- sr-users - kamailio.org

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*

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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

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