels, so I have set it to 1 second. Problem is that 1sec is also too high! The parameter only accepts integer values, so no way to set this to 0.5sec. I thought of having a way for kamailio to be aware that HTTP node #1 is down, so next request it will use HTTP node #2 instead. And once in a while, say, every 60 seconds, kamailio would check status of HTTP node #1 and, if OK, then it will switch back to HTTP node #1 and failover to node #2 when required. The HTTP_CLIENT, as far as I know, doesn't do that. So the question is: what would be the best approach to have this feature on Kamailio? i thought about using HTable and have a flag with value of the HTTP connection to use by default. At worst, only the 1st failover request would take 1 or 2 seconds to timeout....then the subsequent HTTP requests would use node #2. Any ideas? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

4 5

How to inspect/dump a stacked xavp? / How to stack two xavp on each other?
by Benoit Panizzon 10 Dec '24

10 Dec '24

Hi List I'm trying to inspect the content of the $xavp(tm_contacts) to figure out how t_load_contacts exactly behaves when called multiple times. $var(k) = 0; while ($xavp(tm_contacts[$var(k)]) != $null) { $var(out) = "nothing"; xavp_params_implode("tm_contacts[$var(k)]", "$var(out)"); xlog("L_INFO", "$cfg(route): tm_contacts idx: $var(k) => $var(out)\n"); $var(k) = $var(k) + 1; } My stack contains multiple entries. Unfortunately I struggle to access the lower entries. xavp_params_implode("tm_contacts[$var(k)]", "$var(out)"); does nothing when $var(k) > 0 What is the best way to dump a complete xavp stack? I suspect t_load_contacts does flush the complete stack. Is there a way to stack an xavp on top of another one? Something like: xavp_rm("contact_stack"); # Initialize Stack lookup("location", "sip:alice@example.com"); t_load_contacts(0); $xavp(contact_stack) = $xavp(tm_contacts); lookup("location", "sip:bob@example.com"); t_load_contacts(0); $xavp(contact_stack) = $xavp(tm_contacts); # contact_stack now contains alice + bob lookup("location", "sip:carlie@example.com"); t_load_contacts(0); # $xavp(tm_contacts) now contains only charlie. $xavp(tm_contacts) = $xavp(contact_stack); # $xavp(tm_contacts) now contains only alice, bob + charlie. t_next_contacts(); # Push contacts to destination list creating branches. t_relay(); # Invite all contacts 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 ______________________________________________________

6 10

segmentation fault when assigning xavp to another xavp (trying to stack them)
by Benoit Panizzon 10 Dec '24

10 Dec '24

Hi version: kamailio 5.7.6 (x86_64/linux) on Ubuntu from the: http://deb.kamailio.org/kamailio57 repository AOR alice has 4 contacts with q=0.5 AOR bob has 1 contact with q=0.5 I can reproduce a crash with this code snipplet: xavp_rm("contact_stack"); # Make sure XAVP does not exist. lookup("location", "sip:alice@example.com"); t_load_contacts(0); # after this, $xavp(tm_contacts) has four elements $xavp(contact_stack) = $xavp(tm_contacts); # Store the four elements on my stack # Verify, $xavp(contact_stack) now has four elements, copy successfull. lookup("location", "sip:bob@example.com"); # after this, $xavp(tm_contacts) has one element $xavp(tm_contacts) = $xavp(contact_stack); # Append the four previously saved contact with the one in $xavp(tm_contacts) if (!t_next_contacts()) { xlog("L_INFO", "$cfg(route): ERROR NO CONTACTS LOADED \n"); sl_send_reply("500", "Server Internal Error - Cannot get contacts"); } # Load contacts into destination set to parallel fork to five contacts. => segmentation fault! Shall I open an issue on github or am I performing a forbidden action? I'm doing some more test to find out where exactly the crash occurs. I'm not sure if it happens when adding to tm_contacts of when calling t_next_contacts. 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 ______________________________________________________

2 1

uneven load distribution after gw goes down?
by David Villasmil 09 Dec '24

09 Dec '24

Hello guys, I'm seeing the dispatcher module used with ds_select_dst($avp(dsdst), "0") is distributing the calls to a failing gw, to the next available gw instead of distributing it. When a gw fails and has been marked as inactive, shouldn't it be taken out of the list and the list recomputed? is this not the case? Thanks all Regards, David Villasmil email: david.villasmil.work(a)gmail.com

2 2

Updated: Online devel meeting on matrix - 15:00 UTC, Dec 9, 2024
by Daniel-Constantin Mierla 09 Dec '24

09 Dec '24

Hello, we should consider an online devel meeting sometime soon to summarize what was done at (and still needs to be done after) devel meeting in Dusseldorf and plan a bit the targets for next major release 6.0. If considered useful, I propose Dec 9, 2024 (Monday) at 15:00UTC (16:00 Berlin/Paris/Madrid/Rome), but we can also look for other dates as well. Topics to be discussed can be added at: - https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings… Pull requests can be made by users without git access. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

WS and rtpengine_manage
by Knserbrave 09 Dec '24

09 Dec '24

Hi Kamailio Community, When testing rtpengine_manage and websockets, I noticed that if the rtpengine table in the database is not used, a new websocket is created for every call for a given dialog. That is, when an INVITE is received, an offer is sent to rtpengine. When the 200 OK or 18[3|0] is received, an answer is sent. However, the WS identifier often changes and as a result the request is not sent over the same websocket. This is problematic particularly when using FQDNs in the Kamailio configuration. When a load balancer is used between Kamailio and rtpengine, the answer may arrive at a different rtpengine in the cluster. After tracing through the source, I could not find a configuration parameter or set of triggers that I could manipulate to fix this. It leads me to believe it is only possible to reuse the websocket when the rtpengine table is used. Does anybody have more experience than I in this area that may be able to point me in the right direction, making load balancers for rtpengine_manage calls possible?

3 2

Potential Bug in Kamailio: Base64 Decode Transformation Not Working in Lua but Works in kamailio.cfg
by Dirk Brouwers | CM.com 28 Nov '24

28 Nov '24

Dear Kamailio Developers, I am writing to report a potential bug in Kamailio's handling of base64 decoding when using the Lua scripting interface (app_lua). It appears that base64 decoding behaves inconsistently between Lua scripts and kamailio.cfg. When using `$(var(encoded_json){s.decode.base64t})` in kamailio.cfg, where $var(encoded_json) contains the encoded base64 string: `eyJmb28iOiJiYXIifQ` it decodes to the correct `{"foo":"bar"}`. But when using the same transformation in Lua `KSR.xlog.xinfo("Decoded in Lua: ".. KSR.pv.get("$(var(encoded_json){s.decode.base64t})"))` it logs a corrupted encoded base64 string: `Decoded in Lua: ..]..#035z{b.?...#036..-....` Thank you in advance for looking into this issue. Please let me know if you need additional details or test cases. Kind regards, Dbrcm

3 4

"Intermittent Issue: Kamailio Presence Server Not Generating NOTIFY Packets"
by satyaprakash ch 27 Nov '24

27 Nov '24

Hi We have FreeSWITCH and Kamailio presence servers set up. FreeSWITCH sends XML RPC messages to the Kamailio presence servers for PUBLISH operations, and we receive 200 OK responses in return. These responses include the SIP-ETag and Expires headers. However, the Kamailio presence server is not consistently generating NOTIFY packets. This issue seems to occur randomly. Can you please suggest this for presence servers. Thank you in advance.

1 0

record_route() adds empty route header if enable_double_rr is 1
by Benoit Panizzon 26 Nov '24

26 Nov '24

Hi Gang How is record_route() determining if it needs to add two RR Header? Kamailio 5.8.4 I'm attempting to perform route header hiding by by rebuilding some of the topos functionality with perl. So towards the CPE I'm removing all existing RR header and want to replace them with one header containing an attribute which the CPE is then sending back and allows me to restore the Route header. But I came across an issue which an empty RR header being created with this (simplified) script: modparam("rr", "enable_double_rr", 1) perl_exec("hide_route"); # Stores Route-Set and sets $avp(rrattr) as id remove_hf("Record-Route"); msg_apply_changes(); loose_route(); # Required for record_route()? add_rr_param(";rr=$avp(rrattr)"); # Add Param to identify saved route set record_route(); I end up with: Record-Route: <sip:;r2=on;lr;rr=id007> Record-Route: <sip:157.161.x.x;r2=on;lr;rr=id007> The first header missing the IP address, the message being dropped by the CPE There is no transport change. If I set enable_double_rr to 0 only the 2nd RR header is created and this works as expected but I am not sure if some situations would need double RR. What causes the double RR to be created in my case? 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 ______________________________________________________

2 1

app_perl add/remove HF?
by Benoît Panizzon 22 Nov '24

22 Nov '24

Hi List I'm looking into the perl module. Am I right, that I can not add or remove HF from an app_perl subroutine? I would need to pass an AVP from perl to tell Kamailio what to do? Goal is an attempt to re-build part of the topos functionality in perl but only handling RR and Route Header only on one leg, not on both. -- 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 ______________________________________________________

1 0

tcp_accept_haproxy option
by Wolter Eldering 22 Nov '24

22 Nov '24

Hi, I noticed that if I enable tcp_accept_haproxy kamailio still can accept connections without the PROXY-protocol header. The documentation on the wiki has the following note: "Please note that enabling this option will reject any inbound TCP connection that does not conform to the PROXY-protocol spec." https://www.kamailio.org/wiki/cookbooks/devel/core#tcp_accept_haproxy This could be true up until a specific version of kamalio but from 5.7.4+ this limitation is seems to be removed. Enabling tcp_accept_haproxy is breaking the outbound/rr module because it con’t find the socket based on the address and port from the flow token. The flow tokens created by a REGISTER on a connection without a PROXY header can be used. Flow tokens created from a connection with PROXY-protocol header will fail with "cannot find socket from flow-token” once loose_route is invoked. loose.c:561 /* First, force the local socket */ si = find_si(&rcv->dst_ip, rcv->dst_port, rcv->proto); if(si) set_force_socket(_m, si); else { LM_INFO("cannot find socket from flow-token\n"); return -1; } I’m not sure if the address and/or port values stored in the flow token are incorrect or the problem lies in the matiching in find_is(…). Wolter

1 0

RTC DevRoom at FOSDEM 2025
by Daniel-Constantin Mierla 21 Nov '24

21 Nov '24

Hello, FOSDEM'25 will host again a RTC DevRoom during the afternoon of February 1, 2025 (Saturday) -- more details and the call for presentation can be read at: - https://lists.fosdem.org/pipermail/fosdem/2024q4/003584.html I am not sure if I can participate (chances are more towards no, than to yes), but if anyone considers to go to the event, it will be good to submit a proposal to cover a bit Kamailio as well (it doesn't have to be entirely about Kamailio). Note that FOSDEM imposes a strict deadline this time for submissions (Dec 1, 2024), thus is no much time left. Should anyone in these groups plans to submit a proposal, let us know, so the others have an idea about it. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

TM-Module, ignoring
by Christian Berger 19 Nov '24

19 Nov '24

Hi, we would like to replicate a SIP-message ("REGISTER" in our case) to a set of nodes for further asynchronous processing. The message is already processed on the local Kamailio. We thought about using the tm-module in conjunction with the dispatcher module so we can reliably forward the message. This seems to work, however we get a new problem: Kamailio forwards the response it gets from those nodes. When we drop the response in the reply_route, it gets dropped before TM can see it. When we drop the response in the on_reply_route, it still will be forwarded. (adding or removing a header here works, however) $du is set to $null. Is there any clean way to suppress that response being forwarded back to the original requester? Best regards Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

2 2

Not able to use SRTP - from client to proxy
by Sajid Khan 18 Nov '24

18 Nov '24

Im using the module RTPengine. im having issues when ever i try to do a call using SRTP im using microsip i have checked the SDP everything ok. But when ever i place the call it gets rejected. if i disable SRTP then the calls work if (!rtpengine_offer("replace-origin replace-session-connection ICE=force RTP/SAVP RTP/AVP")) { I only need to encrypt client -> proxy from proxy -> asterisk should be plain rtp i hope someone can help me out thank you. The log in rtpengine shows this [1731055305.417896] ERR: [2a721ebf2c73414ab8e739d7bde912f9//1 port 14124]: [srtp] SRTP output wanted, but no crypto suite was negotiated

4 4

Kamailio 5.5.7 crash
by Mitesh Thakkar 18 Nov '24

18 Nov '24

Hi, I have recently migrated from Kamailio 5.2.5 to Kamailio 5.5.7. After a migration, the Kamailio crashed. Also, there were a lot of errors and warnings. Can someone help me to understand why this happened and what could be the solution? Logs (upside down): 18(397) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 2(381) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 30(409) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 8(387) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) OS: Ubuntu 22.04.5 LTS \n \l # kamailio --version version: kamailio 5.5.7 (x86_64/linux) 272fdd 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: 272fdd compiled on 14:21:35 Oct 21 2024 with gcc 11.4.0 # dpkg -l |grep openssl ii openssl 3.0.2-0ubuntu1.18 amd64 Secure Sockets Layer toolkit - cryptographic utility Regards, Miteshkumar Thakkar

2 1

Pua PUBLISH method over xmlrpc/jsonrpc
by dries＠degendt.com 18 Nov '24

18 Nov '24

Hi all I've installed a (pretty old) presence_dfks module that allows setting the presence using the following command: kamctl fifo pua_publish sip:1000@10.10.99.254 3600 as-feature-event application/x-as-feature-event+xml . . . "<?xml version='1.0' encoding='ISO-8859-1'?><ForwardingEvent><device><notKnown/></device><forwardingType>forwardImmediate</forwardingType><forwardStatus>true</forwardStatus><forwardTo>1234</forwardTo></ForwardingEvent>" The "pua_mi" module was however removed in Kamailio and jsonrpcs/xmlrpcs are supposed to be an alternative. I've tried both the following calls, but neither does recognize the pua_publish/pua.publish as a valid method. Attempt with jsonrpc: curl -H "Content-Type: application/json" -X POST -d '{"jsonrpc": "2.0", "method": "pua.publish", "params": [""], "id":1}' https://sbctest.tel.redacted.xx:5061/RPC/ ERROR: jsonrpcs [jsonrpcs_mod.c:1422]: ki_jsonrpcs_dispatch(): method callback not found [pua.publish] Attempt with xmlrpc: curl -H "Content-Type: text/xml" -X POST -d '<?xml version="1.0" ?><methodCall><methodName>pua_publish</methodName><params><param><value><string>sip:jh@sbctest.tel.redacted.xx</string></value></param><param><value><string>7776000</string></value></param><param><value><string>as-feature-event</string></value></param><param><value><string>application/as-feature-event</string></value></param><param><value><string>.</string></value></param><param><value><string>a.1481534683.13958.6.7</string></value></param><param><value><string>sip:127.0.0.1:5080;transport=tcp</string></value></param><param><value><string>P-Flags: 0</string></value></param><param><value><string>Messages-Waiting: yesMessage-Account: sip:jh@sbctest.tel.redacted.xxVoice-Message: 2/0 (0/0)</string></value></param></params></methodCall>' https://sbctest.tel.redacted.xx:5061/RPC/ <?xml version="1.0"?> <methodResponse> <fault> <value> <struct> <member> <name>faultCode</name> <value><int>500</int></value> </member> <member> <name>faultString</name> <value><string>Method Not Found</string></value> </member> </struct> </value> </fault> Anyone who had some luck in this matter? Best regards, Dries

3 6

SIP NOTIFY over TLS fails
by dries＠degendt.com 15 Nov '24

15 Nov '24

Hi all I've installed a custom presence_dkfs module (https://github.com/tombeard/presence_dfks) in an effort to make our Grandstream devices (connected over TLS) display a server set callforward using Broadsoft's Device Feature Key Sync. The module was installed succesfully, but it appears that the NOTIFY upon SUBSCRIBE cannot be sent over TLS. This part seems to use the default presence module however so I'm taking my chances here to ask for some advice. I have already disabled verify_certificate in tls.cfg. The "dst addr: 193.19x.x.x:0" does seem to have an incorrect port as I should expect 5061? INFO: {1 600000 SUBSCRIBE 319937814-5062-3(a)BHC.DA.GB.CD} presence [notify.c:1744]: send_notify_request(): NOTIFY sip:544460@sbctest.tel.redacted.xx via sip:544460@172.30.61.23:5062;transport=tls on behalf of sip:544460@sbctest.tel.redacted.xx for event as-feature-event : 319937814-5062-3(a)BHC.DA.GB.CD ERROR: tls [tls_server.c:1312]: tls_h_read_f(): protocol level error ERROR: tls [tls_util.h:50]: tls_err_ret(): TLS connect:error:14094410:SSL routines:ssl3_read_bytes:sslv3 alert handshake failure (sni: unknown) ERROR: tls [tls_server.c:1316]: tls_h_read_f(): src addr: 172.30.61.23:5062 ERROR: tls [tls_server.c:1319]: tls_h_read_f(): dst addr: 193.19x.x.x:0 ERROR: <core> [core/tcp_read.c:1526]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6d55b4b360 r: 0x7f6d55b4b488 (-1) Your advice is most appreciated! Cheers, Dries

4 7

dlq_req_within() usage for 3pcc-style reinvite media setup
by Alex Balashov 14 Nov '24

14 Nov '24

Hi, I am trying to use dlg_req_within() for a 3pcc-style call setup, to set up media via a reinvite: 1. Store original SDP offer from caller in $dlg_var()s; 2. Later, after e2e ACK is processed for initial INVITE transaction, send reinvite to callee using dlg_req_within(). I call rtpengine_offer() and feed it the original SDP offer (via read_sdp_pv modparam), and take the RTPEngine-transformed SDP and feed that to dlg_req_within(): dlg_req_within("callee", "INVITE", "application/sdp", "$var(sdp_from_rtpengine)"); 3. I have found that event_route[tm:local-response] does not allow me to capture the 200 OK / SDP answer to this reinvite; it is internally absorbed. However, it is possible to arm an onreply_route in event_route[tm:local-request], to receive the 200 OK: event_route[tm:local-request] { if(method == "INVITE" && has_body("application/sdp")) t_on_reply("REINVITE_REPLY"); } 4. My intent is for this reinvite reply handler to set off a reinvite to the caller side: onreply_route[REINVITE_REPLY] { if(method == "INVITE" && has_body("application/sdp") && t_check_status("200")) { ... $var(rtpengine_use_this_sdp) = $rb; rtpengine_answer("..."); dlg_req_within("caller", "INVITE", "application/sdp", "$var(sdp_from_rtpengine)"); } This ensures proper relay symmetry. Otherwise, the caller will continue to send RTP to the previous upstream endpoint of the call, prior to any reinvite. However, dlg_req_within() doesn't work in this later context, even though the documentation says it can be used from ANY_ROUTE. Kamailio doesn't complain, there is no error. It just doesn't initiate a reinvite to the caller. I considered the possibility that this may be because the calling scope is that of a pending reinvite transaction to the callee, but there is no obvious way to defer that into the future. If I send it to an async task worker, the transaction scope required for dlg_req_within() to know which dialog it's operating on will be lost. Any ideas appreciated, and thank you in advance! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 2

KEMI / Python Issues
by Daniel W. Graham 14 Nov '24

14 Nov '24

Experimenting with KEMI for the first time, I ran into a few issues and hoping someone has feedback. kamailio 5.8.3 In the following python example, 'if ksr.is_method("ACK"):' should never be reached. Below is debug. if not ksr.siputils.has_totag(): ksr.info("has to tag") sys.exit() if ksr.is_method("ACK"): if ksr.tm.t_check_trans(): ksr.relay() sys.exit) else: sys.exit() ksr.sl.send_reply(404, "Not here") sys,exit() ..... DEBUG: app_python3s [apy3s_kemi.c:365]: sr_apy_kemi_exec_func_ex(): execution of method: siputils.has_totag DEBUG: siputils [checks.c:122]: has_totag(): no totag DEBUG: app_python3s [apy3s_kemi.c:368]: sr_apy_kemi_exec_func_ex(): execution of method: is_method DEBUG: app_python3s [apy3s_kemi.c:389]: sr_apy_kemi_exec_func_ex(): number of arguments: 1 DEBUG: app_python3s [apy3s_kemi.c:365]: sr_apy_kemi_exec_func_ex(): execution of method: sl.send_reply DEBUG: app_python3s [apy3s_kemi.c:389]: sr_apy_kemi_exec_func_ex(): number of arguments: 2 DEBUG: sl [sl.c:306]: send_reply(): reply in stateless mode (sl) ..... Second, here are a couple errors I have seen a few times but havent been able to identify the underlying issue, mainly hoping for more context on the meaning of 'execution of route type 1 with no name returned -1'. ERROR: app_python3s [apy3s_kemi.c:146]: apy3s_exec_func(): error exception occurred DEBUG: app_python3s [apy3s_kemi.c:230]: sr_kemi_config_engine_python(): execution of route type 1 with no name returned -1 Any input is appreciated. * dan

2 7

WG: [External] RE: Question regarding error message "failed to add new socket to the fd list"
by Floimair Florian 14 Nov '24

14 Nov '24

Thank you Henning! Here are the configured limits for Kamailio Limit Soft Limit Hard Limit Units Max cpu time unlimited unlimited seconds Max file size unlimited unlimited bytes Max data size unlimited unlimited bytes Max stack size 8388608 unlimited bytes Max core file size unlimited unlimited bytes Max resident set unlimited unlimited bytes Max processes 31543 31543 processes Max open files 16384 16384 files Max locked memory unlimited unlimited bytes Max address space unlimited unlimited bytes Max file locks unlimited unlimited locks Max pending signals 31543 31543 signals Max msgqueue size 819200 819200 bytes Max nice priority 0 0 Max realtime priority 0 0 Max realtime timeout unlimited unlimited us This is what I know so far: There seems to be a problem with siptrace module (going via TCP to a heplify-server on another host), but I’m not sure if this is a coincidence or the cause of the problems. I see quite a bit of these before the actual problem arises: Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322822]: ERROR: <core> [core/tcp_main.c:630]: _wbufq_add(): (591 bytes): write queue full or timeout (32498, total 43196, last write 0 s ago) Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322822]: ERROR: siptrace [../../core/forward.h:261]: msg_send_buffer(): tcp_send failed Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322822]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: ERROR: <core> [core/tcp_main.c:4023]: handle_ser_child(): received CON_ERROR for 0x7fe054f8ae10 (id 727321), refcnt 2, flags 0x3096 At some point then all things go south and the log is flooded with these messages Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd 2244, not in [0, 482) Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: ERROR: <core> [core/tcp_main.c:4677]: handle_tcpconn_ev(): io_watch_del(3) failed: for 0x7fe054f8ae10, fd 2244 And then ultimately the above combined with the initially mentioned Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: CRITICAL: <core> [core/tcp_main.c:4264]: handle_ser_child(): failed to add new socket to the fd list And occasionally also something like this Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: ERROR: <core> [core/io_wait.h:373]: io_watch_add(): trying to overwrite entry 2247 watched for 5 in the hash 0x5621dc404ee0 (fd:-601539712, type:22049, data:0x5621dc23c004) with (2247, 2, 0x7fe04f85c2a0) In total these add up to roughly 3600 lines per second (!!) in the log, so it is quickly flooded with these. Florian FLOIMAIR Software Development - Symphony Cloud Services (1568) Von: Henning Westerholt <hw(a)gilawa.com> Datum: Montag, 11. November 2024 um 19:52 An: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Floimair Florian <f.floimair(a)commend.com> Betreff: [External] RE: Question regarding error message "failed to add new socket to the fd list" CAUTION: This email originated from outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. Hello Florian, Any other error messages before that error happens, e.g. something about the memory. Otherwise, it could be indeed a ulimit issue. First step would be to check the actual configured limits for the kamailio user. Cheers, Henning From: Floimair Florian via sr-users <sr-users(a)lists.kamailio.org> Sent: Montag, 11. November 2024 17:45 To: sr-users(a)lists.kamailio.org Cc: Floimair Florian <f.floimair(a)commend.com> Subject: [SR-Users] Question regarding error message "failed to add new socket to the fd list" Hi! We have recently had issues with one of our Production Kamailios. When those happened, the log was filled with the following message: CRITICAL: <core> [core/tcp_main.c:4528]: handle_new_connect(): failed to add new socket to the fd list Now I wonder what the best approach is to prevent this. We are using TCP/TLS only and I think this might be related to the file ulimit, but I am not sure about that. Shared memory is set to 512MB Can you give me a hint on what to look for? Thank you very much! P.S.: Sorry, I accidentally replied to a previous post of a different topic before which is totally unrelated (I think I should stop working for today 😉) FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2072127332] commend.com LG Salzburg / FN 178618z

2 1

Tell topos to create 'duplicate' route header from rr header with same ip
by Benoit Panizzon 14 Nov '24

14 Nov '24

Hi Gang I'm having another go with a dedicated kamailio running topos to replace a B2BUA SBC. I came across this situation: Server Topology: Kamaiolio SBC (topos) <=> Kamailio Registrar <=> Kamailio Router (DLG + CDRS) Core IP Address 10.0.0.1 Registrar IP 10.0.0.2 Two different CPE are registered on Registrar via SBC One CPE is calling the other CPE So the path the call is taking is: CPE A => SBC => Registrar => Core => Registrar => SBC => CPE B Invite A => B goes through, no issue. 180 Ringing, required 100rel B => A requires the route to be recorded therefore contains 3 record route statements: Record Route: <sip:10.0.0.2;lr> Record Route: <sip:10.0.0.1;lr> Record Route: <sip:10.0.0.2;lr> To ensure answer will get route back to the registrar => core => registrar and then the contact pointing to the SBC with topos to restore the last hop. Ringing B => A works as expected and is answered by A with PRACK. topos would now need set Route: in this PRACK message, according to the received Record Route Header. Unfortunately I fear Topos 'removes' the routes with duplicate host ip. I only get: Route: <sip:10.0.0.2> Route: <sip:10.0.0.1> Causing the Core not to be able to route the PRACK back to the registrar. To test, which line is being dropped, I added an mark to each one and determined the 3rd Record Router is being dropped by topos, probably considered a duplicate. I found no setting of the 'do not perform record route header dupe detection'. Interestingly, when I look at the database field topos_d.b_rr all three entries are present in the correct order. Has anyone come over the same issue and found a solution? Maybe related to: https://github.com/kamailio/kamailio/issues/3778 ? 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 ______________________________________________________

1 0

Kamailio Developers Meeting, Nov 19-20, 2024, in Dusseldorf
by Daniel-Constantin Mierla 14 Nov '24

14 Nov '24

Hello, Kamailio SIP Server project is organizing another meeting of its developers and community members during November 19-20, 2024 (Tue-Wed), 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

2 4

Kamailio v5.8.4 Released
by Daniel-Constantin Mierla 12 Nov '24

12 Nov '24

Hello, Kamailio SIP Server v5.8.4 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.4. For more details about version 5.8.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/11/kamailio-v5-8-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.8.4
by Daniel-Constantin Mierla 12 Nov '24

12 Nov '24

Hello, I am considering to release Kamailio v5.8.4 (out of branch 5.8) sometime next week, most likely on Tuesday, Nov 12, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

6 10

fail2ban complains about Kamailio syslog messages
by Juha Heinanen 12 Nov '24

12 Nov '24

When Kamailio message parser produces this kind of syslog message: 024-11-11T21:16:40.757364+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\340#001> fail2ban complains: Consider setting logencoding to appropriate encoding for this jail. Continuing to process line ignoring invalid characters: b'2024-11-11T21:06:15.347318+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\xc0#001>\n': 1 Time(s) Error decoding line from '/var/log/syslog' with 'UTF-8'.: 1 Time(s) Unable to retrieve Watch object associated to <_RawEvent cookie=0 mask=0x2 name='' wd=13 >: 3 Time(s) Would it be possible to somehow ensure that Kamailio syslog messages only contain valid UTF-8 characters? -- Juha

1 0

Last line of onreply_route (drops reply)
by James Browne 11 Nov '24

11 Nov '24

Hi all I noticed, in my config, that if the last line of the reply_route is a function that does not succeed, then the SIP response is not relayed and kamailio throws an error. I've searched documentation to indicate why this might be, but I can't find anything, so this looks to me like a bug. I use 5.8.1 for testing. Here's a sample config that shows what goes wrong. #################### #!KAMAILIO listen=udp:203.0.113.57:5060 debug=1 loadmodule "textops" request_route{exit;} reply_route{remove_hf_re("^X-Custom");} #################### Here's a sample SIP message that works. When I send this to udp:203.0.113.57:5060, kamailio relays it. #################### SIP/2.0 200 Testing Via: SIP/2.0/UDP 203.0.113.57:5060;branch=z9hG4bK-d8754z Via: SIP/2.0/UDP 198.18.0.157:5060;branch=z9hG4bK-d8754z T: <sip:c@s>;tag=676fbf45 F: <sip:c@s>;tag=676fbf45 X-Custom: lkjdfl i: testing-XXX CSeq: 1 INFO l: 0 #################### If I then remove the X-Custom header field from the SIP reply and try again, the message does not get relayed and I see this in the log. 5(1356) WARNING: <core> [core/receive.c:605]: receive_msg(): error while trying onreply script If I put a log command after the remove_hf_re(), then the message gets relayed. It looks like the reply_route always fails if the last function is not successful. Is this normal or is it a bug? James

3 5

Question regarding error message "failed to add new socket to the fd list"
by Floimair Florian 11 Nov '24

11 Nov '24

Hi! We have recently had issues with one of our Production Kamailios. When those happened, the log was filled with the following message: CRITICAL: <core> [core/tcp_main.c:4528]: handle_new_connect(): failed to add new socket to the fd list Now I wonder what the best approach is to prevent this. We are using TCP/TLS only and I think this might be related to the file ulimit, but I am not sure about that. Shared memory is set to 512MB Can you give me a hint on what to look for? Thank you very much! P.S.: Sorry, I accidentally replied to a previous post of a different topic before which is totally unrelated (I think I should stop working for today 😉) FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2072127332] commend.com LG Salzburg / FN 178618z

2 1

apt-key deprecation
by tom＠tomlynn.com 11 Nov '24

11 Nov '24

When I look at the debian repositories maintained by the project, there are instructions to import the public key of the repository for apt using this command: wget -O http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - which results in: Warning: apt-key is deprecated. Manage keyring files in trusted.gpg.d instead (see apt-key(8)). gpg: no valid OpenPGP data found. Is there an alternate method that can be posted in the repository that will get better results? Thank you, TL

5 5

Unable to get body in multipart body
by Anthony Blandin 08 Nov '24

08 Nov '24

Hello, I want to get a body for content-type "application/isup" but I have the following error: 4(16) DEBUG: textops [textops.c:3394]: ki_get_body_part_helper(): found content type application/isup 4(16) ERROR: textops [textops.c:3397]: ki_get_body_part_helper(): no CRLF found after content type Here is the INVITE sent: INVITE sip:+883390000010022@[remote_ip]:[remote_port];user=phone SIP/2.0 From: <sip:+883390000010007@[local_ip];user=phone>;tag=[call_number] To: <sip:+883390000010022@[remote_ip];user=phone> Max-Forwards: 70 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Route: <sip:[local_ip]:[local_port];transport=udp;lr> Record-Route: <sip:[local_ip]:[local_port];transport=udp;lr> Call-ID: [call_id] CSeq: 1 INVITE P-Asserted-Identity: <sip:+883390000010007@[local_ip]:[local_port];user=phone> Accept: application/sdp,application/isup,multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE P-Charging-Vector: icid-value=IBCF-1-161460836711652742337;icid-generated-at=BC00.QX00.ORANGE-MSS.FR;orig-ioi=ORANGE-MSS.FR Supported: 100rel Content-Type: multipart/mixed;boundary=unique-boundary-1 MIME-Version: 1.0 Contact: <sip:[local_ip]:[local_port];transport=udp> Content-Length:[len] --unique-boundary-1 Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[local_ip_type] [local_ip] t=0 0 a=sendrecv m=audio [media_port] RTP/AVP 8 96 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=maxptime:40 --unique-boundary-1 Content-Type: application/isup;version=spirou;base=itu-t92+ Content-Disposition: signal;handling=required \x01\x11\x20\x01\x0a\x00\x02\x0c\x0a\x04\x10\x88\x33\x09\x00\x00\x01\x20\xf2\x0a\x0a\x84\x13\x88\x33\x09\x00\x00\x01\x00\x07\x1d\x03\x80\x90\xa3\x3f\x07\x83\x97\x26\x87\x41\x00\x00\x39\x04\x34\xc0\x3f\xc0\x00 --unique-boundary-1-- CRLF is well added at the end of Content-Type header, is anybody could help me to solve this issue? Regards, Anthony

1 0

[db_unixodbc] how to specify the database with unixodbc？
by 17603019528＠163.com 08 Nov '24

08 Nov '24

Hi Dear Community, I want to use db_unixodbc to connect to the dm8 database. Although I use Opensips, I found that this module db_unixodbc, Kamailio and Opensips are the same. My question is how to specify the database with kamailio？ I add "Database = sbc" into odbc.ini, But nothing happened。 ###### cat /usr/local/etc/odbc.ini [DM8] Description = ODBC DRIVER FOR DM8 Driver = DM8 ODBC DRIVER SERVER = 172.16.7.131 TCP_PORT = 5230 odbcinst -j unixODBC 2.3.12 DRIVERS............: /usr/local/etc/odbcinst.ini SYSTEM DATA SOURCES: /usr/local/etc/odbc.ini FILE DATA SOURCES..: /usr/local/etc/ODBCDataSources USER DATA SOURCES..: /root/.odbc.ini SQLULEN Size.......: 8 SQLLEN Size........: 8 SQLSETPOSIROW Size.: 8 ####### #################### modparam("usrloc", "db_url", "unixodbc://opensips:opensips#$@172.16.7.131:5230/dm8") #################### DBG:db_unixodbc:db_unixodbc_new_connection: opening connection: unixodbc://xxxx:xxxx@172.16.7.131/dm8 Nov 7 17:05:21 [145] DBG:db_unixodbc:db_unixodbc_new_connection: connection succeeded with reply <DSN=dm8;SERVER=172.16.7.131;UID=opensips;PWD=opensips#$;TCP_PORT=5230;> DBG:core:db_do_init: connection 0xfffd5bbbaf08 inserted in pool as 0xfffd5bbbb0e0 Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= select table_version from version where table_name='dialog' Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=42S02:1:-2106:Error in line: 1 ####################

2 1

Draining TCP/TLS SIP sessions from Kamailio and media sessions from RTPEngine
by Evgenii Buchnev 07 Nov '24

07 Nov '24

Hi Dear Community, Would appreciate your input into where to look further to explore my options and would be good to learn your experience with the following. There are many active-active kamailio nodes behind tcp load balancer. On the same VM with each Kamailio there is an instance of RTPEngine to manage Media/Transcoding/Etc for that Kamailio. This scales UP easily - new nodes just added behind tcp load balancer. To be able to scale down and/or update node configuration - I'd like to be able to drain sessions/calls properly and pass any active calls from current Kamailio and RTPEngine to other nodes. DMQ is not yet in place, although it's planned. What would be proper ways of migrating RTP sessions to other RTPEngines? Caveats? Also wonder how properly a tcp/tls based SIP session can be passed to another Kamailio? Thank you in advance for your input. Evgenii Buchnev

3 2

Roadmap to v6.0.0 - the next major release
by Daniel-Constantin Mierla 07 Nov '24

07 Nov '24

Hello, about 8 months since the last major release v5.8.x, therefore it is the time to plan a bit the next steps for getting v6.0.0 out. The Kamailio Developers Meeting is schedule in Dusseldorf, Germany during November 19-20, 2024, where we expect to push out a consistent number of updates. Then maybe we can wait for another 2 weeks or so to settle down/finish those updates. After that, if no other new features are planned by developers in short term, we can enter the testing phase, so the new major version can be released during the second part of January 2025 or first part of February 2025. Therefore, if you have new features under development that might require more than 1-1.5 months from now, and you want to get them in v6.0.0, let us know to plan properly. Other time lines suggestions to get to v6.0.0 are also welcome! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 1

Kamailio 5.5.7 alias not added to ACK
by Mitesh Thakkar 06 Nov '24

06 Nov '24

Hi, I was using kamailio 5.2.5. I have recently upgraded to kamailio 5.5.7. After upgrade, the alias is stop being add to the ACK of reinvite from upstream. Before upgrade, it was working fine. I’ll appreciate any help on this. Thank you. Regards, Miteshkumar Thakkar

1 0

SNMP setup
by Sergio Charrua 05 Nov '24

05 Nov '24

Hi all! I am having difficulties setting up SNMP, and mostly to figure out if what I have setup is correctly configured. This is what I have on Kamailio (5.8.2), which is set as stateless and as a redirect server (simply put, it receives INVITE from SBC, gets a route from routing services, modifies Contact header, and replies to SBC with 300 Multi Choice with new contact header): loadmodule "db_mysql.so" loadmodule "db_cluster.so" loadmodule "http_client.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "jansson.so" loadmodule "snmpstats.so" loadmodule "file_out.so" loadmodule "ctl.so" loadmodule "permissions.so" loadmodule "xhttp.so" loadmodule "xhttp_rpc.so" # ---- SNMP Stats params ---- modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "sipEntityType", "redirectServer") modparam("snmpstats", "sipEntityType", "other") modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "MsgQueueMinorThreshold", 1) modparam("snmpstats", "MsgQueueMajorThreshold", 1) modparam("snmpstats", "dlg_minor_threshold", 1) modparam("snmpstats", "dlg_major_threshold", 1) modparam("snmpstats", "snmpCommunity", "kamailio") I have set the *_minor_threshold values to bear minimum. Should I add anything else? The logs show the following messages: DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 60) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 64) returns address 0x7f03af3bcc50 frag. 0x7f03af3bcc10 (size=64) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 20) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 32) returns address 0x7f03af3bcd00 frag. 0x7f03af3bccc0 (size=32) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcc50), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(464) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bcc10 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcd00), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(471) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bccc0 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 60) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 64) returns address 0x7f03af3bcc50 frag. 0x7f03af3bcc10 (size=64) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 20) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 32) returns address 0x7f03af3bcd00 frag. 0x7f03af3bccc0 (size=32) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcc50), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(464) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bcc10 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcd00), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(471) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bccc0 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) INFO: snmpstats [utilities.c:116]: get_statistic(): failed to retrieve statistics for active_dialogs INFO: snmpstats [utilities.c:116]: get_statistic(): failed to retrieve statistics for active_dialogs The SNMP daemon is set as follows: - /etc/snmp]$ cat snmptrapd.conf # Example configuration file for snmptrapd # # No traps are handled by default, you must edit this file! # # authCommunity log,execute,net public # traphandle SNMPv2-MIB::coldStart /usr/bin/bin/my_great_script cold and on /etc/snmp/snmpd.conf: ############################################################################### # Further Information # # See the snmpd.conf manual page, and the output of "snmpd -H". agentaddress udp:161 rocommunity kamailio master agentx agentXSocket tcp:localhost:705 trap2sink 10.X.Y.Z netcool 162 When running tcpdump and listening on port 162, no traffic is captured....it seems to me that Kamailio is not sending anything to SNMP Any clue, anyone? What am I doing wrong? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 3

Function sdp_get gets extra blank line (when MIME is used)
by James Browne 05 Nov '24

05 Nov '24

The sdp_get() function from sdpops returns an extra blank line sometimes, which looks like a bug to me. When I have an SDP that's in MIME, there's an extra newline at the end of the extracted SDP. Here's a sample config. I use 5.8.1 for testing. #################### #!KAMAILIO listen=udp:203.0.113.57:5060 debug=1 loadmodule "sdpops" loadmodule "pv" loadmodule "xlog" request_route{exit;} reply_route { sdp_get("$avp(sdp)"); xlog("L_NOTICE", "[$ci] The SDP is [$avp(sdp)].\n"); } #################### Here are two sample SIP-message bodies. #################### v=0 o=PCD 1730 1730 IN IP4 198.18.114.148 s=PCD c=IN IP4 198.18.114.148 t=0 0 m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 #################### #################### --e61e7a6a-16a2-46c0-adbb-a41ba86d32a2 Content-Type: application/sdp Content-Length: 130 v=0 o=PCD 1730 1730 IN IP4 198.18.114.148 s=PCD c=IN IP4 198.18.114.148 t=0 0 m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --e61e7a6a-16a2-46c0-adbb-a41ba86d32a2-- #################### When the first body is received, the xlog correctly ends with this. m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ] with length 130 When the second body is received, the xlog incorrectly ends with this, which has an extra newline in it. m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ] with length 132 According to RFC2046, "The CRLF preceding the boundary delimiter line is conceptually attached to the boundary", but it looks like kamailio has a bug that incorrectly processes this. James

2 2

Simultaneous INVITE responses from multiple branches
by Anton Yabchinskiy 04 Nov '24

04 Nov '24

Hello, In our system we've encountered the following situation. There's a Kamailio proxy and a few REGISTER'ed devices behind the proxy. When there's an INVITE to these multiple devices, two of them simultaneously answer the call. Kamailio proxy accepts the first 200 OK from one of the devices and tries to CANCEL all other branches. But the second 200 OK device have already responded OK and doesn't acknowledge the CANCEL. So, there's now one established dialog with media, and another "established" one neither having media nor being correctly terminated. Looks like the similar scenario is described in section 3.1.2 of RFC 5407 (Example Call Flows of Race Conditions in the Session Initiation Protocol (SIP)). Is it possible to mitigate such scenarios in a Kamailio proxy? E.g., is it possible to send an additional BYE to these CANCEL'led branches? In current configuration, Kamailio proxy retrieves contacts with registrar:lookup() and does parallel forking with tm:t_relay().

2 2

Re: Dealing with flood of invalid SIP messages
by Juha Heinanen 04 Nov '24

04 Nov '24

I gave APIBAN a try. After a few days, no requests were banned by APIBAN, but during the same period, lots of request were banned by other checks in my config. That made me wonder if APIBAN really is effective. -- Juha

2 3

Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo
by matilopi17＠gmail.com 01 Nov '24

01 Nov '24

Hello, This is kamailio version 5.8.2 and we`re having this issue every now and then. The error displayed is: ** ERROR: Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo ** ERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters The problem is that the FIFO file /run/kamailio/kamailio_rpc.fifo doesn't exist, even though the "jsonrpc" module is in the Kamailio configuration, and the file is created when Kamailio service is restarted. But as mentioned, after few days the issue is repeated. I`ll appreciate any suggestion or advice of what I should check. Thank you!

4 5

Failed to parse the R-URI with topoh enabled
by Ale 29 Oct '24

29 Oct '24

Hi All, I think I have a problem with topoh, but I can't identify it. Kamailio 5.6.6, used as a stateless proxy, receives a 487 from a phone and propagates it correctly. Next comes the ack that should be forwarded back to the phone, but instead it doesn't forward it and produces the following error "pv_get_ruri(): failed to parse the R-URI" The error is generated by xlog("...$ru from $fu...\n"") as first line of request_route. If I disable topoh everything works perfectly.Topoh is only configured with mask_key only. Furthermore, not all phones generate this problem, at the moment there are some snoms. ACK sip:user@phone_ip:2048;line=kkqgarrj SIP/2.0 Via: SIP/2.0/UDP proxy_sip_ip;branch=z9hG4bK2ec.e7126a3134bef9d974f57dd43ebd4ea2.0 Route: <sip:kamailio_ip:5060;lr;received=sip:phone_ip:2048> Max-Forwards: 66 From: <sip:111111111@sip.example.com>;tag=pDc1BQ0B57Ujj To: <sip:user@sip.example.com>;tag=vslvowy4y2 Call-ID: 40a6c905-c0c8-4c20-b3f7-397b3fce58b6 CSeq: 90285584 ACK Content Length: 0 There is no difference between the ACK of the snom and other phones that work. The only difference I noticed is that the snom 487 contains Contact field in the header. Any suggestions? Thank you

3 5

How to handle e2e ACK with t_check_trans()
by Benoit Panizzon 28 Oct '24

28 Oct '24

Hi List To correctly route ACK belonging to a transaction (totag) the example code does: if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { xlog("L_INFO", "$cfg(route): $rm NOT loose route NO transaction found DROP\n"); exit; } } Unfortunatels I fear that in a parallell forking scenario, the transaction is remove from memory as soon as the first 'negative' ACK is being processed so that the end 2 end ACK to the 200 OK of the branch that connected the call, never makes it through. I attempted to increase wt_timer to 10000 to have the transaction stay around 10 seconds but that does not solve the issue. I read that t_check_trans()'s result shall be considered more as a hint than as a confirmation of an existing transaction. Is there a clean / secure way to process e2e ACK and make sure they belong to a transaction? Or can I just safely remove the t_check_trans() call and route any ACK with a totag? 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 3

Dealing with flood of invalid SIP messages
by mayamatakeshi 27 Oct '24

27 Oct '24

Hi, I was going through some old company tickets that I am assigned to and found a case when possibly an attacker flooded our kamailio server with invalid sip messages like this: 2019-04-27T20:14:05.533554+09:00 IPX051 /usr/local/src/git/sip-router/kamailio[1732]: ERROR: <core> [parser/msg_parser.c:714]: ERROR: parse_msg: message=<[F#016sD#026Z<8D>97<F8><B5>;<A9><E7>-<D2>(<E2><F6> v;/#021k\<CC>8<B1>λ<F4>#004M<B6><BE><EC>#035#003<94><E1>=<A0><FF><E3><AF>Kwzr<8B>A#036B<D7>#027#023cu<82>Y<D4>#037<FB><AC>S_<C4>Qg<AB><DE>F<88>I#006<8C><FA><F4>~#y3G<C7>H<80>b<BC><AD>#035<89>#002<DB><C8>#001U<9E>#007<CB><F9>nT<E5><EE><8E><F1>#0144> At that time we manually banned the IP. But it would be helpful to have this done automatically by fail2ban. So I was thinking this log should include the src IP address. I looked at the latest kamailio commit and core/parser/msg_parser.c does this log the same way so I was thinking in opening an issue for this. But maybe this should be dealt with differently. Any ideas?

4 6

Issue with Retrieving Kamailio Statistics: Missing FIFO File and Errors
by satyaprakash ch 25 Oct '24

25 Oct '24

Hi, I'm encountering an issue while trying to retrieve Kamailio statistics on the server using the command kamctl stats. Instead of the expected output, I am receiving the following errors: kamctl stats: -e \E[37;31mERROR: Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo -e \E[37;31mERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters Additionally, the file /run/kamailio/kamailio_rpc.fifo does not exist. Could you please provide suggestions ?

3 4

Handeling error returned by rtpengine_manage(...)
by Wolter Eldering 23 Oct '24

23 Oct '24

Hello, I have a question about the rtpengine module and how to use it via KEMI/python. Will processing of the native script stop when rtpengine_manage returns with an error? Or will it continue and will the SDP be forwarded unchanged? #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("SIP-source-address replace-origin replace-session-connection"); } else { rtpengine_manage("replace-origin replace-session-connection"); } #!else Regards, Wolter

3 4

Kamailio not deleteing rtp sessions
by Tihomir Culjaga 22 Oct '24

22 Oct '24

Hello, I'm using Kamailio + RTPEngine. Everything works well except one thing. I noticed rtp sessions are not being deleted after call termination. Instead, they remain lingering on RTPEngine waiting for a timeout. kamailio --version version: kamailio 5.5.2 (x86_64/linux) 55e232 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: 55e232 compiled on 14:13:04 Aug 25 2021 with gcc 8.4.1 rtpengine --version Version: 9.5.1.1-1.el8 rtpengine-ctl list totals tells me every session went to timeout :) --- snip --- Total statistics (does not include current running sessions): Uptime of rtpengine :914 seconds Total managed sessions :51 Total rejected sessions :0 Total timed-out sessions via TIMEOUT :51 Total timed-out sessions via SILENT_TIMEOUT :0 Total timed-out sessions via FINAL_TIMEOUT :0 Total timed-out sessions via OFFER_TIMEOUT :0 Total regular terminated sessions :0 Total forced terminated sessions :0 --- snip --- I'm using rtpengine_manage in two routes "route[RELAY]" and onreply_route[MANAGE_REPLY] . Do I need to add a specific rtpengine_delete somewhere ? route[RELAY] { ... $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (!has_totag()) { xlog("L_INFO", "NEXIOS: proto = [$proto], ru = [$ru]"); if ($proto =~ "ws" && !($ru =~ "transport=ws")) { xlog("L_INFO", "####################### FROM WS to any ########################\n"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal"; } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { xlog("L_INFO", "####################### FROM any to WS ########################\n"); setbflag(FLB_TOWEBRTC); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF direction=internal direction=external"; } else if (($proto =~ "udp") && $ru =~ "transport=udp" && $fs =~ "MY_IP2_ADDR") { xlog("L_INFO", "####################### FROM ?? to ?? $ru ########################\n"); } } # we engage RTPEngine if (has_body("application/sdp")) { xlog("L_INFO", "NEXIOS: # $xavp(r=>$T_branch_idx) #\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[MANAGE_REPLY] { ... $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (has_totag()) { xlog("L_INFO", "onreply_route proto = [$proto], ru = [$ru]"); if ($proto =~ "ws" && !($ru =~ "transport=ws")) { xlog("L_INFO", "onreply_route ####################### FROM WS to any ########################\n"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal"; } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { xlog("L_INFO", "onreply_route ####################### FROM any to WS ########################\n"); setbflag(FLB_TOWEBRTC); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF direction=internal direction=external"; } else if (($proto =~ "udp") && $ru =~ "transport=udp" && $fs =~ "MY_IP2_ADDR") { xlog("L_INFO", "onreply_route ####################### FROM ?? to ?? $ru ########################\n"); } } if (has_body("application/sdp")) { xlog("L_INFO", "onreply_route # $xavp(r=>$T_branch_idx) #\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); } } Any ideas are appreciated. Regards, Tihomir.

4 6

parallel forking: how to fix 183 g711 then 200 ok g722?
by Benoit Panizzon 21 Oct '24

21 Oct '24

Hi Gang Parallel forking situation. A => B A => C A invites with g711,g722 which is forked to B and C B reports ringing by sending 183 + SDP g711 C report 180 ringing. C establishes connection 200 OK + SDP g722 https://www.rfc-editor.org/rfc/rfc3261.html#page-78 o If the initial offer is in an INVITE, the answer MUST be in a reliable non-failure message from UAS back to UAC which is correlated to that INVITE. For this specification, that is only the final 2xx response to that INVITE. That same exact answer MAY also be placed in any provisional responses sent prior to the answer. The UAC MUST treat the first session description it receives as the answer, and MUST ignore any session descriptions in subsequent responses to the initial INVITE. Now we run into the situation in which A received the 183 g711 and being RFC compliant, dropped the 2nd SDP from the 200 OK. => No audio! Has anyone encountered this situation and found a nice solution? Like generating an UPDATE or RE-INVITE towards A to change the codec in an RFC compliant way after the call is established? Or is the only solution to remove all codecs except the required g711a one when more than one contact are registered on an AOR? 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 5

Statically Linking Multiple Modules Compile Error
by Steve Bezalel 21 Oct '24

21 Oct '24

Hello, I am trying to compile Kamailio with all its modules statically linked. How should I go about this? I followed through the installation documentation but could not find any information. I tried passing all the modules I want to statically link in the "static_modules" argument of `make` but it seems that definitions of modules clash as such. ``` make static_modules="siputils rr" ``` ``` /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:69: multiple definition of `module_version'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:78: first defined here /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:69: multiple definition of `module_flags'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:78: first defined here /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:69: multiple definition of `module_interface_ver'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:78: first defined here /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:166: multiple definition of `exports'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:241: first defined here /usr/bin/ld: modules/rr/rr_mod.o: in function `mod_register': /playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:926: multiple definition of `mod_register'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:817: first defined here ``` Any help would be appreciated. Thank you.

1 0

$hdr behaviour using jansson+http_connect
by Sergio Charrua 19 Oct '24

19 Oct '24

Hi all! My team and I are finalizing implementation of STIR/SHAKEN. The flow has been simplified and is something like: Call comes from SBC to Kamailio, Kamailio makes an HTTP request to a REST API, response is received by Kamailio (5.8.2) and depending on the response, it replies back to SBC with a SIP 300 Multiple Choice of a SIP 4xx error, denying the call. The INVITEs received by SBC may or may not contain the Identity header. On Kamailio side, the following code gets the required headers to build the JSON object sent to REST API: route[HANDLE_STIRSHAKEN] { if (!is_method("INVITE")) { return; } $var(requestTime) = $utimef(%s); $var(ruri) = "$ru"; #request URI $var(from) = "$hdr(From)"; #$fU From header $var(to) = "$hdr(To)"; #$tU; To header $var(contact) = "$hdr(Contact)"; #Contact header $var(callID) = "$ci"; # Call ID $var(identity) = "$hdr(Identity)"; # Identity header jansson_set("string","from" , $var(from) , "$var(post)"); jansson_set("string","to" , $var(to) , "$var(post)"); jansson_set("string","r-uri" , $var(ruri) , "$var(post)"); jansson_set("string","contact" , $var(contact) , "$var(post)"); jansson_set("string","call-id" , $var(callID) , "$var(post)"); jansson_set("string","identity" , $var(identity) , "$var(post)"); http_connect("api1", "/stsh", "application/json","$var(post)","$var(result)"); xlog("L_DEBUG", "HANDLE_STIRSHAKEN - Result = $var(result)\n"); [... do some other irrelevant stuff ...] } The usual INVITE is similar to this: INVITE sip:+33142XXXXXX@10.242.XXX.YYY:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.240.AAA.BBB:5064;branch=z9hG4bK+30f178b47ee25dd589f986ed7402b4eb3+sip+6+a64eb889 From: "+493XXXXXXX" <sip:+493XXXXXXX@10.240.AAA.BBB ;user=phone>;tag=10.240.AAA.BBB+6+ffe686f8+614d17a3 To: +33142XXXXXX <sip:+33142XXXXXX@10.242.XXX.YYY> CSeq: 1 INVITE Expires: 180 Content-Length: 234 Call-Info: <sip:10.240.AAA.BBB:5064>;method="NOTIFY;Event=telephone-event;Duration=2000" Supported: timer, 100rel Contact: <sip:+493XXXXXXX@10.240.AAA.BBB:5064;user=phone> Content-Type: application/sdp Call-ID: 0gQAAC8WAAACBAAALxYAAHRfFdrMLbnN/iMYu2NTT6kvhuhZAFl7MG1V2ozJLUcXg9dxbyQZ/PtoswtdHtcv9g--(a)10.240.AAA.BBB Min-SE: 90 Session-Expires: 90;refresher=uac X-dtmf: 5 P-Asserted-Identity: "+493XXXXXXX" <sip:+493XXXXXXX@10.240.AAA.BBB> Accept: application/sdp Accept: application/dtmf-relay Max-Forwards: 66 Allow: INVITE,CANCEL,BYE,ACK, INFO Identity: dGhpcyBpcyBzb21lIGR1bW15IGJhc2U2NCBlbmNvZGVkIElkZW50aXR5IGhlYWRlcg==.eW91IHdvdWxkIG5vdCBleHBlY3QgbWUgdG8gbGVhdmUgdGhlIHJlYWwgdmFsdWUgaGVyZSByaWdodD8=.dGhhbmtzIGZvciB5b3VyIGNvbXByZWhlbnNpb24=;info=< https://someURL.com/with/some/parameter;sc697295b.cer>;alg=ES256;ppt=shaken P-Origination-Id: 23A16F8B-F9AB-4536-9884-85C96EF3B964 P-Attestation-Indicator: C has-sdp: present [SDP stuff] (I have replaced some of the values for obvious reasons). - SBC is sending every 70 secs 2 automatically generated INVITEs to kamailio - 1st call contains the Identity header, the other call does not contain Identity header. - Both calls have different Call-IDs and different FROM and TO values, so they are different calls and can be perfectly found in logs as separated calls - Kamailio receives each call and extracts headers, as shown in code above, and sends the JSON request to HTTP REST API Here is the weird part.... every now and then, completely random, the JSON object request sent to the HTTP REST API contains an Identity value, supposedly extracted from the SIP header **but** the SIP message captured in SNGrep doesn't have **any* I*dentity header!! In fact, in the kamailio logs, where I sent the header values extracted from INVITE, the identity is empty! BUT the JSON object has the identity set with value. And the worst part is that, when this issue happens, the value of the Identity set in JSON belongs to a completely different call that was processed a few minutes before ! So I suspected about some Caching behaviour on the http_connect function and added the following lines before sending the JSON to REST API: if ( is_present_hf("Identity") ) { jansson_set("string","identity",* $var(identity),* "$var(post)"); } else jansson_set("string","identity", "", "$var(post)"); forcing the identity property of the JSON to an empty string, but this did not work either! I even added a new property which is the current timestamp making sure every request was unique, and that did not work either! The only way I could make it work was replacing the above lines using these lines: if ( is_present_hf("Identity") ) { jansson_set("string","identity", *(a)hf_value2.Identity*, "$var(post)"); } else jansson_set("string","identity", "", "$var(post)"); So, instead of using $hdr(Identity) , Kamailio extracts the Identity header using @hf_value2.Identity Question: why is this behaviour? Could this be a bug somewhere? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 3

Re: Dispatcher ping: "binding to source address X.X.X.X:XXXX failed: Address already in use" for TCP and TLS destinations
by Mirko Brankovic 15 Oct '24

15 Oct '24

Hi, Has anyone found how to safely suppress the warning? We see it on 5.7.4 Thanks, Mirko

2 2

kamailio - acc sqlite
by marek 14 Oct '24

14 Oct '24

hi, i'm trying setup kamailio for simple CDR logging versions - kamailio 5.7, rocky 9.4 i have configured and populate sqlite with "kamdbctl create" kamailio.cfg #CDR loadmodule "db_sqlite.so" loadmodule "acc.so" modparam("acc", "cdr_enable", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "sqlite:///var/log/kamailio-cdr.sqlite") modparam("acc", "time_mode", 3) but i have this error ERROR: db_sqlite [dbase.c:232]: db_sqlite_submit_query(): failed to prepare statement: table acc has no column named time_attr any ideas where can i find acc schema update for time_attr column? bash# grep -Rni time_attr /usr/share/kamailio/db_sqlite/ shows nothing Marek

2 2

Getting the name of an already set onreply_route
by Denys Pozniak 14 Oct '24

14 Oct '24

Hello! Is it possible to find out the name of the onreply_route that was set before? Something like this: t_on_reply("MANAGE_REPLY"); ... if ( t_is_set("onreply_route") ) { get_onreply_route_name(); ... } -- BR, Denys Pozniak

3 5

Appending Branches Post Request Processing
by Knserbrave 08 Oct '24

08 Oct '24

Hi Community, Is it possible to append a branch to a transaction after the call is in a ringing state? Say a call comes in and an AOR only has two contacts. Branches are created to the contacts. A third comes in why the two are in the ringing state. Can we create a branch to the third?

2 3

TLS module crashes with FIPS OpenSSL
by Marat Gareev 08 Oct '24

08 Oct '24

Hello, I encountered a problem stopping Kamailio with FIPS OpenSSL: Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007ff7292380ac in OPENSSL_sk_pop () from /lib64/libcrypto.so.3 Missing separate debuginfos, use: dnf debuginfo-install kamailio-5.7.3-4816.x86_64 (gdb) bt #0 0x00007ff7292380ac in OPENSSL_sk_pop () from /lib64/libcrypto.so.3 #1 0x00007ff72914bf5b in conf_modules_finish_int () from /lib64/libcrypto.so.3 #2 0x00007ff72914c694 in CONF_modules_unload () from /lib64/libcrypto.so.3 #3 0x00007ff7291efff9 in OPENSSL_cleanup () from /lib64/libcrypto.so.3 #4 0x00007ff72954702b in ?? () #5 0x0000000100061c08 in ?? () #6 0x00007ff7190566c8 in ?? () #7 0x00007ffccf196a20 in ?? () #8 0x000000000071da8a in futex_release (lock=0x7ff729f08b50 <syslog>) at core/mem/../mem/../futexlock.h:134 #9 0x00000000006e9448 in destroy_tls () at core/tls_hooks.c:75 #10 0x000000000041f278 in cleanup (show_status=1) at main.c:594 #11 0x0000000000420af1 in shutdown_children (sig=15, show_status=1) at main.c:721 #12 0x0000000000421717 in handle_sigs () at main.c:752 #13 0x0000000000430c88 in main_loop () at main.c:1988 #14 0x0000000000439d13 in main (argc=14, argv=0x7ffccf1973f8) at main.c:3212 (gdb) Environment: Oracle Linux Server 9.3 Kamailio 5.7.3 yum list --installed | grep ssl openssl.x86_64 10:3.0.7-24.0.3.el9_fips @tools openssl-libs.x86_64 10:3.0.7-24.0.3.el9_fips @tools openssl-pkcs11.x86_64 0.4.11-7.el9 @anaconda xmlsec1-openssl.x86_64 1.2.29-9.el9 @AppStream What can I do for further investigation? Thanks

5 12

t_append_branches() and "dead" tcp connections
by Ihor Olkhovskyi 04 Oct '24

04 Oct '24

Hello, Recently encountered an issue with tsilo (that is using t_append_branches), but not sure found a root cause. So, I have a call, but aparently there are some "dead" TCP connections in a "location" table modparam("usrloc", "handle_lost_tcp", 0) On call initialtion, INVITE's are sent to "last seen" location, but aparently there is no answer and I assume it's in a state "trying to establish TCP connection" Obviously, there is no "100 Trying" answer on any outgoing INVITE. Than, a REGISTER comes from a contact, that is supposed to be called. Usual ts_append("location", "$tu"); is called, but in the logs I see [REGISTER] tm [t_append_branches.c:208]: t_append_branches(): failure to add branches (-1) If there is an active TCP connection, branches are added as expected. I'm getting this only in a case of trying to establish a call to a "dead" TCP connection. What am I missing here or it sounds like a bug? -- Best regards, Ihor (Igor)

2 9

kamailio sending from udp non-listening port
by David Villasmil 04 Oct '24

04 Oct '24

Hello guys, i'm seeing something weird, and i'm working if you can let me know. I have a kamailio in AWS with a private IP listening on several sockets: Listening on udp: 10.1.2.36:5070 udp: 10.1.2.36:5080 udp: 10.1.2.36:5160 advertise 4.3.2.1:5160 udp: 0.0.0.0:5066 tls: 10.1.2.36:443 advertise sip.something.com:443 tls: 10.1.2.36:444 advertise sip.something.com:444 tls: 10.1.2.36:5061 When forwarding a udp invite received on 10.1.2.36:5080 to a public ip provider say on 8.8.8.8:5060, i'm forcing the outgoing socket with force_socket via 10.1.2.36:5160. But the outgoing invite does NOT use 5160, it uses some random port... Anybody knows why this might be? my problem is, that call goes to freeswitch... call is setup properly and connects fine. But 15 minutes later the end provider sends back a reINVITE, which freeswitch then sends TO THE RANDOM PORT kamailio used to send the INVITE... but by this time kamailio doesn't seem to even see the packet... help is greatly appreciated! David Regards, David Villasmil email: david.villasmil.work(a)gmail.com

4 9

relay to voip provider / sip trunk / path?
by Linux Admin 03 Oct '24

03 Oct '24

Hi. I am using https://github.com/havfo/WEBRTC-to-SIP to have a web sipcaller. it uses kamailio + rtpengine + web sipcaller (jssip) right now it is working using internal kamailo account and can call to another sip. But I need to use websipcaller to call to voip provider. I tried to register on my voip provider, and use just outbound websocket proxy, but it not working giving 401 the config I am using is https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Please help adopt it to relay anything to external sip provider. I don't know what I need, relay, sip trunk or path module. kamailio configs are very hard for me to understand. So I need it just as websocket proxy to my voip provider, that doesn't support websocket. thanks!

2 2

app_lua parse multiple files
by iliusha.md＠gmail.com 03 Oct '24

03 Oct '24

I have a Lua script which is split into multiple files, like: /etc/kamailio/routing/ ├── main.lua ├── logger.lua └── db.lua On restart, Kamailio fails to start with an error: main.lua:2: module 'logger' not found:\n\tno field package.preload If I set multiple files in the modparam parameter: modparam("app_lua", "load", "/etc/kamailio/routing/main.lua") modparam("app_lua", "load", "/etc/kamailio/routing/logger.lua") I'm getting the same error. Is it possible to load multiple files, or is this a limit to a single file currently?

4 3

Heads-up: Clean-up of legacy modules in app_python3
by Richard Chan 01 Oct '24

01 Oct '24

HI kamailio users, There is a WIP PR on master (and targeting 6.0) to clean-up legacy modules on app_python3. https://github.com/kamailio/kamailio/pull/3986/commits/a9eb4b35b67b58893772… If you rely on pre-KEMI features (Core / Router / Logger / Ranks) can you chime in here. This work is also in preparation to enable the Kamailio module to use free-threading builds of Python. Regards Richard

1 0

Sync Suspended Transaction
by Knserbrave 30 Sep '24

30 Sep '24

Is it possible to sync a suspended transaction between two nodes using t_suspend() from the TMX module and DMQ (or similar module)? More specifically, can I suspend it on one node and resume the transaction on another in a cluster?

4 6

sys.exit() in app_python3s
by Denys Pozniak 26 Sep '24

26 Sep '24

Hello! I'm planning to rewrite a native Kamailio script in KEMI Python, and I want to understand the differences between app_python3 vs app_python3s in the context of using sys.exit(). My native script uses the exit function in deep nested route[] blocks quite a bit and would like to keep this logic in KEMI without using return codes on top (return -255) towards ksr_request_route() Using sys.exit() in app_python3s gives an error in syslog (although the interpreter does not crash and everything works correctly), in app_python3 no such error ERROR: app_python3s [apy3s_kemi.c:146]: apy3s_exec_func(): error exception occurred How safe is it to use sys.exit() in this case? -- BR, Denys Pozniak

1 0

Re: About error running PCSCF
by LUIS FELIPE RAMOS TORRES 25 Sep '24

25 Sep '24

Dear community: If someone runs this tutorial successfully for dockerized open5GS, please I appreciate it. Web page of OPEN 5GS: [image: image.png] The APNs were created in the docker_host:8080/docs [image: image.png] When i ran the command: docker compose -f srsenb.yaml up - d && docker container attach srsenb [image: image.png] [image: image.png] I created 2 APNs on the phones, but when I select one of them (internet), I have access to the Internet. About VOLTE, I don't know how to try or maybe I must change to the other APN (ims). [image: image.png] Please let me know, I only have the problem with VOLTE service and how to try it. Best wishes, Luis El mar, 24 sept 2024 a las 19:23, LUIS FELIPE RAMOS TORRES (< luis.ramost(a)pucp.edu.pe>) escribió: > > Sorry if I replied to you again. > > If you can explain to me why we need to create 2 APNs, 1 for the ims and > other for the internet. I created the subscriber on Open5GS. > > [image: image.png] > > > I also create the apns. I attached one of them to this email. > [image: image.png] > > I do everything that you put on github, but when I run the command for the > OTA, my cell phone registers to the network, and when I create manually the > 2 APNS, I select the "internet" and have access, but I don't know how to > configure VOLTE. The model is a Huawei p30 lite, and also I don't know if > it supports VOLTE. > > [image: image.png] > [image: image.png] > > Cellular connected to the network: > [image: image.png] > > > Also in this tutorial ( > https://open5gs.org/open5gs/docs/tutorial/03-VoLTE-dockerized/ ), I see > that you use an app called CoIMS for VoLTE. Is that enabled for that > service or what to do exactly? > > Sorry again for my replies and comments. > > Luis > > > El mar, 24 sept 2024 a las 17:04, LUIS FELIPE RAMOS TORRES (< > luis.ramost(a)pucp.edu.pe>) escribió: > >> Hi Supreeth: >> >> My question is if the MSISDN is necessary to add to the parameters, >> because when I configure the SIM card, I don't generate any for that value. >> Or maybe to make a call. >> >> And if I have an error in making the call for VOLTE, maybe it would be a >> problem of a configuration, or maybe the phones. How to know that? >> >> Luis >> >> El mar, 24 sept 2024 a las 2:48, Supreeth Herle (<herlesupreeth(a)gmail.com>) >> escribió: >> >>> Hi Luis, >>> >>> In the next screenshot, I don't know if the next command (docker compose >>>> -f srsenb-yaml up ...) must run in another terminal window or in the same >>>> as I run the first command. >>> >>> >>> In another terminal window. >>> >>> I will not use the RF simulated because it would be physical and I have >>>> the LIMESDR hardware. Also the UE with 4G sim cards. >>> >>> >>> Run only the one which says OTA (over-the-air) >>> >>> And next of the following steps, when I run the others and add the >>>> subscriber, the apn, the ims, and the SIM parameters. What is the next >>>> step? Try the solution by making a call? >>>> >>> >>> yes >>> >>> BR, >>> Supreeth >>> >>> >>> >>> On Tue, 24 Sept 2024 at 02:35, LUIS FELIPE RAMOS TORRES < >>> luis.ramost(a)pucp.edu.pe> wrote: >>> >>>> Hi Supreeth: >>>> >>>> I do nothing but finally it runs successfully. But I have a question. I >>>> am running now the command of the screenshot (docker compose -f >>>> 4g-volte-deploy.yaml up) and I have the next output: >>>> >>>> [image: image.png] >>>> >>>> >>>> In the next screenshot, I don't know if the next command (docker >>>> compose -f srsenb-yaml up ...) must run in another terminal window or in >>>> the same as I run the first command. >>>> >>>> [image: image.png] >>>> I will not use the RF simulated because it would be physical and I have >>>> the LIMESDR hardware. Also the UE with 4G sim cards. >>>> >>>> And next of the following steps, when I run the others and add the >>>> subscriber, the apn, the ims, and the SIM parameters. What is the next >>>> step? Try the solution by making a call? >>>> >>>> Please if you can help me. >>>> >>>> Best wishes, >>>> Luis >>>> >>>> >>>> El dom, 22 sept 2024 a las 1:46, Supreeth Herle (< >>>> herlesupreeth(a)gmail.com>) escribió: >>>> >>>>> Hi Luis, >>>>> >>>>> Please raise an issue in >>>>> https://github.com/herlesupreeth/docker_open5gs >>>>> <https://github.com/herlesupreeth/docker_open5gs/tree/master> repo. I >>>>> will try to resolve it. >>>>> >>>>> I just ran the same command (after removing the image built locally) >>>>> and that step passed. So I would suggest to check your network connection >>>>> or DNS settings of the machine on which docker_open5gs is cloned. >>>>> >>>>> WARN[0000] /home/sherle/docker_open5gs/4g-volte-deploy.yaml: the >>>>>> attribute `version` is obsolete, it will be ignored, please remove it to >>>>>> avoid potential confusion >>>>>> [+] Building 19.0s (8/8) FINISHED >>>>>> >>>>>> >>>>>> >>>>>> docker:default >>>>>> => [osmohlr internal] load build definition from Dockerfile >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => => transferring dockerfile: 2.10kB >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => WARN: JSONArgsRecommended: JSON arguments recommended for CMD to >>>>>> prevent unintended behavior related to OS signals (line 41) >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr internal] load metadata for >>>>>> docker.io/library/debian:bullseye >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr internal] load .dockerignore >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => => transferring context: 2B >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => CACHED [osmohlr 1/3] FROM docker.io/library/debian:bullseye >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr 2/3] RUN apt-get update && apt-get -y install wget >>>>>> gnupg telnet >>>>>> >>>>>> >>>>>> 14.1s >>>>>> => [osmohlr 3/3] RUN wget >>>>>> https://downloads.osmocom.org/packages/osmocom:/latest/Debian_11/Release.key >>>>>> && mv Release.key /etc/apt/trusted.gpg.d/osmocom-latest.asc && echo >>>>>> "deb [signed-by=/etc/apt/trusted.gpg.d/osmocom-latest.asc] >>>>>> https://downloads.osmocom.org/packages/osmocom:/latest/Debian_11/ >>>>>> ./" > /etc/apt/sources.list.d/osmocom-latest.list && apt 4.5s >>>>>> => [osmohlr] exporting to image >>>>>> >>>>>> >>>>>> >>>>>> 0.2s >>>>>> => => exporting layers >>>>>> >>>>>> >>>>>> >>>>>> 0.2s >>>>>> => => writing image >>>>>> sha256:95227b65ff159ba4eab30f2caa460163b76895703c8cc73f41735d1ef4ef6dfc >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => => naming to docker.io/library/docker_osmohlr >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr] resolving provenance for metadata file >>>>> >>>>> >>>>> BR, >>>>> Supreeth >>>>> >>>>> On Sat, 21 Sept 2024 at 20:42, LUIS FELIPE RAMOS TORRES < >>>>> luis.ramost(a)pucp.edu.pe> wrote: >>>>> >>>>>> Hello: >>>>>> >>>>>> I don't know if somebody can successfully run this tutorial for a >>>>>> solution of open5gs with kamailio using docker. I have the next error after >>>>>> the command: >>>>>> >>>>>> docker compose -f 4g-volte-deploy.yaml build >>>>>> and the output is: >>>>>> [image: image.png] >>>>>> >>>>>> [image: image.png] >>>>>> >>>>>> [image: image.png] >>>>>> >>>>>> I am running on Ubuntu 22.04, I also don't know if it is the same as >>>>>> Debian 11 (in the link of the download). Or how can I solve it? >>>>>> >>>>>> Luis >>>>>> >>>>>> El mié, 18 sept 2024 a las 19:30, LUIS FELIPE RAMOS TORRES (< >>>>>> luis.ramost(a)pucp.edu.pe>) escribió: >>>>>> >>>>>>> Hello >>>>>>> >>>>>>> Sorry for writing again, I followed your repository on github: >>>>>>> https://github.com/herlesupreeth/docker_open5gs/tree/master, but I >>>>>>> don't know how to build srslte: >>>>>>> >>>>>>> I have the next error when I build the srslte: docker build >>>>>>> --no-cache --force-rm -t docker_srslte . >>>>>>> >>>>>>> I don't know if it is updated, maybe also I run it on a VM, and >>>>>>> specifically in the "git checkout ......" I had the error. >>>>>>> >>>>>>> [image: image.png] >>>>>>> Luis >>>>>>> >>>>>>> El mar, 17 sept 2024 a las 1:45, Supreeth Herle (< >>>>>>> herlesupreeth(a)gmail.com>) escribió: >>>>>>> >>>>>>>> Hello Luis, >>>>>>>> >>>>>>>> When I use the github of RTPEngine, the master branch, when I run >>>>>>>>> the command dpkg-checkbuildeps >>>>>>>> >>>>>>>> >>>>>>>> I would not recommend installing the master branch of RTPEngine, >>>>>>>> rather use the tag mr9.4.1 and install on Ubuntu 20.04. You can find all >>>>>>>> dependencies to install here - >>>>>>>> https://github.com/herlesupreeth/docker_open5gs/blob/master/rtpengine/Docke… >>>>>>>> >>>>>>>> If you just want a working setup of IMS + RTPEngine I would highly >>>>>>>> recommend using https://github.com/herlesupreeth/docker_open5gs >>>>>>>> repo. You can get the setup running with few steps. >>>>>>>> >>>>>>>> BR, >>>>>>>> Supreeth >>>>>>>> >>>>>>>> On Tue, 17 Sept 2024 at 03:06, LUIS FELIPE RAMOS TORRES < >>>>>>>> luis.ramost(a)pucp.edu.pe> wrote: >>>>>>>> >>>>>>>>> Hello Supreeth: >>>>>>>>> >>>>>>>>> Sorry if I reply to the message, but I am trying to install the >>>>>>>>> tutorials, but I've got the same error. When I use the github of RTPEngine, >>>>>>>>> the master branch, when I run the command dpkg-checkbuildeps, I get errors. >>>>>>>>> >>>>>>>>> If you would have an updated tutorial to run successfully and also >>>>>>>>> recommend the Ubuntu version to avoid more errors, please it would help me. >>>>>>>>> >>>>>>>>> Luis >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> El lun, 16 sept 2024 a las 10:32, Supreeth Herle (< >>>>>>>>> herlesupreeth(a)gmail.com>) escribió: >>>>>>>>> >>>>>>>>>> Hello Luis, >>>>>>>>>> >>>>>>>>>> I am the author of the tutorial here - >>>>>>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. That >>>>>>>>>> tutorial is also not up-to-date. However, you can use the latest master >>>>>>>>>> branch of kamailio, along with instructions here (installing dependencies >>>>>>>>>> etc) - >>>>>>>>>> https://github.com/herlesupreeth/docker_open5gs/blob/master/ims_base/Docker… >>>>>>>>>> and configuration files from master branch of this repo >>>>>>>>>> https://github.com/herlesupreeth/Kamailio_IMS_Config . >>>>>>>>>> >>>>>>>>>> As pointed out earlier, rtpproxy is not needed. It was a step I >>>>>>>>>> added to test out VoIP (not IMS) calling. All you need is RTPEngine to test >>>>>>>>>> out IMS. >>>>>>>>>> >>>>>>>>>> Hope it helps. >>>>>>>>>> >>>>>>>>>> BR, >>>>>>>>>> Supreeth >>>>>>>>>> >>>>>>>>>> On Mon, 16 Sept 2024 at 17:22, LUIS FELIPE RAMOS TORRES via >>>>>>>>>> sr-users <sr-users(a)lists.kamailio.org> wrote: >>>>>>>>>> >>>>>>>>>>> Hello Henning: >>>>>>>>>>> >>>>>>>>>>> Is this tutorial updated? >>>>>>>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ >>>>>>>>>>> >>>>>>>>>>> I don´t know about it. And also I guess that it is installed on >>>>>>>>>>> Ubuntu 18 version. >>>>>>>>>>> >>>>>>>>>>> Cheers, >>>>>>>>>>> Luis >>>>>>>>>>> >>>>>>>>>>> El lun, 16 sept 2024 a las 1:32, Henning Westerholt (< >>>>>>>>>>> hw(a)gilawa.com>) escribió: >>>>>>>>>>> >>>>>>>>>>>> Hello, >>>>>>>>>>>> >>>>>>>>>>>> the tutorial is using a Kamailio version that its long end of >>>>>>>>>>>> life (5.3.x). >>>>>>>>>>>> >>>>>>>>>>>> Some of the instructions given there are also a bit odd, like >>>>>>>>>>>> first installing rtpproxy, and then replacing it with rtpengine. >>>>>>>>>>>> >>>>>>>>>>>> Maybe you can reach out to the author of the tutorial. You need >>>>>>>>>>>> to understand how the individual components work together in the end to be >>>>>>>>>>>> able to debug it. Maybe somebody else figured out on this list how to do it >>>>>>>>>>>> for an updated Kamailio. >>>>>>>>>>>> >>>>>>>>>>>> Cheers, >>>>>>>>>>>> >>>>>>>>>>>> Henning >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> Henning Westerholt - https://skalatan.de/blog/ >>>>>>>>>>>> Kamailio services - https://gilawa.com >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> > -----Original Message----- >>>>>>>>>>>> > From: luis.ramost--- via sr-users < >>>>>>>>>>>> sr-users(a)lists.kamailio.org> >>>>>>>>>>>> > Sent: Montag, 16. September 2024 02:14 >>>>>>>>>>>> > To: sr-users(a)lists.kamailio.org >>>>>>>>>>>> > Cc: luis.ramost(a)pucp.edu.pe >>>>>>>>>>>> > Subject: [SR-Users] About error running PCSCF >>>>>>>>>>>> > >>>>>>>>>>>> > Hi community: >>>>>>>>>>>> > >>>>>>>>>>>> > I am trying to run the Kamailio IMS, but I have the next >>>>>>>>>>>> error in the module >>>>>>>>>>>> > PCSCF >>>>>>>>>>>> > >>>>>>>>>>>> > I installed the Kamailio on a VM Machine, with UBuntu 18.04 >>>>>>>>>>>> > >>>>>>>>>>>> > Commands: >>>>>>>>>>>> > $ sudo -i >>>>>>>>>>>> > # mkdir -p /var/run/kamailio_pcscf >>>>>>>>>>>> > #kamailio -f /etc/kamailio_pcscf/kamailio_pcscf.cfg -P >>>>>>>>>>>> /kamailio_pcscf.pid -DD >>>>>>>>>>>> > -E -e >>>>>>>>>>>> > >>>>>>>>>>>> > Output: >>>>>>>>>>>> > loading modules under config path: >>>>>>>>>>>> > >>>>>>>>>>>> /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kama >>>>>>>>>>>> > >>>>>>>>>>>> ilio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux- >>>>>>>>>>>> > gnu/kamailio/modules/:/usr/local/lib64/kamailio/modules >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <delete_delay> of type <2:int> not found in module >>>>>>>>>>>> > <ims_registrar_pcscf> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 399, column >>>>>>>>>>>> 51: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <ipsec_preferred_ealg> of type <1:string> not found >>>>>>>>>>>> in module >>>>>>>>>>>> > <ims_ipsec_pcscf> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 417, column >>>>>>>>>>>> 59: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <recv_mode> of type <2:int> not found in module >>>>>>>>>>>> <ims_qos> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 435, column >>>>>>>>>>>> 35: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <dialog_direction> of type <2:int> not found in >>>>>>>>>>>> module <ims_qos> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 436, column >>>>>>>>>>>> 42: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks >>>>>>>>>>>> array size 16 >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 260, column >>>>>>>>>>>> 42: unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_create (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 264, column >>>>>>>>>>>> 77: unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 114, column 31: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 180, column 33: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 11, column 34: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 99, column 42: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > ERROR: bad config file (10 errors) >>>>>>>>>>>> > 0(470) WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): >>>>>>>>>>>> > WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: >>>>>>>>>>>> > cdp_avp_destroy(226) >>>>>>>>>>>> > 0(470) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer >>>>>>>>>>>> child stopping >>>>>>>>>>>> > ... >>>>>>>>>>>> > 0(470) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... >>>>>>>>>>>> CDiameterPeer child >>>>>>>>>>>> > stopped >>>>>>>>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: >>>>>>>>>>>> > mod_destroy(): Error destroying spi generator >>>>>>>>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:313]: >>>>>>>>>>>> > mod_destroy(): Error destroying port generator >>>>>>>>>>>> > >>>>>>>>>>>> > I guided form this tutorial: >>>>>>>>>>>> https://ryantheelder.github.io/blog/2023/VoLTE/ >>>>>>>>>>>> > >>>>>>>>>>>> > Please if somebody would help me, I appreciate it >>>>>>>>>>>> > >>>>>>>>>>>> > Finally my objective is to connect with srsRAN and open5GS to >>>>>>>>>>>> have VoLTE >>>>>>>>>>>> > service. >>>>>>>>>>>> > >>>>>>>>>>>> > Best wishes, >>>>>>>>>>>> > Luis >>>>>>>>>>>> > __________________________________________________________ >>>>>>>>>>>> > Kamailio - Users Mailing List - Non Commercial Discussions To >>>>>>>>>>>> unsubscribe >>>>>>>>>>>> > send an email to sr-users-leave(a)lists.kamailio.org >>>>>>>>>>>> > Important: keep the mailing list in the recipients, do not >>>>>>>>>>>> reply only to the >>>>>>>>>>>> > sender! >>>>>>>>>>>> > Edit mailing list options or unsubscribe: >>>>>>>>>>>> >>>>>>>>>>> __________________________________________________________ >>>>>>>>>>> Kamailio - Users Mailing List - Non Commercial Discussions >>>>>>>>>>> To unsubscribe send an email to >>>>>>>>>>> sr-users-leave(a)lists.kamailio.org >>>>>>>>>>> Important: keep the mailing list in the recipients, do not reply >>>>>>>>>>> only to the sender! >>>>>>>>>>> Edit mailing list options or unsubscribe: >>>>>>>>>>> >>>>>>>>>>

1 0

Re: [EXTERNAL] Re: SDP parsing fails when media stream is removed, there is no session-wide "c=" line and offending media stream omits all attributes
by Thomas Baumgartner 24 Sep '24

24 Sep '24

Hi Richard, thanks for your input. And i guess you are correct, when reading the sentence with the wording of the SDP RFC in mind attributes should only be "a=" lines. We will talk to the device vendor again with this new finding! Anyways, maybe it still makes sense to adjust the SDP parser for such scenarios, as it really doesn't make sense to require a "c=" line if the stream is being removed as you said. I will open an git issue aswell. Mit kommunikativen Grüßen I Best regards Thomas

2 1

SDP parsing fails when media stream is removed, there is no session-wide "c=" line and offending media stream omits all attributes
by Thomas Baumgartner 24 Sep '24

24 Sep '24

Hi, we have a problem with a device trying to send T.38 fax. During the T.38 ReInvite Kamailio fails to parse the SDP (via rtpengine_manage), and subsequently rtpengine is failing to do it's job. The problem seems to be that Kamailio is trying to find a media address, since there is no session-wide c= line present it goes to the media c= lines, but since the first media stream which was an audio stream is being removed (port 0) and can therefore omit all attributes (as per RFC 3264 8.2, or am i understanding this wrong?), the Kamailio SDP parser is failing, even if there is another c= line in the T.38 media stream. Here is the offending SDP from the ReInvite (IP changed): v=0 o=xmserver 1726638425 1726638427 IN IP4 169.254.1.1 s=xmserver t=0 0 m=audio 0 RTP/AVP 8 m=image 56002 udptl t38 c=IN IP4 169.254.1.1 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy And here is the corresponding log entry: <core> [core/parser/sdp/sdp.c:523]: parse_sdp_session(): can't find media IP in the message Can anyone confirm my findings? Is it possible to adjust the SDP parser for these kind of scenarios? Mit kommunikativen Grüßen I Best regards Thomas

3 3

t_next_contacts fails with 1 contact
by Igor Potjevlesch 24 Sep '24

24 Sep '24

Hello! I'm parsing 302 with, most of the time, multi contacts returned. I do it like this: get_redirects("*"); t_load_contacts(); and, finally, a test on !t_next_contacts(). This test fails when only one contact is present in 302. Is it the expected behaviour? From the documentation I understand that following t_load_contacts we must use t_next_contacts. If so, I believe that I shouldn't exit on !t_next_contacts() and do an additional test to see if there is, at least, one attempt to do toward the only contact returned? Thanks for your feedback. Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

1 1

Honor Retry-After
by Knserbrave 23 Sep '24

23 Sep '24

Hi Community, Any advice on how to handle a Retry-After header locally without sleeping? I would like to suspend the transaction, schedule an event, wake up the transaction after the event timer triggers, and resume the transaction. I don't believe this is possible though.

3 4

Kamailio packages for 24.04.1 LTS (Noble Numbat)
by Leonid Fainshtein 23 Sep '24

23 Sep '24

Hello, Is it planned to release the Kamailio 5.8 deb packages for 24.04.1 LTS (Noble Numbat)? Best regards, Leonid

2 3

Pseudo Variable containing 'path' of main branch? / How to compose $du manually with path?
by Benoît Panizzon 23 Sep '24

23 Sep '24

Hi List I want to be able to parallel branch a call to multiple registered locations / multiple different AOR. So using the registrar lookup() function can not be used. I loop through all required AOR with reg_fetch_contacts, those could be registered via a proxy and therefore require to use Path: $ulc(aor=>path) in this case, contains the path to that destination. On all 'additional' branches added with append_branch() I can set the path using $branch(path). But i struggle with the main aka first branch on which I directly set: * $ru * $fs * $du From the documentation, if I would use the registrar lookup() function, then $du would correctly be set respecting the path for all contacts on an AOR. But who do I build $du manually with what I find in $ulc(aor=>path) when not using lookup()? -- 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 6

Re: About error running PCSCF
by LUIS FELIPE RAMOS TORRES 21 Sep '24

21 Sep '24

Hello: I don't know if somebody can successfully run this tutorial for a solution of open5gs with kamailio using docker. I have the next error after the command: docker compose -f 4g-volte-deploy.yaml build and the output is: [image: image.png] [image: image.png] [image: image.png] I am running on Ubuntu 22.04, I also don't know if it is the same as Debian 11 (in the link of the download). Or how can I solve it? Luis El mié, 18 sept 2024 a las 19:30, LUIS FELIPE RAMOS TORRES (< luis.ramost(a)pucp.edu.pe>) escribió: > Hello > > Sorry for writing again, I followed your repository on github: > https://github.com/herlesupreeth/docker_open5gs/tree/master, but I don't > know how to build srslte: > > I have the next error when I build the srslte: docker build --no-cache > --force-rm -t docker_srslte . > > I don't know if it is updated, maybe also I run it on a VM, and > specifically in the "git checkout ......" I had the error. > > [image: image.png] > Luis > > El mar, 17 sept 2024 a las 1:45, Supreeth Herle (<herlesupreeth(a)gmail.com>) > escribió: > >> Hello Luis, >> >> When I use the github of RTPEngine, the master branch, when I run the >>> command dpkg-checkbuildeps >> >> >> I would not recommend installing the master branch of RTPEngine, rather >> use the tag mr9.4.1 and install on Ubuntu 20.04. You can find all >> dependencies to install here - >> https://github.com/herlesupreeth/docker_open5gs/blob/master/rtpengine/Docke… >> >> If you just want a working setup of IMS + RTPEngine I would highly >> recommend using https://github.com/herlesupreeth/docker_open5gs repo. >> You can get the setup running with few steps. >> >> BR, >> Supreeth >> >> On Tue, 17 Sept 2024 at 03:06, LUIS FELIPE RAMOS TORRES < >> luis.ramost(a)pucp.edu.pe> wrote: >> >>> Hello Supreeth: >>> >>> Sorry if I reply to the message, but I am trying to install the >>> tutorials, but I've got the same error. When I use the github of RTPEngine, >>> the master branch, when I run the command dpkg-checkbuildeps, I get errors. >>> >>> If you would have an updated tutorial to run successfully and also >>> recommend the Ubuntu version to avoid more errors, please it would help me. >>> >>> Luis >>> >>> >>> >>> El lun, 16 sept 2024 a las 10:32, Supreeth Herle (< >>> herlesupreeth(a)gmail.com>) escribió: >>> >>>> Hello Luis, >>>> >>>> I am the author of the tutorial here - >>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. That >>>> tutorial is also not up-to-date. However, you can use the latest master >>>> branch of kamailio, along with instructions here (installing dependencies >>>> etc) - >>>> https://github.com/herlesupreeth/docker_open5gs/blob/master/ims_base/Docker… >>>> and configuration files from master branch of this repo >>>> https://github.com/herlesupreeth/Kamailio_IMS_Config . >>>> >>>> As pointed out earlier, rtpproxy is not needed. It was a step I added >>>> to test out VoIP (not IMS) calling. All you need is RTPEngine to test out >>>> IMS. >>>> >>>> Hope it helps. >>>> >>>> BR, >>>> Supreeth >>>> >>>> On Mon, 16 Sept 2024 at 17:22, LUIS FELIPE RAMOS TORRES via sr-users < >>>> sr-users(a)lists.kamailio.org> wrote: >>>> >>>>> Hello Henning: >>>>> >>>>> Is this tutorial updated? >>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ >>>>> >>>>> I don´t know about it. And also I guess that it is installed on Ubuntu >>>>> 18 version. >>>>> >>>>> Cheers, >>>>> Luis >>>>> >>>>> El lun, 16 sept 2024 a las 1:32, Henning Westerholt (<hw(a)gilawa.com>) >>>>> escribió: >>>>> >>>>>> Hello, >>>>>> >>>>>> the tutorial is using a Kamailio version that its long end of life >>>>>> (5.3.x). >>>>>> >>>>>> Some of the instructions given there are also a bit odd, like first >>>>>> installing rtpproxy, and then replacing it with rtpengine. >>>>>> >>>>>> Maybe you can reach out to the author of the tutorial. You need to >>>>>> understand how the individual components work together in the end to be >>>>>> able to debug it. Maybe somebody else figured out on this list how to do it >>>>>> for an updated Kamailio. >>>>>> >>>>>> Cheers, >>>>>> >>>>>> Henning >>>>>> >>>>>> >>>>>> -- >>>>>> Henning Westerholt - https://skalatan.de/blog/ >>>>>> Kamailio services - https://gilawa.com >>>>>> >>>>>> >>>>>> >>>>>> > -----Original Message----- >>>>>> > From: luis.ramost--- via sr-users <sr-users(a)lists.kamailio.org> >>>>>> > Sent: Montag, 16. September 2024 02:14 >>>>>> > To: sr-users(a)lists.kamailio.org >>>>>> > Cc: luis.ramost(a)pucp.edu.pe >>>>>> > Subject: [SR-Users] About error running PCSCF >>>>>> > >>>>>> > Hi community: >>>>>> > >>>>>> > I am trying to run the Kamailio IMS, but I have the next error in >>>>>> the module >>>>>> > PCSCF >>>>>> > >>>>>> > I installed the Kamailio on a VM Machine, with UBuntu 18.04 >>>>>> > >>>>>> > Commands: >>>>>> > $ sudo -i >>>>>> > # mkdir -p /var/run/kamailio_pcscf >>>>>> > #kamailio -f /etc/kamailio_pcscf/kamailio_pcscf.cfg -P >>>>>> /kamailio_pcscf.pid -DD >>>>>> > -E -e >>>>>> > >>>>>> > Output: >>>>>> > loading modules under config path: >>>>>> > >>>>>> /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kama >>>>>> > ilio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux- >>>>>> > gnu/kamailio/modules/:/usr/local/lib64/kamailio/modules >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <delete_delay> of type <2:int> not found in module >>>>>> > <ims_registrar_pcscf> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 399, column 51: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <ipsec_preferred_ealg> of type <1:string> not found in >>>>>> module >>>>>> > <ims_ipsec_pcscf> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 417, column 59: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <recv_mode> of type <2:int> not found in module <ims_qos> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 435, column 35: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <dialog_direction> of type <2:int> not found in module >>>>>> <ims_qos> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 436, column 42: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array >>>>>> size 16 >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 260, column 42: >>>>>> unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_create (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 264, column 77: >>>>>> unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 114, column 31: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 180, column 33: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 11, column 34: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 99, column 42: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > ERROR: bad config file (10 errors) >>>>>> > 0(470) WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): >>>>>> > WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: >>>>>> > cdp_avp_destroy(226) >>>>>> > 0(470) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child >>>>>> stopping >>>>>> > ... >>>>>> > 0(470) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer >>>>>> child >>>>>> > stopped >>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: >>>>>> > mod_destroy(): Error destroying spi generator >>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:313]: >>>>>> > mod_destroy(): Error destroying port generator >>>>>> > >>>>>> > I guided form this tutorial: >>>>>> https://ryantheelder.github.io/blog/2023/VoLTE/ >>>>>> > >>>>>> > Please if somebody would help me, I appreciate it >>>>>> > >>>>>> > Finally my objective is to connect with srsRAN and open5GS to have >>>>>> VoLTE >>>>>> > service. >>>>>> > >>>>>> > Best wishes, >>>>>> > Luis >>>>>> > __________________________________________________________ >>>>>> > Kamailio - Users Mailing List - Non Commercial Discussions To >>>>>> unsubscribe >>>>>> > send an email to sr-users-leave(a)lists.kamailio.org >>>>>> > Important: keep the mailing list in the recipients, do not reply >>>>>> only to the >>>>>> > sender! >>>>>> > Edit mailing list options or unsubscribe: >>>>>> >>>>> __________________________________________________________ >>>>> Kamailio - Users Mailing List - Non Commercial Discussions >>>>> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >>>>> Important: keep the mailing list in the recipients, do not reply only >>>>> to the sender! >>>>> Edit mailing list options or unsubscribe: >>>>> >>>>

1 0

Kamailio not listening to SIP despite setup OK
by Sergio Charrua 21 Sep '24

21 Sep '24

Hi all! got a client with 2 Kamailio servers using Pacemaker/Corosync with a VIP. Kamailio is setup with dispatcher, and has been working since 2018. This previous week, client migrated from a Cisco ASA firewall to a Fortinet F5 firewall. One of the Kamailio server is, apparently, working fine, receiving SIP invites and processing them correctly. When moving the VIP to the second Kamailio node, Kamailio is ignoring any SIP message that is not OPTIONS, despite configuration (kamailio.cfg) being exactly the same as in the node #1. Also, note that using SNGrep, we can see clearly that SIP messages are reaching the server, but Kamailio is just not processing any INVITE. In fact, I added an output to log file right at the very beginning of the main route and the line is added to log file ONLY if the SIP message is OPTIONS. When the SIP message is INVITE, nothing is processed on Kamailio's side. The listen parameter is set to 0.0.0.0 port 5060 and advertised IP set to the Public IP. Again, the main route is only : route{ xlog(. ......); } Kamailio version is 5.1.6 (old, I know....) but it has been working since Dec 2018! What could be the issue?! Why only OPTIONS are being , somehow, recognized by Kamailio and all other SIP messages just ignored? Also, to add some more weirdness to the mess, connecting an Asterisk server on the same network, and sending a call to Kamailio node #2, it works!! SIP messages of any kind are processed correctly by Kamailio on node #2. So, why doesn't it work with SIP messages received via Public IP address, even though the messages reach the server correctly (confirmed using SNGrep) ??? I spent 5h this afternoon trying to understand what is wrong, but I'm stuck .... and no clue so far, apart from suspecting of the new Gateway, which I have no access to configurations.... but if OPTIONS are received & processed OK, it wouldn't make sense that received INVITES would be ignored by Kamailio, right? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

6 7

Handling SIP Contact Header Changes and ACK Routing Issues
by XPTO 21 Sep '24

21 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: - *Asterisk (internal)* → *Kamailio (Proxy)* → *SIP Provider (PSTN)* Both Asterisk and Kamailio are under my responsibility. *Problem:* When I initiate a call from Asterisk, I use Kamailio as the *outbound proxy*. The *INVITE* is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a *200 OK*, indicating the call was answered. However, the *200 OK* received from the provider contains a modification in the *Contact* header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent *ACK* should be sent. The issue arises when Kamailio receives this *200 OK*: it overwrites the *Contact* field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the *ACK* back to Kamailio, which, instead of forwarding it to the new IP from the *Contact* in the *200 OK*, sends it to the SIP provider at the same address used for the initial *INVITE*. Since the *ACK* does not reach the IP expected by the provider (the new IP introduced in the *Contact*), the provider continues to send repeated *200 OK* responses, until the call is disconnected due to a timeout, as a valid *ACK* was never received. I looked for conceptual answers in *RFC 3261* to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. *Key Questions:* 1. *Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK?* 2. *Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE?* 3. *How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?*

2 1

Active/Active Registrar Design
by Knserbrave 20 Sep '24

20 Sep '24

Hi Community, Is it a common experience for architects to encounter issues with active/active registrar designs? Specifically, when clients are the UAS of a dialog? Some user agents accept INVITEs from any source once registered while others are not as accepting. There are ways around this using 302 redirect, but there seem to also be ways per RFC to signal trusted nodes to clients. RFC 3608 seems to be an excellent way to address this issue by communicating trustworthy nodes to the client upon registration. However, the Service-Route header seems to give options to the client rather than communicate trust. No where in the RFC does trust seem to be mentioned in its design, however one could imply it should be deduced by the presence of a Service-Route header. Nonetheless, even if it should communicate trust, it doesn't seem to do so in all cases so the more logical approach to active/active is signal via a 302 redirect where the registration lives, handling INVITE transactions on the server-side instead of client-side. I'm curious to hear if there is any advice on from the community overcoming trust issues with user agents in an active/active registrar design.

3 3

sni:uknown error in logs
by Mahmood Alkhalil 20 Sep '24

20 Sep '24

Hello Kamailio! I'm etting up a kamailio server where it will receive STIP TLS connections from Zoom. kamailio is closing TLS connections with error stating "SSL routines::no shared cipher (sni: unknown)" as below Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: tls [tls_server.c:270]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7fbcd1e9dac8 ctx 0x7fbcd2229258 sn []) Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: tls [tls_domain.c:1018]: tls_server_name_cb(): SSL_get_servername returned NULL: return SSL_TLSEXT_ERR_NOACK Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: <core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending... Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: <core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c= 0x7fbcd3cb85d0 n=7 fd=8 Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: <core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf= Sep 18 13:28:02 dalia kamailio[18529]: [3B blob data] Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) ERROR: tls [tls_server.c:1312]: tls_h_read_f(): protocol level error Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) ERROR: tls [tls_util.h:49]: tls_err_ret(): TLS accept:error:0A0000C1:SSL routines::no shared cipher (sni: unknown) did a tcpdump trace to check the ciphers Zoom are using in the TLS client hello, and there are 4 and are supported by openssl on TLSv1.2, BUT the reis no server_name extension in the client hello. is this related to kamailio refusing the connection because there is no server_name in the client hello or something else?, if yes can it be forced to accept TLS connection without server_name specified ? my tls.cfg file is below [server:default] method = TLSv1.2 verify_certificate = no require_certificate = no private_key = /etc/kamailio/key.pem certificate = /etc/kamailio/certificate.pem ca_list = /etc/ssl/certs/ca-certificates.crt ca_path = /etc/ssl/certs [client:default] method = TLSv1.2+ verify_certificate = no require_certificate = no

2 3

Accessing User-Agent from usrloc at runtime.
by Knserbrave 19 Sep '24

19 Sep '24

Hi Kamailio community. There are a few fields in the usrloc cache I am interested in accessing at runtime to influence routing decisions. I can use SQLops to achieve the same functionality, however I'm interested in minimizing DB operations. I would like to know if there is a generally accepted approach to retrieving this information from the memory-hosted usrloc cache rather than dipping into the database.

3 6

Re: Handling SIP Contact Header Changes and ACK Routing Issues
by Knserbrave 19 Sep '24

19 Sep '24

2 1

String Transformation to evaluate first part of header as key=value parameter?
by Benoit Panizzon 19 Sep '24

19 Sep '24

Hi List I have to parse a customer header: P-CH-Origin: intl=example.com;foo=bar;boo=whatever I'm interested in the value of the intl parameter. Unfortunately $(hdr(P-CH-Origin){param.value,intl}) does not seem to work, if this is the first parameter. Is there a built in function to do this? 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 ______________________________________________________

2 2

How to remove my email from Mailing list
by me.projects＠yahoo.com 19 Sep '24

19 Sep '24

Hi, I want to remove my email from mailing list. Thanks, ForTheGuru

2 1

Being added to the email marketing database of a 'kamailio developing' company after posting to list?
by Benoit Panizzon 18 Sep '24

18 Sep '24

Hi Today I received two marketing emails inviting me to the UCX Expo in London from I company I was not aware having any business relationship with. After searching for that company name in my email archive, I stumbled over a post from them in this mailing list. Just out of curiosity, have others on this list also received this invitation and in the past more marketing emails from the same source? I guess posting to this list does not imply, I have given permission to reader of this list to store my personal data in a marketing database of their employer and use my data for marketing purposes, right? 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

Handling SIP Contact Header Changes and ACK Routing Issues
by Josimar Rocha 17 Sep '24

17 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: Asterisk (internal) → Kamailio (Proxy) → SIP Provider (PSTN) Both Asterisk and Kamailio are under my responsibility. Problem: When I initiate a call from Asterisk, I use Kamailio as the outbound proxy. The INVITE is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a 200 OK, indicating the call was answered. However, the 200 OK received from the provider contains a modification in the Contact header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent ACK should be sent. The issue arises when Kamailio receives this 200 OK: it overwrites the Contact field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the ACK back to Kamailio, which, instead of forwarding it to the new IP from the Contact in the 200 OK, sends it to the SIP provider at the same address used for the initial INVITE. Since the ACK does not reach the IP expected by the provider (the new IP introduced in the Contact), the provider continues to send repeated 200 OK responses, until the call is disconnected due to a timeout, as a valid ACK was never received. I looked for conceptual answers in RFC 3261 to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. Key Questions: Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK? Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE? How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?

1 0

Handling SIP Contact Header Changes and ACK Routing Issues
by Josimar Rocha 17 Sep '24

17 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: Asterisk (internal) → Kamailio (Proxy) → SIP Provider (PSTN) Both Asterisk and Kamailio are under my responsibility. Problem: When I initiate a call from Asterisk, I use Kamailio as the outbound proxy. The INVITE is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a 200 OK, indicating the call was answered. However, the 200 OK received from the provider contains a modification in the Contact header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent ACK should be sent. The issue arises when Kamailio receives this 200 OK: it overwrites the Contact field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the ACK back to Kamailio, which, instead of forwarding it to the new IP from the Contact in the 200 OK, sends it to the SIP provider at the same address used for the initial INVITE. Since the ACK does not reach the IP expected by the provider (the new IP introduced in the Contact), the provider continues to send repeated 200 OK responses, until the call is disconnected due to a timeout, as a valid ACK was never received. I looked for conceptual answers in RFC 3261 to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. Key Questions: *Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK? *Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE? *How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?

1 0

About error running PCSCF
by luis.ramost＠pucp.edu.pe 16 Sep '24

16 Sep '24

Hi community: I am trying to run the Kamailio IMS, but I have the next error in the module PCSCF I installed the Kamailio on a VM Machine, with UBuntu 18.04 Commands: $ sudo -i # mkdir -p /var/run/kamailio_pcscf #kamailio -f /etc/kamailio_pcscf/kamailio_pcscf.cfg -P /kamailio_pcscf.pid -DD -E -e Output: loading modules under config path: /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/modules 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <delete_delay> of type <2:int> not found in module <ims_registrar_pcscf> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 399, column 51: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <ipsec_preferred_ealg> of type <1:string> not found in module <ims_ipsec_pcscf> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 417, column 59: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <recv_mode> of type <2:int> not found in module <ims_qos> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 435, column 35: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <dialog_direction> of type <2:int> not found in module <ims_qos> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 436, column 42: Can't set module parameter 0(470) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/register.cfg, line 260, column 42: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_create (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/register.cfg, line 264, column 77: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mo.cfg, line 114, column 31: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mo.cfg, line 180, column 33: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mt.cfg, line 11, column 34: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mt.cfg, line 99, column 42: unknown command, missing loadmodule? ERROR: bad config file (10 errors) 0(470) WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) 0(470) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... 0(470) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): Error destroying spi generator 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:313]: mod_destroy(): Error destroying port generator I guided form this tutorial: https://ryantheelder.github.io/blog/2023/VoLTE/ Please if somebody would help me, I appreciate it Finally my objective is to connect with srsRAN and open5GS to have VoLTE service. Best wishes, Luis

4 3

[Dialog module] Dialog Flag Removal and recommendation of dlg_manage()
by Xenofon Karamanos 16 Sep '24

16 Sep '24

Hello all, We would like to inform you about a recent PR<https://github.com/kamailio/kamailio/pull/3961> (https://github.com/kamailio/kamailio/pull/3961) that introduces changes related to the dialog flag handling. Summary of Changes: * Removal of dlg_flag: The dlg_flag has been removed from the dialog module. * Replace with call of dlg_manage(): We recommend using the function dlg_manage() as a replacement to handle the necessary functionality. Please refer to the updated documentation for more details. Affected Modules: 1. These modules only required updated documentation that references the use of dlg_flag instead dlg_manage(). * acc * qos * pua_dialoginfo * sst 1. Usage Changes (Dialog Flag Handling Removed): In these modules, the code related to reading and checking the dialog flag has been removed. Additionally, the setting of the message internal flag has been removed. If you need to track the dialog or call state, you MUST now call dlg_manage() explicitly in the kamailio config * call_control * cnxcc * mediaproxy * nat_traversal Please review the updated documentation for further guidance on the use of dlg_manage(). Feel free to reach out with any questions, concerns and report any issue due to this change. Best regards, Xenofon Karamanos

1 0

rtpengine module
by Daniel Donoghue 16 Sep '24

16 Sep '24

Hello, I am using rtpengine to encrypt and decrypt DTLS and SDES audio streams for my calls. I have a set of rtpengine instances running, configured using a database, such that it is easy to scale up on demand. This works well for initiating a call and balancing load across multiple instances but I want to be able to recover from rtpengine failure. I have looked at configuring rtpengine with a redis service but do not believe that to be helpful in this case as im using encrypted media and because the configuration of rtpengine with redis has a lot of overheads, limits scaling and requires non standard configuration of a redis cluster. I would like to be able to re-anchor media streams instead of relying on rtpengine to synchronise itself across a redis cluster. For this to work, i believe i would have to use the uac module to send a reINVITE and I speculate that i would need to use an xhttp event to trigger this remotely when a failure is detected. I am unclear, though, how to get to a specific call within kamailio from the xhttp route in order to do anything to it, if i even need to, and i am nuclear how to make the rtpengine module select a new node for sdp for the reinvite? How can i achieve my goal? Perhaps it would be possible to modify the rtpengine module itself to support generating a reinvite and selecting a new node on receipt of an rpc request? Hoping someone can help me find a way forward With every blessing, Daniel

3 6

About Rx AAR in PCSCF for mobile terminated call
by Liu Wing Hong 16 Sep '24

16 Sep '24

Hi, I am using kamailio 5.8.3 as an IMS-PCSCF. On handling SIP INVITE (and response), it sends Rx AAR to PCRF for bearer establishment. For mobile-terminated call, it was found that the call was dropped by handset within several seconds after ringing for some handsets. The corresponding pcap capture is attached for reference. From the pcap capture, it was found that in the Rx AAR request sent (packet 11), in the Media-Sub-Component Flow-Description, the "permit out from ip/port" and "permit in to ip/port" were set to 202.0.169.182 port 24428 instead of 10.51.0.15 port 31124. 202.0.169.182 port 24428 is the one in the incoming INVITE sdp from S-CSCF (packet 2); 10.51.0.15 port 31124 is the one in the processed INVITE sdp sent to UE (packet 5); Also, the sdp data of "offer" inside AAR was set to the sdp in the incoming INVITE from S-CSCF instead of the sdp in the processed INVITE sent to UE. Is that correct? Please advice. Thank you. Regards, Hong

1 0

Setting $du to websocket-secure
by James Browne 13 Sep '24

13 Sep '24

How can I set the destination URI for an INVITE to be a websocket-secure destination? Is it possible? Summary I've a proxy with tcp_connection_match=1, but websocket URIs always have transport=ws (never transport=wss) in them, so relaying a call to a WSS connection always fails. I tested running kamailio 6.0.0-dev2 compiled from a commit made this week. This proxy server uses nathelper rather than outbound module. Detail We know that "transport=ws" is used for both WS and WSS. I've a proxy server that receives an INVITE for a WSS destination, and this proxy supports both WS and WSS. This proxy server must have core parameter tcp_connection_match=1 set, and this leads the t_relay() to fail. When an INVITE comes, these are the steps. - The URI is something like sip:user@anonymous.invalid;alias=198.51.100.10~52833~6;transport=ws. - First handle_ruri_alias() removes the alias (which has ~6 in it, for wss) and sets the $du to something like sip:198.51.100.10:52833;transport=ws. - Then loose_route_preloaded() processes the Route header fields and forces the outbound socket to the TLS websocket one. - Then t_relay() fails to relay the INVITE and responds with 477 or 500. If, however, there's a non-TLS websocket connection open to the proxy, the INVITE would be erroneously relayed over that (using the wrong kamailio-side TCP port). I can go deeper with testing if required. I wonder whether this is a bug. James

3 12

Strange result: $stat(total_size) returns 0
by whliu＠citictel.com 12 Sep '24

12 Sep '24

Hi, I am using kamailio 5.8.3. I just found that strange result is returned from variable $stat(total_size). In the example script under "misc/examples/ims/pcscf/kamailio.cfg", there is a line, $var(used) = 1 - ($stat(free_size) / $stat(total_size)); The trace log said that $stat(total_size) was returning 0, and got a log like this ERROR: <core> [core/rvalue.c:1434]: long_longop2(): rv div by 0 Any idea on this? Please advise. Thank you. Regards, Hong

1 0

Strange result: $stat(total_size) returns 0
by whliu＠citictel.com 12 Sep '24

12 Sep '24

Hi, I am using kamailio 5.8.3. I just found that strange result is returned from variable $stat(total_size). In the example script under "misc/examples/ims/pcscf/kamailio.cfg", there is a line, $var(used) = 1 - ($stat(free_size) / $stat(total_size)); The trace log said that $stat(total_size) was returning 0, and got a log like this ERROR: <core> [core/rvalue.c:1434]: long_longop2(): rv div by 0 Any idea on this? Please advise. Thank you. Regards, Hong

1 0

SIP parse error for self-generated Websocket reply
by Sebastian Damm 12 Sep '24

12 Sep '24

Hello, We try to introduce a small websocket sidecar to our Kamailio servers, which runs on the same host. It connects to the regular TCP socket of our Kamailios, and in the XHTTP request route, we do just this: if path == "/sidecar": KSR.websocket.handle_handshake() return 1 On every new connect, Kamailio logs a parse error for the reply it generates. Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 22) Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols Sia: SIP/2.0/TCP 100.74.11.1:46422 Sec-WebSocket-Protocol: sip Upgrade: websocket Connection: upgrade Sec-WebSocket-Accept: 60dPbzcRW6up7z92kDlxkqL2WlY= Content-Length: 0 > Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/msg_translator.c:3402]: build_sip_msg_from_buf(): parsing failed I saw this error in other questions from a while ago, but without a solution. What could be the reason for this? Thanks in advance, Sebastian

2 2

Suspected memory leak in module ims_registrar_pcscf
by whliu＠citictel.com 09 Sep '24

09 Sep '24

Hi, I just tried to send numbers of SIP register to kamailio (release 5.8.2). It is reported that there are some memory leakage in shared memory after the expiry of all the registration sessions. After drill down the code. function save_pending(...) is called. (in ims_registrar_pcscf/save.c). Inside that function, it seems that "sec_verify_params" is not deallocated upon the function completes. Findings: - sec_verify_params uses the result from function cscf_get_security_verify(...) - Function cscf_get_security_verify(...) returns the result of function parse_sec_agree(...) - Function parse_sec_agree(...) returns "params" where "shm_malloc(...)" is called It seems that the memory pointed by 'sec_verify_params' is not further referenced by others. Should shm_free(...) be called when leaving function save_pending(...) in order to free the unused memory of 'sec_verify_params'? Please advice. Thank you. Regards, Hong

2 2

RFC: Support for icc or suncc compilers
by Daniel-Constantin Mierla 09 Sep '24

09 Sep '24

Hello, I was wondering if icc or suncc compilers are still used in the community, and by that if it still worth trying to support them via Makefiles and compile time flags (defines). Personally I haven't had to deal with them for more than 10-15 years, I have no access with a system offering them. Furthermore, I am not even sure if the code compiles with them. Maybe someone (or many) in the community use one or the other and confirm that they still worth to be kept. If not, maybe we can consider them for removing, it can be a good task to be done during Kamailio development meeting planned in Dusseldorf, Germany during November. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, October 8-10, 2024 -- asipto.com

1 0

Problem on uac_req via port
by whliu＠citictel.com 09 Sep '24

09 Sep '24

Hi, I just tried to use kamailio 5.8.3 as an IMS-PCSCF. After handling SIP registration, it seems that value of port in the "via" header in the request sent from PCSCF to UE is a bit strange. When using uac_req to send OPTIONS to UE, it is found that in the OPTIONS sent out from kamailio, the value of port inside the "via" header is set to port_pc (pcscf ipsec client port). Is this correct? Or should the via header port be set to port_ps (pcscf ipsec server port) instead? Also, in uac_req, is the value of via port header configurable? Please advise. Thank you. Regards, Hong

1 0

Error when passing task to an async task group
by elhar.mohamed＠gmail.com 06 Sep '24

06 Sep '24

Hello, I'm using async task group to handle some pourcentage of the traffic, while testing with traffic of 1200cps I got the following error : ERROR: {1910812149 INVITE 6785db25bc3f2478@ip} <core> [core/async_task.c:439]: async_task_group_push(): failed to pass the task [0x7f4000c3a5c0] to group [async_group_name]. conf: async_workers_group="name=async_group_name;workers=45;nonblock=1;usleep=100" If someone can help please?

1 0

Disable topos for a specific peering IP
by Denys Pozniak 06 Sep '24

06 Sep '24

Hello! I need to disable topos for one specific SIP trunk (in-out), it looks like it’s enough to use event_route with IP address filtering. But for some reason, the incoming INVITE from the peer still gets processed by topos and I also don’t see a mention of [msg-incoming] in the logs, only this: WARNING: <script>: [msg-outgoing] OPTIONS/you/1.1.1.1 Code snippet: loadmodule "topos.so" modparam("topos", "db_url", DBURL) modparam("topos", "contact_mode", 1) modparam("topos", "header_mode", 1) modparam("topos", "methods_noinitial", "OPTIONS,SUBSCRIBE,PUBLISH") modparam("topos", "dialog_expire", 7210) modparam("topos", "rr_update", 1) modparam("topos", "event_mode", 5) /* 1 - execute event_route[topos:msg-outgoing] 2 - execute event_route[topos:msg-sending] 4 - execute event_route[topos:msg-incoming] 8 - execute event_route[topos:msg-receiving] */ request_route { .... event_route[topos:msg-outgoing] { if ( $sndto(ip) == "1.1.1.1" ) { xlog("L_WARN","[msg-outgoing] $rm/$rU/$sndto(ip) \n"); drop; } } } event_route[topos:msg-incoming] { if ( $si == "1.1.1.1" ) { xlog("L_WARN","[msg-incoming] $rm/$rU/$si \n"); drop; } } # kamailio -v version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty -- BR, Denys Pozniak

6 9

t_check_trans in Lua
by nizam＠ctapps.com 06 Sep '24

06 Sep '24

Dear experts, Would appreciate if anyone can explain why in the given sample Lua script the 'Handle Retransmission' block checks the return code of t_check_trans (if KSR.tm.t_check_trans == 0 then), but in the Kamailio native config standard file if just calls the t_check_trans() function without checking the return code. What is the intention here and how does it work? Thank you in advance. Handle retransmission script from native config file: if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } Handle retransmission from sample Lua script: if not KSR.is_ACK() then if KSR.tmx.t_precheck_trans() > 0 then KSR.tm.t_check_trans(); return 1; end if KSR.tm.t_check_trans() == 0 then return 1; end end

2 2

secsipid and secsipid_proc module RPM build
by Володимир Іванець 04 Sep '24

04 Sep '24

Hello! I was wondering if someone was trying to build Kamailio RPM packages containing modules secsipid and secsipid_proc and faced the same problem. I compiled secsipidx library and copied secsipid.h, libsecsipid.h as well as libsecsipid.a files into the source directory of secsipid_proc module. And when I'm trying to build RPM packages, I get the following error: CC (gcc) [M secsipid_proc.so] secsipid_proc_mod.o secsipid_proc_mod.c:28:10: fatal error: secsipid.h: No such file or directory #include <secsipid.h> ^~~~~~~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: secsipid_proc_mod.o] Error 1 make[1]: *** [Makefile:508: modules] Error 1 make[1]: Leaving directory '/builddir/build/BUILD/kamailio-5.8.2/src' make: *** [Makefile:34: every-module] Error 2 Thanks a lot!

2 4

Kamailio v5.8.3 Released
by Daniel-Constantin Mierla 04 Sep '24

04 Sep '24

Hello, Kamailio SIP Server v5.8.3 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.3. For more details about version 5.8.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/09/kamailio-v5-8-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, October 8-10, 2024 -- asipto.com

1 0

Permissions module - automatic reload
by Sergio Charrua 04 Sep '24

04 Sep '24

Hi all, Just need some clarifications about the Permissions module: I currently use it with db_mode = 1 and was expecting that the parameter "trusted_cleanup_interval" would refresh the cache tables every X seconds, but apparently this is not the case. I was looking for a way for Kamailio to automatically reload the address table into memory every 5 minutes, for example, and I (mis)understood that "trusted_cleanup_interval" would do the trick, but it doesn't work as expected. Question: is there any command, besides kamcmd, to automatically reload the permissions table? I would like to avoid having to add a line in the crontab.... Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

1 0

Using a different OpenSSL
by David Cunningham 04 Sep '24

04 Sep '24

Hello, We have compiled openssl 3.0.9 from source because it's FIPS validated, and want to use it with Kamailio. The server also has the Ubuntu openssl 3.0.2 package installed. Does anyone know how we can tell Kamailio to use the openssl library in /opt/openssl/lib64, and how we can verify that it really is using it? Thanking you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

5 13

Releasing v5.8.3
by Daniel-Constantin Mierla 04 Sep '24

04 Sep '24

Hello, I am considering to release Kamailio v5.8.3 (out of branch 5.8) sometime next week, most likely on Wednesday, Sep 4, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

kamailio stri shaken configuration
by Vishal Pai 02 Sep '24

02 Sep '24

Hello All I need to integrate the kamailio with stri shaken with pem. What is the best way to achieve this? Thanks

3 2

app_mono module to be archived
by Victor Seva 31 Aug '24

31 Aug '24

Hi kamilians, we have the intention to archive the app_mono module in a few[0]. Most likely at the end of next week. If you use app_mono and you don't want to see it fade away now it's the time to rise your voice. Cheers, Victor [0] https://github.com/kamailio/kamailio/pull/3964

1 0

How to restrict Kamailio adding Local VIA header
by panda.paresh＠gmail.com 30 Aug '24

30 Aug '24

Hi All, I am trying to replay a request to the destination via Kamailio, and I am using t_relay_to_tcp for that , but in that case Kamailio is adding its own address as the top VIA header. Is there a way I can instruct Kamailio not to add local via header ?

3 4

http_async and tm
by alexis 30 Aug '24

30 Aug '24

Hello all, context first, we have an REST API that performs queries to external devices in the network (diameter to DRA's, REST to different servers) and based on n conditions returns the content for a Contact header to be used in a SIP 302. Now we're consuming this API with http_client (synchronously) and as there's no way to speed up the API (pipeline executions, delays on external api's etc etc) we're hitting a limit where all children become busy waiting for the API to answer. So i decided to move to http_async_client and started working on it on the lab with this first and base concept to test. request_route { #for testing purposes only if(is_method("ACK")){ exit; } $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = 500; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string", "event", "sip-routing", "$var(cre_query)"); xlog("L_INFO","API ASYNC ROUTING REQUEST: $var(cre_query)\n"); $http_req(body) = $var(cre_query); t_newtran(); http_async_query("http://192.168.86.128:8000/", "CRE_RESPONSE"); } http://192.168.86.128:8000/ receives the POST, randomly creates a delay between 0.5 and 1 second and responds (simulating the real api with an excess delay to probe the concept) Then route[CRE_RESPONSE] { if ($http_ok && $http_rs == 200) { xlog("L_INFO","CRE RESPONSE: $http_rb\n"); # for testing purpose, Contact content will be replaced from the received api response append_to_reply("Contact: <sip:1234@google.com>\r\n"); send_reply(302,"Moved Temporarily"); exit; } send_reply(500, "Internal error"); exit; } INVITE is received and processed, API is called, after API response, 302 is replied and then an ACK (ignored by now). Situation is that the 302 retransmitted 37 1519.846253067 192.168.86.34 → 192.168.86.128 SIP/SDP 585 Request: INVITE sip:service@192.168.86.128:5060 | 38 1519.848100380 192.168.86.128 → 192.168.86.34 SIP 318 Status: 100 Trying | 39 1520.094997642 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 40 1520.102323728 192.168.86.34 → 192.168.86.128 SIP 453 Request: ACK sip:service@192.168.86.128:5060 | 41 1520.591300933 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 42 1521.591061065 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 43 1523.591227956 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [timer.c:634]: wait_handler(): finished transaction: 0x7f6d75184cc8 (p:0x7f6d74f600c8/n:0x7f6d74f600c8) 18(24) DEBUG: tm [h_table.c:132]: free_cell_helper(): freeing transaction 0x7f6d75184cc8 from timer.c:643 Any help to avoid the retransmission and make the transaction just finish right after the 302 will be appreciated. regards

6 27

apply keepalive to only udp
by James Lipski 30 Aug '24

30 Aug '24

Hello, I wanted to know if this can be accomplished, or if I'm doing something wrong. What I'm hoping to accomplish is to have kamailio periodically send OPTIONS packets to endpoints that are connected over UDP only. As of right now, I currently have the following configured under usrloc: modparam("usrloc", "timer_interval", 5) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 0) modparam("usrloc", "ka_mode", 1) modparam("usrloc", "ka_method", "OPTIONS") modparam("usrloc", "ka_from", "sip:ping@sip-domain.com") modparam("usrloc", "ka_domain", "sip-domain.com") modparam("usrloc", "ka_timeout", 125) modparam("usrloc", "ka_interval", 60) With these parameters, all endpoints will be sent OPTIONS packets periodically as expected. When setting "ka_mode" to "4", as expected, endpoints registered with UDP only receive OPTIONS packets; which is what I wanted. The issue that I'm having with these parameters is that endpoints registered with anything other that UDP are triggering a contact-expired event after a little over 2 minutes. I have the registrar set to 3600 as an expiry; I'm assuming the contact is expiring due to the "ka_timeout" value. -------------------------- Is there a way to only have keep-alive packets/ and timeouts apply to endpoints registered on UDP? Thank you.

3 5

Kamailio integration with REST services
by Sergio Charrua 29 Aug '24

29 Aug '24

Hi all! I need to integrate a REST Service with Kamailio. Integration is easy, not a problem, but my concern is that the REST Services has a JWT token authentication required for each request, common to many (it not all) REST Services. So, it is required to POST for authentication (username and password) and once I get authentication done, and receive a JWT token, I will need to use this token in all other requests to the REST Service. The token is valid for 24h. My question is: - how to save the token for use during 24h, on all calls/sessions, without the need to authenticate on every call? - how to detect the token has expired and re-authenticate? I looked for some modules, but could not find any that suited me. I am thinking of executing a Python script for this, but concerned about latency and PDD.... Any thoughts? Any suggestions? Any known best-practice? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

6 12

kamailio randomly crashes
by Maharaja Azhagiah 28 Aug '24

28 Aug '24

Hi all, I have kamailio running in GKE (Google Kubernetes) and recently seeing kamailio crashes frequently and I couldn't find the reason for the crash kamailio version version: kamailio 5.4.9 (x86_64/linux) 7d022d 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: 7d022d compiled on 14:26:30 Jul 10 2024 with gcc 8.3.0 kamailio core generation: Core was generated by `kamailio -e -E -DD -M 128 -m 2048'. Program terminated with signal SIGSEGV, Segmentation fault. #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 416 if (rd->carriers[i]->domains[j]->sum_prob == 0.0) { bt: (gdb) bt #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 #1 0x00007ca474c8cc50 in reload_route_data () at cr_data.c:178 #2 0x00007ca474cabfdc in cr_rpc_reload_routes (rpc=0x7ca4f5a32aa0 <func_param>, c=0x7ca4f5a329c0 <_jsonrpc_ctx_global>) at cr_rpc.c:59 #3 0x00007ca4f5a158a1 in jsonrpc_exec_ex (cmd=0x7ffd774f34b0, rpath=0x0) at jsonrpcs_mod.c:1425 #4 0x00007ca4f5a15ccd in jsonrpc_exec (msg=0x7ca4f641a9f8, cmd=0x7ca4f61bde20 "x\335\033\366\244|", s2=0x0) at jsonrpcs_mod.c:1445 #5 0x00005745eb3fcb19 in do_action (h=0x7ffd774f3dc0, a=0x7ca4f61f1a50, msg=0x7ca4f641a9f8) at core/action.c:1085 #6 0x00005745eb40b35c in run_actions (h=0x7ffd774f3dc0, a=0x7ca4f61f0468, msg=0x7ca4f641a9f8) at core/action.c:1584 #7 0x00007ca474b92f8d in timer_handler (ticks=96802941, tl=0x7ca47b2f7950, data=0x7ca47b2f7868) at timer.c:216 #8 0x00005745eb49e3c4 in slow_timer_main () at core/timer.c:1105 #9 0x00005745eb3a85bd in main_loop () at main.c:1765 #10 0x00005745eb3b34c3 in main (argc=8, argv=0x7ffd774f4688) at main.c:2942 bt full: (gdb) bt full #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 res = 0x7ca4f636a4a8 row = 0x7ca4f642a7f8 i = 0 j = 110 ret = 56 tmp_carrier_data = 0x7ca479ff5068 query_str = {s = 0x7ca474cdd800 <query> "SELECT DISTINCT domain FROM carrierroute WHERE carrier=8", len = 56} tmp_scan_prefix = {s = 0x5745ed889f68 "13235791374", len = 11} tmp_rewrite_host = {s = 0x5745ed889f7a "sip.telnyx.com;transport=TLS", len = 28} tmp_rewrite_prefix = {s = 0x5745ed889f99 "", len = 0} tmp_rewrite_suffix = {s = 0x5745ed889f9a "", len = 0} tmp_host_name = {s = 0x40 <error: Cannot access memory at address 0x40>, len = 1965559808} tmp_reply_code = {s = 0x7ca474ccf2b0 <__func__.5843> "reload_route_data", len = 1959587872} tmp_comment = {s = 0x5745ed889f9b "IMPORTED - WTJHSX", len = 18} p_tmp_comment = 0x7ffd774f30b0 __func__ = "load_route_data_db" n = 0 #1 0x00007ca474c8cc50 in reload_route_data () at cr_data.c:178 old_data = 0x5745ed850150 new_data = 0x7ca482a48840 i = 22341 __func__ = "reload_route_data" #2 0x00007ca474cabfdc in cr_rpc_reload_routes (rpc=0x7ca4f5a32aa0 <func_param>, c=0x7ca4f5a329c0 <_jsonrpc_ctx_global>) at cr_rpc.c:59 __func__ = "cr_rpc_reload_routes" #3 0x00007ca4f5a158a1 in jsonrpc_exec_ex (cmd=0x7ffd774f34b0, rpath=0x0) at jsonrpcs_mod.c:1425 rpce = 0x7ca4f6425fb8 ctx = 0x7ca4f5a329c0 <_jsonrpc_ctx_global> ret = -1 nj = 0x0 val = {s = 0x5745ed845010 "cr.reload_routes", len = 16} scmd = {s = 0x7ca4f61bdd78 "{\"jsonrpc\": \"2.0\", \"method\": \"cr.reload_routes\", \"id\": 2}", len = 57} __func__ = "jsonrpc_exec_ex" #4 0x00007ca4f5a15ccd in jsonrpc_exec (msg=0x7ca4f641a9f8, cmd=0x7ca4f61bde20 "x\335\033\366\244|", s2=0x0) at jsonrpcs_mod.c:1445 scmd = {s = 0x7ca4f61bdd78 "{\"jsonrpc\": \"2.0\", \"method\": \"cr.reload_routes\", \"id\": 2}", len = 57} __func__ = "jsonrpc_exec" #5 0x00005745eb3fcb19 in do_action (h=0x7ffd774f3dc0, a=0x7ca4f61f1a50, msg=0x7ca4f641a9f8) at core/action.c:1085 ret = -5 v = 0 dst = {send_sock = 0x0, to = {s = {sa_family = 41744, sa_data = "6\366\244|\000\000\035u\271t\244|\000"}, sin = {sin_family = 41744, sin_port = 63030, sin_addr = {s_addr = 31908}, sin_zero = "\035u\271t\244|\000"}, sin6 = {sin6_family = 41744, sin6_port = 63030, sin6_flowinfo = 31908, sin6_addr = {__in6_u = {__u6_addr8 = "\035u\271t\244|\000\000\005u\271t\244|\000", __u6_addr16 = {29981, 29881, 31908, 0, 29957, 29881, 31908, 0}, __u6_addr32 = {1958311197, 31908, 1958311173, 31908}}}, sin6_scope_id = 2001680800}, sas = {ss_family = 41744, __ss_padding = "6\366\244|\000\000\035u\271t\244|\000\000\005u\271t\244|\000\000\240\071Ow\375\177\000\000/.:\365\244|\000\000\320:Ow\375\177\000\000\327\337j\353EW\000\000\320\071Ow\375\177\000\000\020@ \016\366\244|\000\000\030\245\066\366\244|\000\000\020@\016\366\244|\000\000\020:Ow\375\177\000\000\356u\236\365\244|\000\000\030\245\066\366\244|\000\000\000\000\000\000\000\000\000", __ss_align = 134217728}}, id = 130848392, send_flags = {f = 0, blst_imask = 0}, proto = 56 '8', proto_pad0 = 5 '\005', proto_pad1 = 36} tmp = 0x7ca4f641a9f8 "\344\032" new_uri = 0xed3 <error: Cannot access memory at address 0xed3> end = 0x7ca474b9751b "\r\n" crt = 0x2 <error: Cannot access memory at address 0x2> cmd = 0x7ca4f617b7d8 len = 0 user = 15 uri = {user = {s = 0x9774f3800 <error: Cannot access memory at address 0x9774f3800>, len = 621}, passwd = {s = 0x400000000 <error: Cannot access memory at address 0x400000000>, len = 1721781302}, host = {s = 0x0, len = -2147483648}, port = {s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, params = {s = 0x7ffd774f38e0 "09Ow\375\177", len = -167557169}, sip_params = { s = 0x7ca4f53704f0 <dialog_table_name> "l^5\365\244|", len = -164192016}, headers = {s = 0x1700000000 <error: Cannot access memory at address 0x1700000000>, len = 2001681040}, port_no = 14800, proto = 30543, type = 32765, flags = (unknown: 4130775280), transport = {s = 0x0, len = 2009201968}, ttl = {s = 0x6774f3a50 <error: Cannot access memory at address 0x6774f3a50>, len = -181058744}, user_param = { s = 0x17200000000 <error: Cannot access memory at address 0x17200000000>, len = 2099608304}, maddr = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -344235896}, method = {s = 0x0, len = -2147483648}, lr = { --Type <RET> for more, q to quit, c to continue without paging-- s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, r2 = {s = 0x7ffd774f38d0 "`9Ow\375\177", len = -166931844}, gr = {s = 0x7ca4f53704f0 <dialog_table_name> "l^5\365\244|", len = -164192016}, transport_val = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -181636538}, ttl_val = {s = 0x7ffd774f3930 "`9Ow\375\177", len = -167555326}, user_param_val = {s = 0x0, len = 2}, maddr_val = { s = 0x7ca4f60c1f77 <db_mysql_submit_query> "UH\211\345AWAVAUATSH\201\354\210", len = -166889584}, method_val = {s = 0x177d6038c8 <error: Cannot access memory at address 0x177d6038c8>, len = 2001681040}, lr_val = {s = 0xe00000000 <error: Cannot access memory at address 0xe00000000>, len = 1958311183}, r2_val = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -164191464}, gr_val = { s = 0x7ffd774f39f0 "\373\377\377\377", len = -344973614}} next_hop = {user = {s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, passwd = {s = 0x7ffd774f3720 "\020\070Ow\375\177", len = -166975523}, host = { s = 0x7ffd774f3740 "\020\070Ow\375\177", len = -166887566}, port = {s = 0x7ca4f60431d0 <sql_str> "@@w\355EW", len = -164192016}, params = {s = 0x7ffd774f3d50 "", len = -164192016}, sip_params = {s = 0x0, len = -30810208}, headers = {s = 0x0, len = -181051830}, port_no = 8055, proto = 62988, type = 31908, flags = (unknown: 4128077712), transport = { s = 0x4774f37d0 <error: Cannot access memory at address 0x4774f37d0>, len = 2001680544}, ttl = {s = 0x669fa379 <error: Cannot access memory at address 0x669fa379>, len = -164192016}, user_param = { s = 0x5745ed7743b0 "'sip:+13194676789@us.test.com',NULL,0)", len = 1721781302}, maddr = {s = 0x400000000 <error: Cannot access memory at address 0x400000000>, len = -167513741}, method = { s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, lr = {s = 0x7ffd774f3810 "\340\070Ow\375\177", len = -167567737}, r2 = { s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, gr = {s = 0x7ffd774f3810 "\340\070Ow\375\177", len = -166977673}, transport_val = {s = 0x7ca4f60431d0 <sql_str> "@@w\355EW", len = -164192016}, ttl_val = {s = 0x7ffd774f3a90 "\330\267\027\366\244|", len = 23}, user_param_val = { s = 0x5745ed77417a "3071,3795,'6e1a9c34-cec5-4ebb-bcd9-3356d80bcf9e',' sip:+13194004696@sip.telnyx.com','c5SS3F27B8rDN',' sip:+13194676789@us.test.com','f9b9b2c48ef41ef90eb05056b0a794','tls: 10.16.0.74:520"..., len = -164192016}, maddr_val = {s = 0x7ca4f60dcf18 "`", len = -31060352}, method_val = {s = 0x0, len = -180627954}, lr_val = {s = 0x7ca4f60c1f77 <db_mysql_submit_query> "UH\211\345AWAVAUATSH\201\354\210", len = -166889584}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} u = 0x7ca4f5817bc8 port = 0 dst_host = 0x7ca474b9751d i = -344941541 flags = 0 avp = 0x7ffd774f3c40 st = {flags = 3984015424, id = 22341, name = {n = -310010607, s = {s = 0x5745ed859d11 "", len = 2001679968}, re = 0x5745ed859d11}, avp = 0x66a04c36} sct = 0x60f60e6120 sjt = 0x5f60e6a50 rve = 0x8000000 mct = 0x0 rv = 0xffffffff rv1 = 0x5745eb7fc993 c1 = {cache_type = 10, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 0}, re = 0x0}, pval = {rs = {s = 0x0, len = 0}, ri = 2001679912, flags = 32765}}, i2s = "`6Ow\375\177\000\000\031\241\a\366\244|\000\000\250\003\000\000\000"} s = {s = 0xffffffff <error: Cannot access memory at address 0xffffffff>, len = -164179120} srevp = {0x3, 0x64870454268b4200} evp = {data = 0x0, obuf = {s = 0x0, len = 0}, rcv = 0x0, dst = 0x0, req = 0x0, rpl = 0x0, rplcode = 0, mode = 0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = { number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __func__ = "do_action" #6 0x00005745eb40b35c in run_actions (h=0x7ffd774f3dc0, a=0x7ca4f61f0468, msg=0x7ca4f641a9f8) at core/action.c:1584 t = 0x7ca4f61f1a50 ret = 1 tvb = {tv_sec = 0, tv_usec = 0} tve = {tv_sec = 0, tv_usec = 0} tz = {tz_minuteswest = 60, tz_dsttime = 0} tdiff = 4131498432 __func__ = "run_actions" #7 0x00007ca474b92f8d in timer_handler (ticks=96802941, tl=0x7ca47b2f7950, data=0x7ca47b2f7868) at timer.c:216 msg = 0x7ca4f641a9f8 a = 0x7ca47b2f7868 ra_ctx = {rec_lev = 1, run_flags = 0, last_retcode = 1, jmp_env = {{__jmpbuf = {95957815091336, 8333654393230933038, 0, 2147483648, 134217728, 95957815380371, 8333654393012829230, 2510245087605373998}, __mask_was_saved = 0, __saved_mask = {__val = {137047930071236, 0, 25985340079663656, 140726605135616, 137045787154760, 70721158768, 137045787160648, 140726605135472, 137045776334895, 1095350878208, 137045782276504, 140735195070144, 140726605135504, 137045776144708, 0, 140726605135552}}}}} keng = 0x0 evname = {s = 0x7ca474b9715a "timer", len = 5} __func__ = "timer_handler" --Type <RET> for more, q to quit, c to continue without paging-- #8 0x00005745eb49e3c4 in slow_timer_main () at core/timer.c:1105 n = 12 ret = 4294967295 tl = 0x7ca47b2f7950 i = 1010 __func__ = "slow_timer_main" #9 0x00005745eb3a85bd in main_loop () at main.c:1765 i = 16 pid = 0 si = 0x0 si_desc = "udp receiver child=15 sock=10.16.0.74:5101 ( 35.202.133.184:5101)\000AOw\375\177\000\000d\366\023\376\244|\000\000 \000\000\000\060\000\000\000@AOw\375\177\000\000\200@Ow \375\177\000\000\000B\213&T\004\207d\223\311\177\353EW\000\000\240\274\065\366\244|\000" nrprocs = 16 woneinit = 1 __func__ = "main_loop" #10 0x00005745eb3b34c3 in main (argc=8, argv=0x7ffd774f4688) at main.c:2942 cfg_stream = 0x5745ed74d2a0 c = -1 r = 0 tmp = 0x7ffd774f5d9f "" tmp_len = 0 port = 0 proto = 0 ahost = 0x0 aport = 0 options = 0x5745eb7b8d78 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1555962608 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 2 n_lst = 0x7ffd77580248 p = 0x7ca4fe189995 <handle_intel+197> "H\205\300uf\213t$\004M\215\216\244" st = {st_dev = 0, st_ino = 0, st_nlink = 0, st_mode = 0, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 0, st_blksize = 0, st_blocks = 0, st_atim = {tv_sec = 0, tv_nsec = 0}, st_mtim = {tv_sec = 0, tv_nsec = 0}, st_ctim = {tv_sec = 0, tv_nsec = 0}, __glibc_reserved = {0, 0, 0}} tbuf = "\b\000\000\000\000\000\000\000\210FOw\375\177\000\000\320FOw\375\177\000\000\330+,\376\244|\000\000\000\000\000\000\000\000\000\000\020\vH\376\244|\000\000\001", '\000' <repeats 19 times>, " \000\000\000(&G\376\244|\000\000\320EOw\375\177\000\000\312tH\376\244|\000\000\000\000\000\000\000\000\000\000@e*\376\244|\000\000\320\263+\376\244|\000\000\364EOw\375\177\000\000\006", '\000' <repeats 15 times>, "/", '\000' <repeats 39 times>, "\200\037\000\000\377\377", '\000' <repeats 186 times>... option_index = 0 long_options = {{name = 0x5745eb7baf46 "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x5745eb7b6524 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x5745eb7baf4b "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x5745eb7baf51 "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x5745eb7baf57 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x5745eb7baf60 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x5745eb7baf6a "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x5745eb7baf74 "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x5745eb7baf7f "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x5745eb7baf88 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x5745eb7baf93 "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x5745eb7baf99 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main" (gdb) list 411 } 412 } while(RES_ROW_N(res) > 0); 413 414 for (i = 0; i < rd->carrier_num; ++i) { 415 for (j = 0; j < rd->carriers[i]->domain_num; ++j) { 416 if (rd->carriers[i]->domains[j]->sum_prob == 0.0) { 417 LM_ERR("All routes with carrier id %d (%.*s) and domain id %d (%.*s) have probability 0.\n", 418 rd->carriers[i]->id, rd->carriers[i]->name->len, rd->carriers[i]->name->s, 419 rd->carriers[i]->domains[j]->id, rd->carriers[i]->domains[j]->name->len, rd->carriers[i]->domains[j]->name->s); 420 } Please help to understand the reason for the crash and how to fix the problem. Thank you in advance. Regards *Maharaja Azhagiah*

3 7

Limiting Video calls(Volte to video conversion)
by dushy 28 Aug '24

28 Aug '24

Hello All, I am trying to limiting the number of video calls (converting Volte to Vilte). To reduce traffic on the enodeb. Its like if there are 10 volte calls going on then I should be able to limit the number calls that can be converted to Vilte. I tried to write a route in kamailio_pcscf.cfg using module ims_dialog. But it doesnt work and get some errors while doing a modparam. Tried another method of counting eRAB at eNB (ENBUEID and ERABID pair with qci-2). This worked, but after sometime the extra call got established. Looks like traffic is going through qci 9 bearer. Is there any method where we can achieve this. Thanks in Advance. Regards, Dushyanth

1 0

Re: certificate verify failed (sni: unknown) integration with ms teams
by Muhammad Sohaib 27 Aug '24

27 Aug '24

Hi Batanai, After disabling #!define WITH_NAT in Kamailio, calls from the Asterisk endpoint to Microsoft Teams are functioning correctly. Both inbound and outbound calls exhibit proper two-way audio and no disconnection issues but there is no Bye when hangup from ms teams. However, I am encountering a problem with outbound calls originating from Microsoft Teams. When a call is placed from MS Teams, the recipient's phone rings as expected. Once the call is answered, Microsoft Teams disconnects the call immediately. The root of the issue seems to be that the ACK message is not being sent or received properly. As a result, Microsoft Teams does not receive the ACK, leading to an immediate call termination. Additionally, Asterisk does not receive a BYE message from Kamailio. *Summary of the Issue:* - *Inbound Calls*: From Asterisk to MS Teams work fine, with full audio and no disconnections but no BYE from ms teams. - *Outbound Calls*: From MS Teams to Asterisk result in the call being disconnected immediately after being answered due to a missing ACK, and Asterisk does not receive a BYE message. Thanks, On Tue, Aug 27, 2024 at 5:45 PM Batanai Musiiwa <batanai.musiiwa(a)etel.co.zw> wrote: > Hi @ > Muhammad. May you advise if the calls are disconnecting in either > direction? > > ; > > On Tue, Aug 27, 2024 at 11:05 AM Muhammad Sohaib via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi everyone, >> >> Thank you all for your kind help with the TLS configuration. Palany's >> suggestion worked flawlessly, and now both inbound and outbound calls are >> functioning. >> >> However, I'm facing a one-way audio issue, and calls are disconnected >> after 30 seconds. I suspect this might be related to NAT. Here’s the >> current setup: >> >> *Asterisk <===> Kamailio (as SBC) <=====> MS Teams* >> >> Could this be related to my NAT configuration, or is there something else >> I should investigate? Any guidance would be greatly appreciated. >> >> Thanks again for your support! >> >> On Thu, Aug 15, 2024 at 11:02 PM palany <palany(a)advancedzim.com> wrote: >> >>> >>> >>> Hi Muhammad >>> >>> >>> >>> Can you try your tls cfg as below and make sure your certificates have >>> the right permissions. >>> >>> >>> >>> [server:default] >>> >>> method = TLSv1.2+ >>> >>> verify_certificate = no >>> >>> require_certificate = no >>> >>> private_key = /etc/letsencrypt/live/MYDOMAIN/privkey.pem >>> >>> certificate = /etc/letsencypt/live/MYDOMAIN/fullchain.pem >>> >>> >>> >>> [client:default] >>> >>> method = TLSv1.2+ >>> >>> verify_certificate = no >>> >>> require_certificate = no >>> >>> private_key = /etc/letsencrypt/live/MYDOMAIN/privkey.pem >>> >>> certificate = /etc/letsencrypt/live/MYDOMAIN/fullchain.pem >>> >>> *From:* Muhammad Sohaib via sr-users [mailto:sr-users@lists.kamailio.org] >>> >>> *Sent:* Thursday, 15 August 2024 3:46 PM >>> *To:* Kamailio (SER) - Users Mailing List >>> *Cc:* Muhammad Sohaib >>> *Subject:* [SR-Users] certificate verify failed (sni: unknown) >>> integration with ms teams >>> >>> >>> >>> Dear all, >>> >>> >>> >>> Trying to integrate Kamailio with MS Teams by following >>> https://skalatan.de/en/blog/kamailio-sbc-teams >>> >>> >>> >>> kamcmd dispatcher.list | egrep "URI|FLAGS" >>> >>> >>> URI: sip: >>> sip.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> URI: sip: >>> sip2.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> URI: sip: >>> sip3.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> >>> >>> >>> Kamailio Logs: >>> >>> /usr/local/sbin/kamailio[412158]: INFO: <script>: Sent out tm request: >>> OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012 >>> Via: SIP/2.0/TLS >>> x.x.x.x:5061;branch=z9hG4bK5dad.92de50b2000000000000000000000000.0#015#012 >>> To: <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012 >>> From: <sip:test.mytest.com >>> >;tag=5d0939b82abe9b1bbf185d963b6e6c88-edeb3c71#015#012 >>> CSeq: 10 OPTIONS#015#012 >>> Call-ID: 2db6bede5631d9b6-412158(a)88.99.244.106#015#012 >>> <http://2db6bede5631d9b6-412158@88.99.244.106#015%23012> >>> Max-Forwards: 70#015#012Content-Length: 0#015#012 >>> User-Agent: kamailio (5.8.2 (x86_64/linux)) >>> >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1312]: >>> tls_h_read_f(): protocol level error >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_util.h:49]: >>> tls_err_ret(): TLS write:error:0A000086:SSL routines::certificate verify >>> failed (sni: unknown) >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1316]: >>> tls_h_read_f(): src addr: 52.114.75.24:5061 >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1319]: >>> tls_h_read_f(): dst addr: x.x.x.x:0 >>> /usr/local/sbin/kamailio[412165]: ERROR: <core> [core/tcp_read.c:1524]: >>> tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fa74d265d40 r: >>> 0x7fa74d265e68 (-1) >>> >>> >>> tls.cfg: >>> >>> [server:default] >>> method = TLSv1.2+ >>> verify_certificate = yes >>> require_certificate = yes >>> private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem >>> certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> server_name = test.mytest.com >>> >>> [client:default] >>> method = TLSv1.2+ >>> verify_certificate = yes >>> require_certificate = yes >>> private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem >>> certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> >>> Please suggest what I am missing. >>> >>> --- >>> >>> Thanks, >>> >> >> >> -- >> Thanks, >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > -- Thanks,

1 0

certificate verify failed (sni: unknown) integration with ms teams
by Muhammad Sohaib 27 Aug '24

27 Aug '24

Dear all, Trying to integrate Kamailio with MS Teams by following https://skalatan.de/en/blog/kamailio-sbc-teams kamcmd dispatcher.list | egrep "URI|FLAGS" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP Kamailio Logs: /usr/local/sbin/kamailio[412158]: INFO: <script>: Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012 Via: SIP/2.0/TLS x.x.x.x:5061;branch=z9hG4bK5dad.92de50b2000000000000000000000000.0#015#012 To: <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012 From: <sip:test.mytest.com >;tag=5d0939b82abe9b1bbf185d963b6e6c88-edeb3c71#015#012 CSeq: 10 OPTIONS#015#012 Call-ID: 2db6bede5631d9b6-412158(a)88.99.244.106#015#012 Max-Forwards: 70#015#012Content-Length: 0#015#012 User-Agent: kamailio (5.8.2 (x86_64/linux)) /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1312]: tls_h_read_f(): protocol level error /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_util.h:49]: tls_err_ret(): TLS write:error:0A000086:SSL routines::certificate verify failed (sni: unknown) /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1316]: tls_h_read_f(): src addr: 52.114.75.24:5061 /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1319]: tls_h_read_f(): dst addr: x.x.x.x:0 /usr/local/sbin/kamailio[412165]: ERROR: <core> [core/tcp_read.c:1524]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fa74d265d40 r: 0x7fa74d265e68 (-1) tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem server_name = test.mytest.com [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem Please suggest what I am missing. --- Thanks,

5 5

I have mo media and I must park
by Alexandru Covalschi 27 Aug '24

27 Aug '24

Hello list! Please help me understand if putting my b2bua in sensory deprivation is actually the best approach to punish it for sole reason of its existence. The background of this horror story is pretty cliche, call center needs a way to infinitely expand, so I'm building a scaling telephony core. Our main protagonist is an innocent b2bua with a very grounded backstory: step 0: receive external instruction to start a call step 1: originate step 2: park step 3: play moh until: - - a) told to pass it to other component - - b) told to hang up - - c) hang up on local timeout A simplified diagram of the call flow looks something like this (repeating names are same components): ### Signaling ### Remote Destination ^ Kamailio-external ^ Freeswitch-originator V Freeswitch- queue V Kamailio-local V Agent ### end of signaling ### ### Media ### Remote Destination ^ Rtpengine-external ^ Freeswitch-originator V Freeswitch-queue V Rtpengine- local V Agent ### end of media ### Due to specifics of the setting, re-negotiating the media and any non-essential signaling with remote destination is a very much last resort. And, as you can imagine, after step3a (pass it to other component), the character development of Freeswitch-originator service stops completely. Well, at least it is doing an honest job by playing the music on hold, which a little bit justifies its presence on the media path. But not for long. In the world of near future, pull request number 3956 is merged with the master branch, exposing the rtpengine API and leaving me no reason not to move MoH to a kernel media player. At this moment I feel a wild urge to get rid of Freeswitch-originator at all, but then I read the sources for mohq and dialog to remind myself that Kamailio is a proxy and proxies can't do this... So I fill b2bua's dialplan with 387.44 million lines of hateful comments and remind it to rcvonly every time it tries to warn the others about the grim fate. When the queue designates an agent to his call, theoretically I don't even need the second freswitch as well - if agen't media session has been established (sometimes ICE take tooo long) with Rtpengine- local, I can bridge their media sessions using publish and subscribe. I'd like to ask your opinion, list: do you see any plotholes potential issues with such deisgn? I'd honestly like to set Freeswitch to bypass-media, so it wont' waste ports and cpu for nothing at all. Are there any issues if I add body for the iniitial invite in the request_route as if it was received in there? Maybe there's a better way, overall? Any critique is welcome! Thank you -- AC

1 0

How to invoke RTPengine for re-Invite 200 OK within dialog
by Shah Hussain Khattak 27 Aug '24

27 Aug '24

Hello Experts, I need your advice on the following issue if you can kindly help. I am working on a hold call flow: UA - > Kamailio (RTP Engine) - > FS initial call establishes fine, RTP engine is invoked and SDP media IP is modified correctly. Then 1. I am receiving a hold re-invite with a send-only 2. I see Kamailio is changing the media IP in the SDP (with my RTP engine IP) when it sends the Invite to my core FS server. 3. FS sends 200 OK with a=recvonly 4. Kamailio relays the message as it is to the other leg. can someone advise me, on how can I invoke rtpengine at step 4, so the SDP is sent with my media server IP in the 200 OK? --- 200 OK from FS ---- SIP/2.0 200 OK Via: SIP/2.0/UDP 13.54.01.02:5060;branch=z9hG4bKd2e8.8a573af8d927f67c1db4550676f8be88.0;i=1 Via: SIP/2.0/TCP 192.168.10.117:51280;received=218.33.01.02;rport=51280;branch=z9hG4bKPj7d1d8968830d4ed3922eec7da7cdc17c;alias Record-Route: <sip:13.54.01.02:5060;r2=on;lr=on> Record-Route: <sip:13.54.01.02;transport=tcp;r2=on;lr=on> From: <sip:0403225908@trunk.xyz.corp>;tag=6a180ffa7f824972a2c59ba47f293acb To: <sip:+61285038000@trunk.xyz.corp>;tag=DZrpcpgS1BDKF Call-ID: 4154062196634c34a2dbcb6b612bf44c CSeq: 12263 INVITE Contact: <sip:+61285038000@54.88.54.88:5060;transport=udp> User-Agent: UserAgentX/v1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1724635755 1724635757 IN IP4 54.88.54.88 s=FreeSWITCH c=IN IP4 54.88.54.88 t=0 0 m=audio 21202 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly a=ptime:20 a=rtcp:21203 IN IP4 54.88.54.88 --- 200 OK relayed towards the UA ---- 2024/08/26 17:22:39.923639 172.16.8.126:5060 -> 218.33.78.33:51280 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.10.117:51280;received=218.33.01.02;rport=51280;branch=z9hG4bKPj7d1d8968830d4ed3922eec7da7cdc17c;alias Record-Route: <sip:13.54.01.02:5060;r2=on;lr=on> Record-Route: <sip:13.54.01.02;transport=tcp;r2=on;lr=on> From: <sip:0403225908@trunk.xyz.corp>;tag=6a180ffa7f824972a2c59ba47f293acb To: <sip:+61285038000@trunk.xyz.corp>;tag=DZrpcpgS1BDKF Call-ID: 4154062196634c34a2dbcb6b612bf44c CSeq: 12263 INVITE Contact: <sip:+61285038000@54.88.54.88:5060;transport=udp> User-Agent: UserAgentX/v1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1724635755 1724635757 IN IP4 54.88.54.88 s=FreeSWITCH c=IN IP4 54.88.54.88 t=0 0 m=audio 21202 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly a=ptime:20 a=rtcp:21203 IN IP4 54.88.54.88 my configs for within dialog route: ##### route[WITHINDLG] { append_hf("X-Test-Header: route_WITHINDLG\r\n"); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if (is_method("ACK")) { xlog("L_INFO", "Relaying ACK\n"); } if (is_method("INVITE|UPDATE")) { if (has_body("application/sdp") && search_body("m=image") && search_body("T38")) { xlog("L_INFO", "T.38 re-INVITE detected, skipping rtpengine invocation\n"); } else { xlog("L_INFO", "Non-T.38 INVITE detected, invoking rtpengine\n"); rtpengine_manage("replace-origin replace-session-connection"); } } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); } exit; } } ##### looking forward to your usual guidance. Thank you! Regards, Shah

2 2

socket names for command line parameter "listen"
by Xenofon Karamanos 23 Aug '24

23 Aug '24

Hello all, I wanted to inform you about an upcoming change regarding the syntax we use for defining socket names in the listen (-l) command line argument. Current Syntax: -l udp:10.10.10.10:5060/11.11.11.11:5060 Proposed Options: 1. -l udp:10.10.10.10:5060/11.11.11.11:5060=s1 2. -l s1=udp:10.10.10.10:5060/11.11.11.11:5060 ( A draft PR about this option can be found in https://github.com/kamailio/kamailio/pull/3954) [https://opengraph.githubassets.com/69f8a4a1d2c8988372b4942f84bde95de94502d2…]<https://github.com/kamailio/kamailio/pull/3954> main: Add support for naming socket in cli by xkaraman · Pull Request #3954 · kamailio/kamailio<https://github.com/kamailio/kamailio/pull/3954> Pre-Submission Checklist Commit message has the format required by CONTRIBUTING guide Commits are split per component (core, individual modules, libs, utils, ...) Each component has a single... github.com Please note that the socket name (s1) in both options is/should be optional. The change will allow for better clarity and flexibility when defining sockets, particularly when a name needs to be assigned and match the config file style. I would appreciate your feedback on which syntax you find more intuitive or any concerns you might have with these options. Your input is important to ensure a smooth transition. Best regards, Xenofon

2 2

Kamailio cluster options
by Ali Taher 22 Aug '24

22 Aug '24

Hello all, I'm using Kamailio as SIP redirect (act as routing decision for SBC) depending mainly on sqlops and db_postgres modules, and it's working perfectly. What I need now is to have a kind of cluster of Kamailio servers for high availability purpose. What is the best option for me ? BR

2 1

push server tcp problem
by Bernd Krueger-Knauber 20 Aug '24

20 Aug '24

Hi, I implemented an apple and android push server/proxy with kamailio 5.8.1. The server/proxy is transparent. It works fine with UDP, since the registration after the push comes from the same ip and port. So t_continue works without problems. (We use the new registration as signal thet the phone is ready) If we use tcp at the client, he gets the push, wakes up and send the new registration on an other port. The SIP stack uses now an other connection. t_continue fails, because it uses the old data of the original invite acccording to the first registration. Is it possible to change the stored connection values according to the new registration data? Best regards, Bernd

2 2

low performance with udp_children and async_workers
by elhar.mohamed＠gmail.com 19 Aug '24

19 Aug '24

Hello, Based on business requirements, 35% of invites should be handled by async_workers and 65% with be handled by main children. The target is 1500cps. I'm using the following conf : children=64 async_worker=128 when doing performance tests using SIPp, I figured out the 1500cps is well supported by the conf above, but when I updated the data in database to not use anymore async workers, the cps is getting decreased and it was between 200 and 500. This a client requirements is to be able to enable/disable feature that needs to be handled by async workers. Could you please help me on this? Thanks in advance

4 12

Inbound call to webRTC failing with TCP/TLS connection (id: 0) for WebSocket could not be found
by hollis.rhonda＠gmail.com 16 Aug '24

16 Aug '24

We're seeing an issue where some webRTC clients are not receiving inbound calls. Kamailio logs show the following error: WARNING: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: <core> [core/msg_translator.c:3007]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: <core> [core/msg_translator.c:2086]: build_req_buf_from_sip_req(): could not create Via header ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_fwd.c:484]: prepare_new_uac(): could not build request ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_fwd.c:1764]: t_forward_nonack(): failure to add branches DEBUG: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_funcs.c:358]: t_relay_to(): t_forward_nonack returned error -2 (-2) DEBUG: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_funcs.c:376]: t_relay_to(): -2 error reply generation delayed It seems to be happening after the client has been connected to Kamailio for more than 24 hours. The socket connection details look correct in the external DB and in the output of ws.dump, core.tcp_list, and ul.dump. A restart of Kamailio or a reload of the webRTC client, triggering a new websocket connection, will clear the issue. However, we haven't been able to determine exactly when and why a client ends up in this state. While a client is in this state, REGISTER requests are still handled successfully, they can make outbound calls and TCP keepalives from the client are working. I'm still working on trying to debug this with GDB so I can see what the value of send_info is in https://github.com/kamailio/kamailio/blob/master/src/core/msg_translator.c#…, but I wasn't having luck with multiple child processes running, and now my client has reloaded and is working again. I've restarted Kamailio with a single child process, so when it does start failing again, I may have better luck with GDB. Any thoughts on what might be causing clients to get into this state? Whatever additional information I can provide that might help, I'm happy to share. Thanks!

2 4

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 16 Aug '24

16 Aug '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

3 2

[http_client] KEMI use "curl_connect_post" send http request error
by 17603019528＠163.com 16 Aug '24

16 Aug '24

ALL: Hi, here is my lua script: ``` local body = '{"ip":"172.16.4.111", "ports":[5261, 5260]}' local resp = "" local code = KSR.http_client.curl_connect_post("configapi", "", "application/json", body, resp) ``` the error is : ``` 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[0] for: curl_connect_post is str: configapi 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[1] for: curl_connect_post is str: 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[2] for: curl_connect_post is str: application/json 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[3] for: curl_connect_post is str: {"ip":"172.16.4.111", "ports":[5261, 5260]} 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[4] for: curl_connect_post is str: 24(2406) ERROR: <core> [core/pvapi.c:429]: pv_cache_get(): invalid parameters 24(2406) ERROR: http_client [http_client.c:739]: ki_curl_connect_post(): failed to get pv spec for: ``` I know the error means: "resp" type is not right, what is the correct type ? I search google for this issue, no one knows, someone please help me. Thinks

2 2

TLS module doesn't support TLSV1
by o.atef＠fiberme.com 15 Aug '24

15 Aug '24

Hello Brothers, I've installed kamailio throw "apt install" on Debian and it's installed version: kamailio 5.6.3 (x86_64/linux). Now I've a big problem that kamailio cannot running with TLSv1 and it has to be TLSv1.2+ as tls.cfg doc said: # We do not enable anything else than TLSv1.2+ # over the public internet. Clients do not have # to present client certificates by default. How could I avoid this restriction please to enable TLSv1? Thank you,

5 5

[PHONENUM] Match with country on Lua/KSR
by João Silva 14 Aug '24

14 Aug '24

Hello everyone! The function phonenum_match_cn from module PHONENUM is not available on KSR. Is there any work around to use it on Lua config? Best regards, Joao

2 2

Incoming call with one one way audio and ctting off after thirty seconds
by palany 14 Aug '24

14 Aug '24

I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as my SBC. Calls from freepbx to teams are working fine with audio and I am having challenges with calls from teams to freepbx. The calls rings and after answering there is only way audio and the call drops off after thirty seconds. My pbx is behind a microtik router and I have port forwarded my sip port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is hosted in AWS. I have attached both my pcap from Kamailio, freepbx and my Kamailio.cfg. I am thinking It's a misconfiguration challenge or something else. Please help I am failing to see my challenge. Best Regards,

3 3

Authentication
by Vishal Pai 13 Aug '24

13 Aug '24

Hello All I am using kamailio sip account as gateway at Freeswitch side but when I enable caller id from = true then call from Freeswitch to kamailio didn’t authenticate As I check it’s about from header when I disable caller id from = false the call get authenticate since it get sip username and domain but when there is caller id the from header get callerid@ip address Can anyone guide me Thanks

2 1

Kamailio hangs on get_profile_size method.
by satyaprakash ch 12 Aug '24

12 Aug '24

HI, We are facing a hang issue with Kamailio for the get_profile_size method. Randomly, the Kamailio got hangs, and the CPU went 100%. During the hanging time, we generated the TRAP and we could see that the hang happened on the get_profile_size method. We are getting this hang randomly in dlg_profile.c of the following lines: The hang occurs, causing the CPU to spike to 100% at one of the following lines randomly: 1. if(value->len == ph->value.len && memcmp(value->s, ph->value.s, value->len) == 0) { 2. ph = ph->next; 3. while(ph != profile->entries[i].first); During our analysis, we found that PROCESS ID 7608 gets a spike of 100% utilization. We suspect there may be an issue with the processing of the 'ph' variable, but we couldn't identify the root cause. I've attached the Kamailio TRAP for further details. Could you please suggest what might be causing this problem and how we can resolve it? Thank you.

1 1

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 10 Aug '24

10 Aug '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria [aws](https://www.igmguru.com/cloud-computing/aws-developer-certification-tr…

4 4

Kamailio hangs on get_profile_size method.
by satyaprakash ch 09 Aug '24

09 Aug '24

HI, We are facing a hang issue with Kamailio for the get_profile_size method. Ramdomly, the Kamailio got hangs, and the CPU went 100%. During the hanging time, we generated the TRAP and we could see that the hang happened on the get_profile_size method. We are getting this hang randomly in dlg_profile.c of the following lines: The hang occurs, causing the CPU to spike to 100% at one of the following lines randomly: 1. if(value->len == ph->value.len && memcmp(value->s, ph->value.s, value->len) == 0) { 2. ph = ph->next; 3. while(ph != profile->entries[i].first); During our analysis, we found that PROCESS ID 7608 gets a spike of 100% utilization. We suspect there may be an issue with the processing of the 'ph' variable, but we couldn't identify the root cause. I've attached the Kamailio TRAP for further details. Could you please suggest what might be causing this problem and how we can resolve it? Thank you.

1 0

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