- sr-users - kamailio.org

Incoming call with one one way audio and cutting off after thirty seconds
by palany 08 Aug '24

08 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. Below are google drive links logs for the incoming call in 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. kamailio.pcap - https://drive.google.com/file/d/1aoJhksgwcH-52unwxuyzn8GAmG9PEnOM/view?usp=s haring freepbx.pcap - https://drive.google.com/file/d/1JJXs9UOUFak5Y4uWI_ARB56vQfVYfSDj/view?usp=s haring kamailio.cfg - https://drive.google.com/file/d/1v2ZtylOf8fIMYH4OckGmnP489PO7WI7J/view?usp=s haring Best Regards

2 2

Dialog timer reset on UPDATE
by Alex Balashov 08 Aug '24

08 Aug '24

Hi, I have a dialog where I have set: dlg_set_property("timeout-noreset"); But it seems the timeout is reset anyway when handling an in-dialog UPDATE request, at least, on Kamailio 5.5.4. This does not seem to happen with reinvites. I haven't been able to get more information thus far, other than to be able to say that there is nothing unusual about the UPDATE request and there are no additional errors at log level 2. I was wondering if anyone else has encountered this. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

1 0

app_python3 method not found or is not callable
by Westhuis, Johannes 08 Aug '24

08 Aug '24

Hi, we are using kamailio with app_python3. It works very good, but in some cases, there are error logs stating: app_python3 [python_exec.c:91]: apy_exec(): <python method> not found or is not callable It happens sporadically, and a restart of kamailio fixes it. As far as i can see it might be linked to concurrent arrival of several messages. Best regards, Johannes # kamailio -v version: kamailio 5.5.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_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: compiled on 12:01:03 Jul 22 2024 with gcc 10.2.1

1 0

http_async_query on carrierfailureroute changes rd back to carrieroute primary
by Maharaja Azhagiah 07 Aug '24

07 Aug '24

Hi I am running kamailio 5.4 on debian I have carrierfailureroute configured incase of primary service provider fails. I also have Stirshaken configured to add Identity header on outbound calls. Issue is when call fail overs to carrierfailureroute, http_async_query changes $ru to the primary carrier From the debug logs, when primary carrier sends a 488 (primary carrier expects SIP TLS but my call is UDP - to test the failover scenario) 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} tmx [t_var.c:561]: pv_get_tm_reply_code(): reply code is <488> 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:178]: set_next_domain_on_rule(): searching for matching routing rules39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:197]: set_next_domain_on_rule(): next_domain is 47987 Carrier route rewrites the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:706]: cr_do_route(): uri 14371234567 was rewritten to sip:14371234567@sip.primaryprovider.com, carrier 1, domain 47987 Before http_async_query rd and ru are still the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2976] - Debug testing ----- rd is sip.primaryprovider.com ----- ru is sip:14371234567@sip.primaryprovider.com 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:469]: async_send_query(): transaction suspended [5261:1830449764] 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:625]: async_push_query(): query sent [ https://authn-uat.ccid.neustar.biz/ccid/authn/v2/identity?apiKey=randomkey] (0x7fdcad097e60) to worker 1 However, when the route is being called after the http_async_query it changes to the primary one: 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) DEBUG: http_async_client [async_http.c:235]: async_http_cb(): resuming transaction (5261:1830449764) 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) INFO: <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2995] - Debug testing ----- rd is 1.2.3.4 ----- ru is sip:14371234567@1.2.3.4:5061;transport=TLS Due to this, call keeps going to the primary and it fails if ( http_async_query(STIRSHAKEN_AS_URL, "AS_RESPONSE") == -1 ) { xlog("L_ERR ", "[cfg:$cfg(line)] Failed to connect AS service for token $fu -> $tu \n"); return; } route[AS_RESPONSE] { xlog("L_INFO", "[callid: $ci] - [cfg:$cfg(line)] - Debug testing ----- rd is $rd ----- ru is $ru\n"); if ($http_ok) { xlog("L_INFO", "[cfg:$cfg(line)] Resuming outbound call transaction for $fu -> $tu Received - $http_rb \n"); # Add identity and Date headers if (jansson_get("identity", $http_rb, "$var(identity)")) { insert_hf("Identity: $var(identity)\n", "Content-Length"); } if (jansson_get("date", $http_rb, "$var(date)")) { if ($hdr(Date) != $null){ remove_hf("Date"); } insert_hf("Date: $var(date)\n", "Identity"); } } else { xlog("L_ERR", "[cfg:$cfg(line)] Resuming outbound call transaction. Error - $http_err)\n"); } route(RELAY); exit; } Please help to understand why rd / ru changes to primary carrier. Regards, Maharaja Azhagiah

3 7

Kamailio hang Issue
by satyaprakash ch 06 Aug '24

06 Aug '24

Hi, We are encountering a hang issue after deploying the following code: *route(DLG_MANAGE);* *if ($dlg_var(voip_id) == $null) { $dlg_var(voip_id) = $hdr(X-Voip-ID);}route[DLG_MANAGE] { if (is_method("INVITE")) { dlg_manage(); }}* This code includes the set_dialog_profile. We have this setup with 2 proxy servers with DMQ enabled, but the issue only occurs on one proxy. When the Kamailio proxy hangs, no logs are written. If you have any suggestions to resolve this issue, please help. Thanks.

2 2

Kamailio-IMS: After a abornamal Call, other call is not working (even after a new registration)
by deep.upreti＠gmail.com 05 Aug '24

05 Aug '24

Hello, Can anyone help me to resolve one issue? scenario is - 1- Register two UE (UE-A and UE-B) 2- Call from UE-A to UE-B for a specific duration (lets say: 100 seconds) 3 Verify if call is working: Answer is YES, call is working. 4. shutdown the eNB 5. Observation: No BYE/De-registration message is not coming as expected. 6. Again start eNB and register same UE (UE-A and UE-B) 7. Register UE-A and UE-B both 8. It will not remove previous contacts or not updating the reg-state. 9. I am running with Rx enable mode.

1 1

Multiple TLS connections to the same IP:Port
by Nick Digalakis 05 Aug '24

05 Aug '24

Hello everyone, I am trying to use a single Kamailio server to register to multiple MS Teams Direct Routing endpoints. The config snippet I am using is this: sht_iterator_start("i1", "teams-endpoints"); while(sht_iterator_next("i1")) { $var(teams_endpoint) = $shtitkey(i1); $xavp(tls=>server_name) = $var(teams_endpoint); $xavp(tls[0]=>server_id) = $var(teams_endpoint); $uac_req(method)="OPTIONS"; $uac_req(ruri)="sip:sip.pstnhub.microsoft.com:5061;transport=tls"; $uac_req(furi)="sip:" + $var(teams_endpoint); $uac_req(turi)="sip:sip.pstnhub.microsoft.com:5061;transport=tls"; $uac_req(hdrs)="Contact: <sip:" + $var(teams_endpoint) + ":" + "5061" + ";transport=tls>\r\n"; ### Create a unique Call-ID based on the Timestamp and the Message Body in MD5 $var(unhashed_cid) = $TV(Sn) + $mb + ""; $uac_req(callid)=$(var(unhashed_cid){s.md5}); uac_req_send(); } sht_iterator_end("i1"); When the HTable has only one endpoint, everything works fine. When I add a second endpoint, the first one continues to work but the second one failing with the error from Microsoft: Q.850;cause=63;text="85babcde-e0b5-4a85-8f4a-12345678c9ae;SBC certificate is not issued correctly. Provided trunk FQDN 'endpoint-02.domain.com' is not included in certificate's CN or SAN list. Certificate allows following FQDNs only: endpoint-01.domain.com. After some digging around, I realized that all endpoint after the first fail because Kamailio is re-using the same TLS connection for all subsequent OPTIONS as well, but of course the connection has been established with the certificate of the first endpoint. I have tested the certificates by switching around the first endpoint, so that shouldn't be a problem. Is there any way I can force it to establish a new TCP/TLS connection for each subsequent request? Any help would be much appreciated, I have been pulling my hair out with this one! Best regards, Nick [https://s-install.avcdn.net/ipm/preview/icons/icon-envelope-tick-green-avg-…]<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free.www.avg.com<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…>

2 2

How to log datetime in files
by 陈子妍 04 Aug '24

04 Aug '24

Hello! I am working on with Kamailio, I have already understood to add log prefix like `log_prefix="NAT {$rm $mt $hdr(CSeq) $ci}: "`, but I also want to record event time, so how can I config `datetime` in log files. Thanks~

3 2

Why does $T(id_index) return 0 even though t_suspend() is executed successfully?
by mm e 01 Aug '24

01 Aug '24

if(t_suspend()) { $dlg_var(start_time) = $timef(%Y/%m/%d %H:%M:%S); xlog("L_NOTICE", "==> processing suspended in transaction [$T(id_index):$T(id_label)] "); } else { xlog("L_ERR", "t_suspend err"); exit; } Sometimes, I get this log==> processing suspended in transaction [0:13614] "，as far as I know, 0 means that the value cannot be obtained. Executing t_continue(tindex, tlabel, rtname) with parameter 0 will fail.by the way, It's a question of probability.

1 0

Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 31 Jul '24

31 Jul '24

Hello Daniel, Regarding your question, I think only the avp(msg) should be destroyed as others avp's can be used elsewhere in the script. I tried the lastest git master and when restarting kamailio I got a segmentation fault: Jul 18 21:17:16 labcom070 systemd[1]: Started Kamailio - the Open Source SIP Server. Jul 18 21:17:17 labcom070 kernel: kamailio[2285]: segfault at 0 ip 00000000007eb66a sp 00007ffd7f80cc00 error 6 in kamailio[400000+533000] Jul 18 21:17:17 labcom070 kernel: Code: 41 56 41 55 41 54 53 48 81 ec c8 00 00 00 48 89 bd 48 ff ff ff 48 89 b5 40 ff ff ff 48 89 95 38 ff ff ff 48 8b 85 48 ff ff ff <48> c7 00 00 00 00 00 48 8b 85 48 ff ff ff c7 40 08 00 00 00 00 48 Jul 18 21:17:17 labcom 070 systemd[1]: Started Process Core Dump (PID 2286/UID 0). Jul 18 21:17:18 labcom070 systemd-coredump[2287]: Resource limits disable core dumping for process 2285 (kamailio). Jul 18 21:17:18 labcom070 systemd-coredump[2287]: Process 2285 (kamailio) of user 993 dumped core. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Main process exited, code=dumped, status=11/SEGV Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Failed with result 'core-dump'. Jul 18 21:17:18 labcom070 systemd[1]: systemd-coredump(a)6-2286-0.service: Succeeded. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 1. Jul 18 21:17:18 labcom070 systemd[1]: Stopped Kamailio - the Open Source SIP Server. Jul 18 21:17:18 labcom070 kernel: kamailio[2298]: segfault at 0 ip 00000000007eb66a sp 00007ffcd101c240 error 6 in kamailio[400000+533000] Jul 18 21:17:18 labcom070 kernel: Code: 41 56 41 55 41 54 53 48 81 ec c8 00 00 00 48 89 bd 48 ff ff ff 48 89 b5 40 ff ff ff 48 89 95 38 ff ff ff 48 8b 85 48 ff ff ff <48> c7 00 00 00 00 00 48 8b 85 48 ff ff ff c7 40 08 00 00 00 00 48 Regards, Vanderlei ________________________________ De: Daniel-Constantin Mierla <miconda(a)gmail.com> Enviado: quinta-feira, 18 de julho de 2024 02:51 Para: Vanderlei Torres Batistela <v.batistela(a)hotmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Assunto: Re: [SR-Users] Possible Kamailio memmory leak - version 5.8.2 Hello, indeed the previous commit was destroying a single avp, I pushed a new one to destroy all of them with that name. I am wondering if all avps, no matter the name should be destroyed. Anyhow, could you fetch the latest git master version and try your scenario again? Cheers, Daniel

1 0

[TM] [ASYNC] : Freeing transaction after sending 302 response
by elhar.mohamed＠gmail.com 31 Jul '24

31 Jul '24

Hello, I'm using the async_task_group_route for async module to handle some invites, when testing with sipp tools, I got some Dead call error message and after investigating It seems that for the same invite I sent multiple 302 message to sipp because the ACK sent by sipp was not handle before thr retr_timer1 & retr_timer2. Please can you help me to free the transaction after sending the first 302 ? Thank you in advance.

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '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)

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '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)

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '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)

1 0

Job: System Engineer VoIP - Germany / HO
by Gerrit Jacobsen 25 Jul '24

25 Jul '24

VIER is looking for a System Engineer VoIP What VIER does VIER is a global contact center technology provider. VIER develops and runs its own technology in several data centers. You will be part of the Real-time Media Processing Team which is responsible for the telephony connections between carriers, applications and call-center agents. Your mission You are responsible for the support of the VoIP systems as well as the associated infrastructure Since you have sovereignty over the VoIP systems, you are also responsible for performing system upgrades and configurations. You have a lot of freedom to design, plan, and implement new VoIP system components for internal and external projects. You manage and implement change requests and are a specialist for special topics in the communication area. You show a high-sense of responsibility and foresight to keep systems running 24/7 You are prepared to work occasionally also in the evening hours to roll out new features Your skills You have several years experience in the day-to-day operation of telecom systems You have worked extensively with opensource VoIP software such as Kamailio / OpenSIPS, FreeSWITCH You can administer Linux systems such as RockyLinux and Debian You know SIP, RTP, RTCP, TLS, WebRTC protocols and can debug problems with Wireshark You have also worked with MySQL, Git You are communicative and like to work with team colleagues Further beneficial skills Experience with Ansible or Puppet for automation and configuration management. CheckMK, Homer, Hepic, Redis, CGRates, CDR Tools, Python Own VoIP software projects German language skills Please apply here: https://vier.jobs.personio.de/job/1638419?language=en&display=en

1 0

RTJSON: No Json for routing
by felipe.nunes＠comuniki.me 24 Jul '24

24 Jul '24

Good afternoon I have the following error in the application: [rtjson_routing.c:701]: rtjson_update_branch(): no json for routing This error occurs regardless of whether there is high demand or not and the application sending the data correctly. Can you help me with this? Felipe Nunes

2 1

ds_select from xhttp:request
by Christian Berger 24 Jul '24

24 Jul '24

Hi, we want to execute ds_select from an xhttp:request event route. The idea is to be able to get all active targets of a dispatcher set so our monitoring can check that via http which would be convenient for us. So are there any reasons why ds_select wouldn't work in an xhttp:request even route? 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

3 2

Listening on multiple TLS ports
by João Silva 23 Jul '24

23 Jul '24

Hello everyone! Is there a way to define a listen port range for TCP/TLS? Or any macro I can use to create the LISTEN commands? And finally is there a performance penalty for having multiple (2k) listening ports at the same time? I am in a strange situation where I need to register multiple credentials on a single SIP trunk, for each registration I need to use a different local port otherwise the trunk will simply overwrite the previous registrations. Furthermore, when I place a call on that trunk I have to be consistent with the port I used for registration for that specific credential. So I am looking into explicitly defining the port to be used per credential, UACREG allows me to use the contact_addr/socket for the registration part and for placing of outbound calls I can use $fsn to force the correct socket. Best regards, Joao

3 4

IMS PCSCF force source IP address
by Anthony Blandin 23 Jul '24

23 Jul '24

Hello, I am using Kamailio pcscf with ipsec. I have one ipsec listen address which is used by the UE to connect to the PCSCF and one other listen address on port 5060. During the registration until the ipsec is not established, the UE sends the requests to the ipsec listen address and the pcscf is using the other listen address to forward the sip request to the ICSCF. That's is ok for me. But when the ipsec is established after the 200 OK REGISTER, the UE sends a SUBSCRIBE but the pcscf forwards the SUBSCRIBE to the SCSCF using its ipsec listen address instead of the other listen address. I tried to use $fs to force the socket but without success. Do you know how to force the source IP address to be used on the pcscf? Thanks for your support. Regards, Anthony

1 0

mysql error after migration from 5.7.5 to 5.7.6
by Alejandro Guralnik 23 Jul '24

23 Jul '24

Hello, After upgraded from kamailio 5.7.5 to 5.7.6, when did start kamailio process, got the following error: WARNING: db_mysql [km_my_con.c:179]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring After downgrade to 5.7.5 dind't happened. (we don't use mysql ssl connection).

2 3

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '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 https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…

1 0

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '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 <a href="https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…">aws</a>

1 0

SHM Memory leak?
by Ihor Olkhovskyi 23 Jul '24

23 Jul '24

Hello! I'm trying to get where all the SHM memory gone. For the moment I'm using these settings: -m 8192 -M 1024 which means 8G of SHM meory. Kamailio parameters are the following fork=yes children=8 tcp_children=12 enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 And having ~1000 TLS/WSS clients I'm getting this after 3 days of running: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 2796", "shmem:free_size = 3737947072", (3.7 G) "shmem:max_used_size = 4857418512", (4.8 G) "shmem:real_used_size = 4851987520", "shmem:total_size = 8589934592", "shmem:used_size = 4838988096" (4.8 G) ], "id": 984479 } Means half of SHM memory is gone When running kamcmd mod.stats all shm I'm getting (output truncated for the largest values) Module: core { ... tcpconn_new(1201): 130669280 (0.13 G) ... Total: 131657632 (0.13 G) } Module: sl { ... Total: 7520 } Module: siptrace { ... Total: 13520 } Module: rr { Total: 0 } Module: dialog { ... Total: 146080 } Module: permissions { ... Total: 62144 } Module: htable { .... Total: 3359552 (3.3 M) } Module: rtpengine { ... Total: 31456 } Module: textopsx { Total: 0 } Module: tsilo { ... Total: 75072 } Module: tm { .... Total: 3459504 (3.4 M) } Module: usrloc { .... Total: 1217616 } Module: pua_dialoginfo { ... Total: 8720 } Module: pua { ... Total: 150848 } Module: pike { ... Total: 9504 } Module: websocket { wsconn_add(198): 54139856 (54 M) ... Total: 54148096 (54 M) } Module: debugger { ... Total: 21872 } Module: tmx { Total: 512 } Module: kex { .... Total: 1888 } Module: tls { ... ser_malloc(364): 87246112 (87 M) ... Total: 87997168 (87 M) } Module: secfilter { ... Total: 768 } Module: exec { ... Total: 16 } Module: dispatcher { ... Total: 2992 } Module: cfgutils { ... Total: 48 } Module: app_python3 { ... Total: 32 } So, I'm guessing, where all the memory gone? Is there any way to get more detailed info on SHM? Kamailio 5.8.2 -- Thanks in advance, Ihor

2 6

Not responding to OPTIONS if existing tcp connection
by Jon Bonilla (Manwe) 22 Jul '24

22 Jul '24

Hi all version 5.7.6 Debian 12 I'm having issues with a kamailio server not sending 200 to some OPTIONS requests. It does send 200 to some subscribers but not 100% of the time to 100% of subs. I have enabled debug=3 in that server and this is what I found: If the log says that it found existing tcp connection and it's reusing it I can't see the 200ok in a sngrep capture. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 2 <core> [core/tcp_main.c:2614]: tcpconn_send_put(): send from reader (360359 (18)), reusing fd <core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending... <core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c= 0x7fb0faa893d8 n=384 fd=14 <core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf=#012SIP/2.0 200 Keepalive#015#012Via: SIP/2.0/TCP 10.35.190.105:5060;branch=z9hG4bK15ebf764;rport=55894;received=1.1.1.186#015#012From: "Unknown" <sip:contesta01@10.35.190.105>;tag=as5311c3a7#015#012To: <sip: 1.1.1.138>;tag=054524a74bcccf444a0f08f67c5b7657.d07b5401#015#012Call-ID: 15dcd38c46f915576ded132a11d8ff4c@10.35.190.105:5060#015#012CSeq: 102 OPTIONS#015#012Server: aaa SBC#015#012Content-Length: 0#015#012#015#012 <core> [core/receive.c:531]: receive_msg(): request-route executed in: 302 usec On the other side, if kamailio doesn't find an existing tcp connection (at least nothing in the logs about that) I can see the 200 ok in sngrep. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/receive.c:531]: receive_msg(): request-route executed in: 295 usec All subs use tcp. And the kamailio.cfg is very simple regarding OPTIONS: Sanity, check maxfwd and sl_send_reply(200). Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

Drouting using TLS and domain names
by SAMUEL MOYA TINOCO 19 Jul '24

19 Jul '24

Hi everyone, I'm trying to configure drouting module but i'm not sure how to configure dr_gateways table to use domain names and TLS instead of IPs How should I do it? In dispatcher module is by using sips:domain but when I configure it like this I have problems to send invites (maybe I'm doing something wrong). Does anyone know how to configure it? Thanks in advance for your help Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DAD156.D8CFF170] Soluciones inteligentes para la autonomía personal

2 3

Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 18 Jul '24

18 Jul '24

Hello all, I am using a very simple scritp with corex module enabled . Normal calls works fine, but a scenario where INVITE does not receive response and the call is released with SIP 408 (request timeout) leads to memmory leak. I verified it making a lot of calls with this scenario and printing the shared memory (using kamcmd core.shmmem) . Its ocurred with modparam("corex", "nio_intercept", 1). If I modify to modparam("corex", "nio_intercept", 0) the memmory leak does not ocurr anymore. The problem also occurs in another scenario where the SIP Cancel message is not responded to. My script has this event route # event route event_route[network:msg] { xlog("L_INFO", "Dentro do Event_Route \n"); if (is_incoming()) { xlog("L_INFO", "Received message '$mb' \n"); $avp(msg) = $mb; xinfo("avp(msg) = $avp(msg)"); } else { xlog("L_INFO", "Sending message '$mb' \n"); $avp(msg) = $mb; }; } Test done wtih 4000 calls: [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 } I attached my kamailio script used and a log with one call using the command "kamcmd corex.shm_summary" with modparam("corex", "nio_intercept", 1) and modparam("corex", "nio_intercept", 0) Regards, Vanderlei

3 6

Re: Difference betwwen TM functions in KEMI
by Duarte Rocha 18 Jul '24

18 Jul '24

Hello Alex, You're totally right, that was the issue. Thank you so much for the help. Helpful as always. Cheers, Duarte Rocha

1 0

Difference betwwen TM functions in KEMI
by Duarte Rocha 17 Jul '24

17 Jul '24

Hello all, I'm using Kamailio with native script and KEMI in Javascript. I noticed that functions from the TM module can have different results if I call them on KEMI or Native script. Is this intended behaviour? Don't they share the state? In order to test this i made the following code : KEMI Code : function ksr_request_route() { try{ if ( KSR.is_method("BYE") ) { print.error("BYE Request KEMI BYE") KSR.route("exists_test") if(KSR.tm.t_exists()) { print.error("KEMI : Transaction exists") } else { print.error("KEMI : Transaction does not exist") } Native Code : route[exists_test] { if( t_exists()) { xerr("Native Script : Transaction exists \n"); } else { xerr("Native Script : Transaction does not exist \n"); } } I've also tried to change function call order but every time I get "Transaction does not exist" on native and "Transaction Exists" on KEMI. Is this intended behaviour? Should I only call tm functions in native or Kemi in order to have consistency ? Thanks in advance, Best Regards,

2 1

Dns record not alone?
by Pyry Aaltonen 17 Jul '24

17 Jul '24

Hi, I recently updated my testing env to 5.8.1 and after that I’ve been receiving dns_cache.c warning “record not alone”. I found out that it’s been added to quite recent commit: https://github.com/kamailio/kamailio/commit/d8a35b3b6c837b36779e232b65fce61… And that’s probably the why I haven’t seen it before. As far as I know, I have no other problem with dns or anything else than the warning appearing in the logs 😊 I can’t quite get what the warning is about? Clearly refers to dns queries, and with debug I found out that maybe its somehow related to ip which has several domains to pointing it (and queried by one of those domains by this kamailio). Can it be something with the PTR-record? And this warning comes ones every hour, so probably when some ttl expires, and meanwhile the cache is ok with the situation. And as a background, I haven’t done any specific dns-configurations in my config, so settings should be pretty much in defaults. Any advice what I should fix here to get the warning disappear? -Pyry

2 1

Add a Header to CANCEL request
by sadik.oualla.mohamed＠gmail.com 17 Jul '24

17 Jul '24

Hello all, I am facing an issue with Kamailio when trying to add custom headers to CANCEL requests. My current configuration adds the "Reason" header perfectly: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); } ``` However, when I attempt to add headers like "X-Reason" or "TestHeader", they are not added: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("X-Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); # or append_hf("TestHeader: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); #nothing changes in the relayed CANCEL request :-/ } ``` I've read in this discussion "https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…" that this might work if sending CANCEL in stateless mode, but I want to keep it stateful. Any suggestions on how to achieve this? Regards, Mohamed.

1 0

Changing ksr_tcp_accept_iplimit from default 1024
by Who AmI 16 Jul '24

16 Jul '24

Hi list, I hope you are all well? I am currently loadtesting some services internally and I am getting an odd error around TCP connections from the same source IP below: "/usr/sbin/kamailio[1806789]: CRITICAL: <core> [core/tcp_main.c:4459]: handle_new_connect(): hit the limit of connections per source IP (my_ip:39043) - rejecting" Its from the core the error so I was looking at the cookbooks and documentation but cannot get around this specific limit or find a configuration to tweak that fixes it. The limit only affects TCP (will affect TLS as well but I haven't got that far) and is 1024 it seems. Kamailio version is: 5.7.6 OS: Ubuntu 22.04 Things I have done so far: I have set limitNOFILE=65535 in the service file Set the tcp/tls_max_connections at 8192 (default is 2048) Increase the tcp_rd_buf_size to 16k+ The service I am using to loadtest is SIPp with flags "-t tn" for TCP multisocket as this service will have much more than 1024 concurrent TCP connections to it from a single source IP. I think I have missed something basic here as the devel and the source code references: "ksr_tcp_accept_iplimit" here at line 541: https://github.com/kamailio/kamailio/blob/master/src/main.c#L541 I cannot find a way to change this with a configuration line it seems, am I being blind or is my only option to build from source and change this? Many Thanks, John.

2 2

Weird telephone-event payload issue - need help
by Tim Bowyer 16 Jul '24

16 Jul '24

Hello Everyone, I've been grappling with this issue for some months, time to raise the white flag. Essentially I need to work out a way to configure Kamailio/RTPengine to normalise telephone-events to 101 earlier in the piece. 1. Call comes in from Carrier with INVITE w/SDP, telephone-event payload of 96. 2. INVITE sent to Asterisk with the same SDP, payload = 96 3. INVITE sent from Asterisk to Client with payload = 101 * 200 OK w/SDP from Client has payload = 101 * 200 OK w/SDP to Kamailio has payload = 96 * 200 OK w/SDP to Carrier has payload = 96 4. Asterisk (with Direct Media enabled) sends INVITE to Kamailio with payload = 96, and new media IP of Client 5. Kamailio passes this onto Carrier with media IP of RTPengine 6. 200 OK w/SDP from Carrier has payload = 96 sent to Kamailio 7. Kamailio passes this 200 OK to Asterisk, which responds with ACK and doesn't do anything further +---------+ +----------+ +---------+ +---------+ | Carrier | <----1-----> | Kamailio | <-----2----> | Asterisk| <----3------>| Client | | | | | | | | | +---------+ +----------+ +---------+ +---------+ ^ | | | | | |3| | | | | | +------5---------> +-----------+ <-----------------+------------4-----------+ | RTPengine | +-----------+ Big issue is, the Client is expecting telephone-events with a payload of 101, whereas RTPengine is passing it 96 from the Carrier. * Inbound DTMF only works at client-side if Asterisk stays in path (far from ideal). * The issue is exacerbated when calls are sent back out to the same or another carrier (forward scenario). * There's nothing I can seem to do in Asterisk to change how it handles DTMF in a direct media/re-invite setting. Any ideas? Cheers, Tim Bowyer

3 12

how to upgrade kamailio minor release from 5.7.5 to 5.7.6
by sadik.oualla.mohamed＠gmail.com 15 Jul '24

15 Jul '24

I am currently using Kamailio version 5.7.5, which I installed from the source code. I would like to upgrade to version 5.7.6 but have been unable to find documentation specifically addressing minor upgrades between these minor releases. I have found only wikis detailing the process for upgrading for major releases (like 5.7.x to 5.8.0). Could you please point me to the documentation for my current need or share with me steps on how to perform this minor upgrade? Thank you. Cheers.

3 4

How to get the rtpengine IP used by the current call among multiple rtpengines
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine

1 0

How to get the id or ip in the multiple rtproxies for LB with weights。
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine kamailio selected for the call.

1 0

Re: (no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi Brett! From what I could understand, my client (a Carrier) assigns services and products to its customers identified by trunkgroups (depending on the customer and the service requested, as well as the ingress trunkgroup the call will be forwarded to a specific egress trunkgroup). . The INVITE is sent by customer to the SBC (ingress), where some kind of process (which i have no details) will decide what IP:Port to forward the messages to Kamailio with additional infos (Kamailio is listening on different NICs/IPs/Ports). Then, Kamailio will check on the (external) routing engine the route to take and forwards to the outbound endpoint. Depending on the IP:Port that Kamailio receives the call (SBC may choose one NIC or the other) the request is then received by Kamailio and verified/processed by the routing engine replying back to kamailio with the route to take, and then Kamailio forwards to the outbound endpoint (egress SBC...). It is a bit confusing, and I want to simplify things as much as possible, but apparently the old/current solution is set this way and they pretend to keep the new solution as similar as possible to the old/current solution. This is mostly due to the billing process, which is very (overly?) complex and they do not pretend to change it. IMHO, the routing engine should be much simpler but it is a much more complex process the Carrier currently has...and wishes to keep. (sorry for posting without a subject. I did, however, re-posted with corrected subject) Atenciosamente / Kind Regards / Cordialement, *Sérgio Charrua* On Fri, Jul 12, 2024 at 8:11 PM Brett Nemeroff <brett(a)voicefoxtelephony.com> wrote: > Is there a reason you have to identify the trunk group based on received > IP/Port? I've seen this as a requirement from some more antiquated systems > and it's always a pain. It's always better to use source IP or even a trunk > prefix (dialed number prefix) which isn't really secure, but works when > security isn't an issue. > > If you have to do it that way, I'd probably turn the $Ri/$Rp into some > sort of cache key like $Ri_$Rp and map it to something like a dispatcher > setid. > > There may be a more modern way to do this, but I don't think there is. > > Good luck! > -Brett > > > On Fri, Jul 12, 2024 at 1:40 PM Sergio Charrua via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi all! >> >> for a project under development, one requirement is to be able to >> forward/relay the call to a specific destination depending on which IP >> Address and Port the call was received by Kamailio. >> This also means that Kamailio will be listening on multiple IP Address >> and Ports >> listen=udp:IP_1:port_1 # trunkgroup 1 >> listen=udp:IP_2:port_2 # trunkgroup 2 >> listen=udp:IP_3:port_3 # trunkgroup 3 >> >> I know the pv $Ri and $Rp can be used to identify from which IP address >> and port the message reached Kamailio and having those details, depending >> on the value, I can define some conditions that allows Kamailio to relay >> the call to different destination endpoints. >> For example: >> - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC >> address A.B.C.D >> - but calls from +32 to +39 received on trunkgroup2 need to go to Germany >> on SBC address 1.2.3.4 (as it is cheaper) >> >> Is there a better way of doing this without using IF/THEN/ELSE or SWITCH >> blocks and "hard code" destination SBC addresses? >> I have read the DRouting module's documentation but not sure if it could >> help and how... >> >> Any suggestions? >> >> >> >> Atenciosamente / Kind Regards / Cordialement, >> >> >> *Sérgio * >> __________________________________________________________ >> 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: >> > > > -- > > ======================================================== > Brett Nemeroff > Voice Fox Telephony LLC > Office: 512-670-8369 > Email: brett(a)voicefoxtelephony.com >

1 0

Change the To Header in the failure_route
by sadik.oualla.mohamed＠gmail.com 12 Jul '24

12 Jul '24

Hello, I need help with modifying the "To" header in the "failure_route[ID]{...}". Specifically, I want to add the "to tag" parameter from the previous SIP messages of the current dialog(e.g., 180 Ringing, PRACK or final responses). Here's the scenario: 1. Kamailio receives and relays the SIP messages: INVITE, 180 Ringing, PRACK, 200 OK, 486 Busy Here. 2. In my case, the 180 Ringing includes the "to tag" parameter. 3. When Kamailio receives the 486 response, it triggers "failure_route[ID]" using "t_on_failure(ID)" method. 4. I extract the SIP response code and reason from the final response of the INVITE transcation in this failure_route. 5. After processing and applying some instructions, I send a reply using "t_send_reply" with the code and reason extracted. The problem: - The failure_route[ID] holds the initial INVITE, which has no "to tag" parameter. - When I send a reply using "t_send_reply()", Kamailio generates a random "to tag" different from the previous SIP messages, which confuses my UAC. I think adding the "to tag" parameter manually might be a solution, but I'm not sure if I'm doing it correctly or if there is a better approach. I also tried using uac_replace_to() method, but it doesn't work in the failure_route[ID]. How can I make sure the reply in failure_route[ID] uses the correct "to tag" from the previous SIP messages? Are there any other possibilities to achieve this? Any assistance would be appreciated. Thank you!

5 11

Kemi Python3 throws a TypeError
by Michel Pelletier 12 Jul '24

12 Jul '24

Hello, Please help. I am using Kemi with the python3 module. I get random TypeError exceptions thrown within the Kamailio code. For example I have the following code: event = str(KSR.pv.get("$hdr(Event)")) log.logInfo( "KamailioStateful.ksr_request_route_loopback(): " + "Passing the request to the presence " + "publish handler for event: " + event ) rc = KSR.presence.handle_publish() if rc != 1: log.logError( "KamailioStateful.ksr_request_route_loopback(): " + "The presence publish handler returned " + "an error code: " + str(rc) ) This is what I see in the logs: 21(120) ERROR: PY3 {PUBLISH}: app_python3 [python_support.c:167]: python_handle_exception(): apy_exec: ksr_request_route((null)): Unhandled exception in the Python code: TypeError: expected bytes, str found The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 91, in ksr_request_route return ksf.ksr_request_route(msg, "Loopback") ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 100, in ksr_request_route return self.ksr_request_route_loopback(msg) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 240, in ksr_request_route_loopback rc = KSR.presence.handle_publish() ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <built-in function handle_publish> returned a result with an exception set If I put the offending line within a Try/Except it hides it but I would rather not see it not happen at all. Any Kemi Python3 users experiencing something similar? What could be causing this? Regards, Michel Pelletier

2 1

routing call depending on which IP address SIP message reached Kamailio
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! For a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

2 3

(no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! for a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

1 0

deb package build error (v5.7.6) [error: debian/rules binary subprocess returned exit status 2]
by Denys Pozniak 12 Jul '24

12 Jul '24

Hello! I'm trying to build standard deb packages for Kamailio v5.7.6 for Debian 12.6, but I get an error. By the way, there are no problems with building v5.7.2 ... #git clone --single-branch --branch 5.7.6 https://github.com/kamailio/kamailio.git /usr/src/kamailio #cd /usr/src/kamailio && ln -s pkg/kamailio/deb/bookworm debian #make cfg && RADCLI=1 make deb .... Makefile.defs defs skipped make[6]: 'libkamailio_ims.so.0.1' is up to date. gcc -shared -g -m64 -Wl,-O2 -Wl,-E -Wl,-z,relro -pthread -rdynamic "-ldl" -Wl,-Bsymbolic-functions cmd.o ims_ipsec_pcscf_mod.o ipsec.o sec_agree.o spi_gen.o spi_list.o spi_list_tests.o -lm -lmnl -L../../lib/ims/ -lkamailio_ims -Wl,-rpath,/usr/src/kamailio/src/lib/ims -o ims_ipsec_pcscf.so make[4]: Leaving directory '/usr/src/kamailio/src' make[3]: Leaving directory '/usr/src/kamailio' make[2]: Leaving directory '/usr/src/kamailio' *dpkg-buildpackage: error: debian/rules binary subprocess returned exit status 2* make[1]: Leaving directory '/usr/src/kamailio/src' -- BR, Denys Pozniak

2 3

New Member Introduction
by Maria Jonas 11 Jul '24

11 Jul '24

Hi everyone, I am new to this forum and excited to be here. I'm eager to learn more about Kamailio and its features and I'm looking forward to participating in discussions here. Could someone please guide me on how to ask questions here ? Where should I post if I have a query ? Looking forward to your advice and connecting with you all. Thanks,

2 1

usrloc ka_interval timer is not correct in kamailio 5.8
by Michal Popovič 11 Jul '24

11 Jul '24

Hello everybody, I am using kamailio in AWS behind load balancer and after upgraded from version 5.7 to 5.8 kamailio is not able to keep connection open. After quick investigation I found out that usrloc module got new parameter ka_interval that replaced timer_interval. Change is here. https://github.com/kamailio/kamailio/commit/53ace443020075f7ecd12c72f154193… Problem is that this interval is not reliable, as it is most probably still called only when timer_interval is executed. If I set values to timer_interval = 30 ka_interval = 30 ka_randomize = 20 Maximum time for keep-alive is not 30+20 seconds but 30+30+20 seconds. That is too late for AWS load balancer. I have fixed it by enabling tcp keepalives on kernel level. Is there any way how to specify exact time in version 5.8? Thanks. Michal

1 0

HTTP Client configuration path
by Sergio Charrua 11 Jul '24

11 Jul '24

Hi all! I am using HTTP_CLIENT module, and have created a http configuration file. The module's config_file parameter is set as: modparam("http_client", "config_file", "/etc/kamailio/http_client.cfg") and it works OK on my DEV environment, because there is a symlink /etc/kamailio pointing to physical folder /usr/local/etc/kamailio/ (using Kamailio 5.7.4) However in our integration environment (also using Kamailio 5.7.4), which is identical to production, and due to limitations on access rights, we cannot create a symlink /etc/kamailio pointing /usr/local/etc/kamailio/, so all cfg files are in the installation folder. So I thought that the config_file parameter should be a relative path, starting with " ./ " or by just setting the config file name, without path (as per documentation at http_client (kamailio.org) <https://kamailio.org/docs/modules/5.1.x/modules/http_client.html#idp38265140> . The issue is that even though the file http_client.cfg is present in /usr/local/etc/kamailio/, as well as other cfg files, the cfg_parser fails to load it: Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/cfg_parser.c:632]: cfg_parser_init(): cfg_parser: Unable to open file > './http_client.cfg' > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [curlcon.c:782]: http_client_load_config(): Failed to init http_client > config file parser > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [http_client.c:307]: mod_init(): Failed to load http_client connections > from [./http_client.cfg] > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/sr_module.c:1030]: init_mod(): Error while initializing module > http_client (/usr/local/lib64/kam > ailio/modules/http_client.so) I have set the parameter to > modparam("http_client", "config_file", "./http_client.cfg") and also to > modparam("http_client", "config_file", "http_client.cfg") but it only works if I add it as : > modparam("http_client", "config_file", > "/usr/local/etc/kamailio/http_client.cfg") or by adding > chroot="/usr/local/etc/kamailio" in the main cfg script, which was my final solution. Also checked for permissions, and all is fine. Note that this only seems to happen with http_client config_file parameter. I have other include commands in the main cfg file with relative paths and all works fine. I have also analysed the cfg_parser.so file, which loads the configuration file for http_client module, but could not find anything wrong with it (at 1st glance....) What am I doing wrong? Any clue? Atenciosamente / Kind Regards / Atentamente / Cordialement, *Sérgio *

1 0

SIP Question: B2BUA not ACKing 200 OK response
by Ghulam Mustafa 11 Jul '24

11 Jul '24

Hi Community, We are setting up a new provider in our VoIP setup for outgoing calls (FreeSWITCH --> Kamailio --> Provider). When the provider sends a 200 OK in response to our INVITE, FreeSWITCH's SIP stack discards the message, and we are unable to determine the cause of this behavior. I am sharing the SIP message exchange below. Could any SIP experts assist us in understanding why FreeSWITCH is not sending an ACK in response to the 200 OK message? We understand that the issue is not related to Kamailio but seems to be within the SIP stack or FreeSWITCH itself. Below are the actual INVITE and its response, which was discarded by FreeSWITCH: send 1203 bytes to tcp/[10.xxx.x.30]:5060 at 15:48:23.001054: ------------------------------------------------------------------------ INVITE sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport;branch=z9hG4bKD8p0KmvB8ZKma Max-Forwards: 69 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:gw+commsgroup@10.xxx.x.10:5080;transport=tcp;gw=commsgroup> User-Agent: FreeSWITCH-mod_sofia/1.10.11-release~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 347 X-FS-Support: update_display,send_info Remote-Party-ID: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10 >;party=calling;screen=yes;privacy=off v=0 o=Z 5395246197255529491 693195311 IN IP4 10.xxx.x.225 s=Z c=IN IP4 10.xxx.x.225 t=0 0 m=audio 55319 RTP/AVP 106 9 98 101 0 8 3 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-mux tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:8400 outgoing_send() nta: sent INVITE (85714459) to tcp/10.xxx.x.30:5060 tport.c:4191 tport_pend() tport_pend(0x150e6c0291c0): pending 0x150e6c0c52f0 for tcp/10.xxx.x.30:5060 (already 2) nua_session.c:4131 signal_call_state_change() nua(0x150d9c107460): call state changed: init -> calling, sent offer soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x150e6c0c0210, [0x150e91772798], [0x150e91772790], [(nil)]) called nua_stack.c:299 nua_stack_event() nua(0x150d9c107460): event i_state INVITE sent nua_stack.c:389 nua_application_event() nua: nua_application_event: entering nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering 2024-07-09 15:48:22.985772 100.00% [DEBUG] sofia.c:7493 Channel sofia/external/+86xxxxxxxxxxx entering state [calling][0] nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA going to sleep tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0cb390 from (tcp/10.xxx.x.30:5060) has 352 bytes, veclen = 1 recv 352 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:23.004029: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport=37555;branch=z9hG4bKD8p0KmvB8ZKma;received=10.xxx.x.10 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Server: Kamailio 5.8 Content-Length: 0 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0cb390 (352 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 100 Trying for INVITE (85714459) nta.c:3445 agent_recv_response() nta: 100 Trying is going to a transaction nta.c:9697 outgoing_estimate_delay() nta_outgoing: RTT is 3.038 ms tport.c:4253 tport_release() tport_release(0x150e6c0291c0): 0x150e6c0c52f0 by 0x150e6c0be9a0 with 0x150e6c0cb390 (preliminary) tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:5904 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x150e91973a50) nta.c:7270 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 2/3 free nta.c:1310 agent_timer() nta: timer not set tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0c9d90 from (tcp/10.xxx.x.30:5060) has 1195 bytes, veclen = 1 recv 1195 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:27.070278: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TCP 10.xxx.x.10:5080;received=10.xxx.x.10;rport=37555;branch=z9hG4bKD8p0Km From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx >;tag=7bQ0OSl9Dl-39455-57712329-1057554-14 Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:127.0.0.8;line=sr-H46V9.fbsXNDlkfEJjGbujK3J4aDo1GzP5GOJ8hcsXwV9LAWw1abHkqWlXczwL7*> Record-Route: <sip:13x.xxx.xxx.xx;transport=tcp;lr;ftag=ZSvBBtypKypDS;did=cbd.e581;nat=yes> Supported: sdp-anat Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY Require: timer Session-Expires: 2000;refresher=uas Server: Mediant VE SBC/v.7.40A.250.265 P-Asserted-Identity: <sip:4006100666@hk.chinatelecom.com;user=phone> Accept: application/sdp, application/isup, multipart/mixed, application/dtmf, application/dtmf-relay Content-Type: application/sdp Content-Length: 219 User-Agent: Kamailio 5.8 v=0 o=- 846721169 1919912032 IN IP4 13x.xxx.xxx.xx s=- c=IN IP4 13x.xxx.xxx.xx t=0 0 m=audio 49708 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:49709 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0c9d90 (1195 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 200 OK for INVITE (85714459) nta.c:3494 agent_recv_response() nta: 200 OK was discarded Any insights or suggestions would be greatly appreciated, Specifically, why is the 200 OK response being discarded?. Thank you! Regards, Mustafa

2 2

Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 11 Jul '24

11 Jul '24

Hi, Daniel. Thank you for the quick response! I will update you after testing the fix. Regards, Vanderlei

1 0

Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 10 Jul '24

10 Jul '24

Hello Henning, Thank you for your quick response. The increase happens in the core module (create-avp). See bellow: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 <---------------------------- counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 <------------------ } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } These reports above are shown after the end of calls (4000) where the outgoing INVITE is not answered. In normal calls the issue does not occur and the reports are shown below: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 430304 } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 6", "shmem:free_size = 64143184", "shmem:max_used_size = 4070816", "shmem:real_used_size = 2965680", "shmem:total_size = 67108864", "shmem:used_size = 2627856" ], "id": 683948 } Regars, Vanderlei

2 1

Intercepting OPTIONS from nathelper
by Denys Pozniak 10 Jul '24

10 Jul '24

Hello! I need advice on how to intercept local SIP OPTIONS requests from the nathelper module for some adjustments? As far as I tested, it does not fall into the *event_route[tm:local-request]* Thanks in advance! # kamailio -v version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty -- BR, Denys Pozniak

2 5

handle_subscribe(), handle_publish() and DB usage
by Володимир Іванець 04 Jul '24

04 Jul '24

Hello everyone! According to the documentation functions handle_subscribe() and handle_publish() update information not only in memory but also in the database. Isn't there a way to disable database usage for this functionality? Setting subs_db_mode to 0 does not make a change. Thank you!

1 0

Contact does not exist in PCSCF usrloc error in log
by Shantanu K Roy 03 Jul '24

03 Jul '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu Kumar Roy

3 6

Kamailio v5.6.6 Released
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, Kamailio SIP Server v5.6.6 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.6. Note that 5.6 is now the third last stable branch, v5.6.6 being the last planned released in 5.5.x series. The latest two stable branch are 5.7 and 5.8, with v5.8.2 being released a while ago. For more details about version 5.6.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-6-6-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 and Development Services Kamailio Advanced Training -- asipto.com

1 0

Releasing v5.6.6
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, I am considering to release Kamailio v5.6.6 (out of branch 5.6) later this week (likely on Wednesday, July 3, 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. This is going to be the last release out of branch 5.6 to mark the end of official/packaging maintenance. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

Extracting and Handling Via Headers in failure_route
by sadik.oualla.mohamed＠gmail.com 02 Jul '24

02 Jul '24

Hello, I am currently working on a scenario where I need to use Kamailio in the failure_route to process SIP responses, specifically handling negative SIP replies (e.g., 486 Busy Here) that Kamailio may receive. Here is the topology: ##B2BUA (acts as a UAC with the Kamailio, because it sends the INVITE) -> Kamailio -> Callee## In my setup, Kamailio needs to extract all Via headers from the received SIP response in the failure_route, focusing particularly on the bottom Via header (from the B2BUA' UAC part). However, I am encountering difficulties in achieving this. Here is what I have tried so far: 1. To extract the Via headers, I used the following line of code in the failure_route: ```$var(viaHeaderValues) = $T_rpl($hdr(Via));``` However, this only returns the topmost Via header. 2. I attempted to get the first and last Via headers values with: ```$var(viaHeaderValueFirst) = $T_rpl($hdr(Via)[0]); $var(viaHeaderValueLast) = $T_rpl($hdr(Via)[1]);``` Unfortunately, this also only returns the topmost Via header, which contains the address of Kamailio. My goal is to make sure that when Kamailio sends the SIP reply to the UAC with send_reply method on the failure_route, it uses the extracted *Via header* of the incoming SIP response (only the bottom one). Here is an example scenario: - Received SIP Response in kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.4;branch=z9hG4bK9157.090080a91105c0ec6279bb56882d1dc8.0 Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` - Desired SIP Response from kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` The current issue is that Kamailio modifies the Via header when sending the reply with send_reply method, resulting in: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m;received=200.200.200.3 From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` Due to this change, the UAC does not correctly react to or understand the SIP response "SIP Reply follows the RFC standards, but my current UAC waits for a Via that looks like Via of the previous SIP messages in the structure of parameters.". It is crucial for the Via header to maintain the same important order such as the branch, received, and rport values as the previous exchanged SIP messages. Could you please provide guidance on how to force Kamailio to send the SIP reply using send_reply method with the extracted bottom Via header from the incoming SIP response? Any advice or examples on handling this situation would be greatly appreciated. Thank you for your time and assistance.

1 0

Kamailio v5.7.6 Released
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, Kamailio SIP Server v5.7.6 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.6. For more details about version 5.7.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-7-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.2 was released on June 12, 2024. 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 v5.7.6
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, I am considering to release Kamailio v5.7.6 (out of branch 5.7) on Tuesday, July 2, 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

SRV in advertise?
by David Villasmil 01 Jul '24

01 Jul '24

Hello guys, I'm trying to use an SRV as an advertised address but kamailio fails to start. Is this not possible? Thanks, David Villasmil email: david.villasmil.work(a)gmail.com

4 7

SecSIPID Assistance
by Blake Ivey 01 Jul '24

01 Jul '24

Hi everyone. Wanting to see if someone could point me in the right direction. Still very knew to Kamailio but I am beginning to understand it better. I'm making an outbound proxy and have everything working well besides stir/shaken. I'm looking at the module page and have went back and forth with chatGPT and can't seem to figure this part out. I keep getting errors on the modparam lines. Obviously this is a self signed cert because I'm just testing. I am able to reach and download the cert from the Web server. Thank you for any assistance. # SECSIPID for Stir/Shaken modparam("secsipid", "private_key", "/etc/kamailio/secsipid/private.key") modparam("secsipid", "certificate", "/etc/kamailio/secsipid/cert.crt") modparam("secsipid", "authority_cert", "/etc/kamailio/secsipid/ca.crt") modparam("secsipid", "expire", 600) modparam("secsipid", "timeout", 2) route[STIRSHAKEN] { if (is_method("INVITE")) { if (!secsipid_add_identity("$fU", "$rU", "A", "", " http://myIPaddress.com/stir_shaken_cert.crt <http://myipaddress.com/stir_shaken_cert.crt>", "/etc/kamailio/secsipid/private.key")) { xlog("L_ERR", "Failed to sign call with ID: $ci - From: $fU\n"); send_reply("500", "Internal Server Error"); exit; } else { xlog("L_INFO", "Successfully signed call with ID: $ci - From: $fU\n"); } } # Relay the call after signing route(RELAY); }

3 8

removing Via Header with received=
by mosbah abdelkader 30 Jun '24

30 Jun '24

Hello All, I am trying to build a simple proxy that forwards SIP requests to a remote portaone server. But, kamailio is adding a Via header with received=@IP to replies got from portaone. How to disable this behavior as Asterisk (client in my setup) is not accepting this header. Thank you.

1 0

misctest module not found in kamailio debian packages
by Patrick Karton 29 Jun '24

29 Jun '24

Hello, how do i get misctest module in debian packages ? it seems not included in a package. is compilation the only way to get it ? im using kamailio 5.7.5 on debian (bullseye) 11.8 version: kamailio 5.7.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 Thanks.

1 0

Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org
by Xenofon Karamanos 28 Jun '24

28 Jun '24

Any news on regarding the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thanks, Xenofon ________________________________ From: Xenofon Karamanos via sr-dev <sr-dev(a)lists.kamailio.org> Sent: Wednesday, June 19, 2024 11:51 To: sr-dev(a)lists.kamailio.org Cc: Xenofon Karamanos <xk(a)gilawa.com> Subject: [sr-dev] Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org Hey everybody, While attempting to use the official Ubuntu package for Kamailio version 5.7.4 on Ubunut 24.04, we encountered an issue where TLS functionality does not seem to work as expected but 5.8 branch does. Given this, I am inquiring about the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thank you in advance for your assistance. Best regards, Xenofon

1 0

replies using the wrong protocol
by sarah.martin＠nextiva.com 27 Jun '24

27 Jun '24

I am extremely new at this, but trying to set up TLS with a carrier. TLS connection is good, Invite goes out, we get the 100 and the 200, but subsequent messages (ACK and BYE) are being sent with UDP and I cannot figure out how to get them to maintain the TLS transport. Any suggestions? I think this is the section I'm looking for. # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { xdbg("rtpengine_manage loop-protect MANAGE_REPLY"); #rtpengine_manage("loop-protect"); } }

6 21

Kamailio cannot forward request - invalid sendtoparameters
by serkan celik 27 Jun '24

27 Jun '24

Hi all, I face a problem when trying to forward an UPDATE request to the IMS which communicates with UDP transport protocol on one side UDP and SCTP on other side. When I try to send UPDATE towards the UDP side of IMs I see following error on kamailio side and request cannot be forwarded Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: ERROR: <core> [core/udp_server.c:597]: udp_send(): sendto(sock, buf: 0x7fc180deb5f0, len: 1509, 0, dst: (172.18.0.20:5060), tolen: 16) - err: Invalid argument (22) Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: CRITICAL: <core> [core/udp_server.c:603]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net The UPDATE message which is tried to be forwarded is as follow: Here there are double rr one is UDP and one is sctp (Route: < sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>) ) That would be great if you can help why this UPDATE cannot be forwarded. By the way double rr is on on kamailio config Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: DEBUG: <core> [core/forward.c:577]: forward_request(): Sending:#012UPDATE sip:alice@10.10.141.188;ob;sbc-id=xpllrk47odqh4pb2ofpib6vsym SIP/2.0#015#012Via: SIP/2.0/UDP 127.0.0.1:50601;branch=z9hG4bKddab.3554948ce3cfe321729448a2b082c2f3.0#015#012Via: SIP/2.0/UDP 127.0.0.1:7060;branch=z9hG4bKnHN8Q3ya#015#012To: < sip:alice@ims.wingcon.com>;tag=49IZjY2irfb#015#012From: < sip:bob@ims.wingcon.com>;tag=hfdqjj2pmui#015#012Call-Id: 3mzzecWt3Cbgvy#015#012CSeq: 1 UPDATE#015#012Max-Forwards: 70#015#012Contact: <sip:bob@10.10.141.188;ob;sbc-id=o7dpxe7qu2r34zwzio5pis2vqi>#015#012Content-Type: application/sdp#015#012Route: <sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, <sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;transport=sctp;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.120;lr>#015#012Session-Expires: 180;refresher=uas#015#012Supported: replaces, 100rel, timer, norefersub#015#012Min-SE: 90#015#012P-Charging-Vector: icid-value=e277783432fd11efba03fa163eed425f;icid-generated-at=172.18.0.120;orig-ioi=172.18.0.120;term-ioi=172.18.0.120#015#012User-Agent: PJSUA v2.14 Linux-5.4.0.172/x86_64/glibc-2.35#015#012Content-Length: 409#015#012#015#012v=0#015#012o=- 3928313948 3928313950 IN IP4 172.18.0.120#015#012s=pjmedia#015#012b=AS:84#015#012t=0 0#015#012a=X-nat:0#015#012m=audio 30106 RTP/AVP 8 96 97 121 120#015#012c=IN IP4 172.18.0.120#015#012b=TIAS:64000#015#012a=ssrc:1986855169 cname:1fca52e812ced466#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 AMR/8000#015#012a=rtpmap:97 AMR-WB/16000#015#012a=rtpmap:121 telephone-event/16000#015#012a=fmtp:121 0-16#015#012a=rtpmap:120 telephone-event/8000#015#012a=fmtp:120 0-16#015#012a=sendrecv#015#012a=rtcp:30107#015#012. Regards Serkan

1 0

Handling final response retransmissions after a reInvite
by hollis.rhonda＠gmail.com 26 Jun '24

26 Jun '24

I have a scenario where Kamailio is receiving a retransmit of a 200 OK to a late offer INVITE after it has sent the ACK, and after it has begun to handle a reInvite from the calling party. This results in the following commands to the rtpengine: Offer (reInvite), Offer (200 retransmit), Answer (ACK to retransmitted 200). At this point, I see STUN binding errors both in Chrome, where the webRTC client (called party) is running, and in the rtpengine logs. ERROR:port.cc(498)]: Received STUN BINDING error response: class=4 number=1 reason='Unauthorized' [core] STUN authentication mismatch from x.x.x.x:63396 [ice] Received ICE/STUN response code 487 for candidate pair dl2efuRG06eK4nFs:2713745946:1 from x.x.x.x::64509 to x.x.x.x [ice] ICE role change, now controlled [ice] Recalculating all ICE pair priorities [ice] Triggering check for dl2efuRG06eK4nFs:2713745946:1 [ice] ICE/STUN response with unknown transaction received (from x.x.x.x:64509 on interface x.x.x.x:16184) So, briefly, the call flow up to the point of error looks like this: INVITE w/o SDP 200 OK w/SDP, Offer to rtpengine ACK w/SDP, Answer to rtpengine reInvite w/SDP, Offer to rtpengine retransmission of 200 OK, Offer to rtpengine ACK to retransmitted 200, Answer to rtpengine I have dialog tracking enabled in my config, and it seems like Kamailio is behaving as expected when the retransmitted 200 and ACK are forwarded on, but I'm curious if other people have run into a scenario like this and how they have dealt with it.

1 0

Need help on how to implement presence_xml
by Michel Pelletier 25 Jun '24

25 Jun '24

Hello, I need to implement Event: presence, application/pidf+xml as described in RFC-3856. I understand I need to use the presence_xml module and generate PUBLISH requests to update the information and ensure that the NOTIFY get sent. I know nothing abour XCAP or wether I need it or not. Any information source out there that could help me? Regards, Michel Pelletier

1 0

TLS: client doesn't check certificate CN
by Marat Gareev 20 Jun '24

20 Jun '24

Hello, I noticed that Kamailio can route either to IP and FQDN. This means the server certificate CN is not checked by the client. How to enable something like the 'verify_peer' option? Thanks

1 0

Possibility to print values of Core Keywords in Kamailio configuration file
by sadik.oualla.mohamed＠gmail.com 20 Jun '24

20 Jun '24

Hello everyone, I have a question regarding the usage of core keywords, specifically the possibility of printing their values in log messages. While I understand that core keywords such as 'dst_port' and 'af' etc... can be accessed directly in if conditions, I am interested to know if there is a way to print the values of these core keywords for logging purposes. Thank you.

4 6

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Wed, 19 Jun 2024 11:54:05 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > What you should expect using this method is blocking while the exec runs > which could run you out of child processes while they complete. I'd also > expect to see an "unusual amount of CPU activity" which will be the result > of the forked shells being created and destroyed. > > All of that being said, yes it works. It's architecturally overly simple > and very inefficient. If your load is tiny, it might be ok. Anything > moderate like Henning and others have said, will use an inordinate amount > of cpu. I would not personally risk it. > > Thank you Brett, Henning and Alex Sadly the load won't be tiny and cannot risk it. For now I'll deploy a local nginx and fpm to run the scripts, which are in php, and will use the http_async method. Sadly we have to "http all the things" nowadays. Let's see if I'm able to write down all the logic tomorrow and do some testing. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

Adding custom headers to HTTP server response xhttp
by Sasivarunan 19 Jun '24

19 Jun '24

HI Team, I use tls and xhttp module to enable websocket interface for webrtc client to connect. I wanted to add "Strict-Transport-Security" header to xhttp response to add security. But currently I don't see a way to add it anywhere. I can see only server_header as an option, But this will get added to SIP messages as well. It 'd be great if anybody has a suggestion around this. Thanks Varunan

2 2

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Tue, 18 Jun 2024 13:54:41 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > Just want to add that exec is heavy and slow. I would not recommend it. > > Is there a reason you want to do this over http-ifying your script and > using async? > > Not really. The scripts are provided. Maybe I'm against "http all the things" which seems to be what everyone does nowadays. Why would http client be lighter than executing a script? That would be a design problem, right? -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

exec concurrency
by Jon Bonilla (Manwe) 18 Jun '24

18 Jun '24

Hi all I've been working with async http client and async db queries in the past but now I have to execute scripts and store the return values in avps. Since I have no experience with that, I wonder how to achieve concurrency with that scenario. Is there a way to exec async or suspend until the exec has finished and resume later? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 2

Psresence dialoginfo question
by Stefan Mititelu 17 Jun '24

17 Jun '24

Hello, I am doing BLF but I want different users to be able to setup different dialog info initial states. My question is if there's an already implemented way for doing that? I've looked over docs and C code so far and this is what I've found: 1. modparam("presence_dialoginfo", "force_dummy_dialog", 1) => will force sending initial NOTIFY with dialog state "terminated" 2. Function calls: dlginfo_agg_nbody()->dlginfo_agg_nbody_empty() which will build the pre-defined xml body with that "terminated" state. If not implemented so far, maybe this can be used to set a different initial state for a user (e.g. export a cfg function or take that info from DB?) Any opinions on this? Thank you, Stefan

2 2

db_mysql startup warnings
by Juha Heinanen 17 Jun '24

17 Jun '24

When I start Kamailio master version that has been build on Debian 12 with default-libmysqlclient-dev depending on libmariadb-dev-compat, I get several warnings like this: 2024-06-17T14:48:27.702827+03:00 siika /usr/bin/sip-proxy[297246]: WARNING: db_mysql [km_my_con.c:187]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring How can I prevent them? -- Juha

1 1

How to Check Source Port in Application Layer with Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Jun '24

16 Jun '24

Hello kamailio lovers, I am currently working on a Kamailio configuration where I need to check the source port of incoming SIP messages. Initially, I used the $sp pseudo-variable to get the source port, which worked perfectly while testing with SIPp. Here is the snippets code that I use " if ($sp != 5080 && $sp != 8080 && $sp != 5060){ sl_send_reply(403, "forbidden"); xlog("this $rm received, is forbidden\n"); exit; } " However, I encountered an issue when I switched to using sipsak for testing. It seems that sipsak uses a randomly assigned port for the actual network connection, even though it sets the source port in the SIP headers (e.g., Via, To and Contact headers) to a specified port using the -l option. Here is the sipsak command that I use "sipsak -vv -s sip:jack@192.168.1.2:5060-l 5080" My goal is to have Kamailio check the source port as specified in the application layer, rather than the transport layer port (i.e., the ephemeral port assigned by the OS randomly like 54691, 33017...). Could anyone guide me on how to properly configure Kamailio to extract and check the port from the application layer or SIP headers, or am I doing this in the wrong way? Any advice or examples would be greatly appreciated. "I am not a native english speak, so I use chatGPT to make my questions clear" Thank you!

1 0

topos module issue on hangup
by Arun K R 13 Jun '24

13 Jun '24

Hello, I am using topos module to hide Record-Route and Route header as per the requirement from carrier. But the topos module hides, contact, via and routing headers also. Due to this reason when I am getting bye from the carrier I am not able to route the bye back to the agent. Bye from Agent to carrier is working fine. Bye from carrier to my internal network interface is not routing to client through public interface. I assume this may be because Via / Contact header has been hidden and changed by the topos module with the invite please help me for this Thanks Arun

3 5

Kamailio v5.8.2 Released
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, Kamailio SIP Server v5.8.2 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.1. 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 v5.8.2. For more details about version 5.8.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/06/kamailio-v5-8-2-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 v5.8.2
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, I am considering to release Kamailio v5.8.2 (out of branch 5.8) on Wednesday, June 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

2 2

New features in v.5.8
by Leonid Fainshtein 10 Jun '24

10 Jun '24

Hi, In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support for server (RFC7339)". Where can I find some more details about the feature? Also, the similar question about the new "influxdbc" module. What are the possible use cases for this module? Thank you, Leonid Fainshtein Xorcom Ltd

2 2

handling 300 redirect requests
by Rob Green 09 Jun '24

09 Jun '24

I am trying to proxy a request from server A to server C with me being server B. i send the invite from A to B, then i need to send to server R which returns back to me a 300 redirect, which i then need to take the answer from the Contsct and then proxy the original invite from A to C. server A can not handle 3xx responses and must be proxied. I havent been able to find any examples where the 300 is intercepted and a new invite to the target is sent. i have the following code but it results in no new INVITE and the original session is hung until it is canceled. Any idea what i may be doing wrong? In the example below i am just hardcoding the uri for the new invite instead of getting from the Contact as a debug exercise. I am using Kamailio 5.7.3 route[HANDLEREDIRECT] { xlog("L_INFO", "DBG: HANDLEREDIRECT"); msg_apply_changes(); $var(test) = $hdr(Contact); xlog("L_INFO","DBG: redirect headers [$var(test)]: from=[$fu] from_user=[$fU], to=[$tu] to_user=[$tU]"); # send to gateway $var(new_uri) = "sip:" + $tU + "@" + "gateway"; if ($var(new_uri) != "") { xlog("DBG: Redirecting to: $var(new_uri)\n"); $ru = $var(new_uri); $tu = $ru; record_route(); route(RELAY); exit; # i have also tried using #t_newtran(); # and #t_relay(); } else { xlog("DBG: No contact header found in 300 response\n"); } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("DBG: new branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "DBG: manage branch"); route(NATMANAGE); return; } onreply_route[MANAGE_REPLY] { xdbg("DBG: incoming reply\n"); xlog("L_INFO", "DBG: manage reply"); if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: got 3xx redirect"); route(HANDLEREDIRECT); drop; } return; } I also tried using uac_redirect module, but it stalls the incoming invite also, and i cant figure out how to modify the uri from the contact before sending (to add params) # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: following 3xx redirect"); get_redirects("3:1"); t_relay(); exit; } } Rob Green Chief Information Officer (CIO) Tel: 1-352-218-1490 Cell: 1-407-620-0932 Email: rob.green(a)nuwave.com Web: www.nuwave.com This communication is the Property of NUWAVE Communications and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in Error, Please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.

2 2

Setting Key/Value Pairs at Dialogue Level
by Pavan Kumar 08 Jun '24

08 Jun '24

Hi everyone, I am creating a dialogue when I receive INVITE. Is there a way to set key/value pair like attributes at dialog level, so that, when I receive BYE message I can retrieve that information and do some processing? Thank you, Pavan Kumar

2 2

Recommendation for a signaling B2BUA to work with Kamailio on Ubuntu
by Benoît Panizzon 07 Jun '24

07 Jun '24

Hi I wonder if anyone could maybe recommend what to use as signaling B2BUA to work with Kamailio to handle spiralling / looping calls and to reduce the number of Route, Via Header and message size, which cause issue with a large number of CPE. Requirements: * New CallID needs to be generated * Handling Route and Via * ~ 2000 concurrent calls. * Preserving custom SIP X- header * Routing NOTIFY for MWI etc. Handling RTP is not required! What I have been looking at more closely: = FreeSwitch = Got stuck trying to solve the dependencies to get it to run/build on ubuntu jammy. Has anyone experience in running it in a docker container? Will it perform? = Sippy = Very small, very simple, unfortunately removes customer SIP headers and does not forward NOTIFY. (It's python, so maybe this could be easily implemented/fixed). = SEMS = Does not seem to be actively developed in the last, couple of years. Looks quite bloated to me for what I need. = Asterisk = Each customer header has to be copied, wants to handle RTP and bridge by issuing Re-Invites. Not sure how to route stuff like NOTIFY. Also from experience I fear asterisk would translate sip reply codes (like 608 and 607) it does not know to some 4XX default. -- 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 ______________________________________________________

4 3

Filtering codes in db acc
by Jon Bonilla (Manwe) 06 Jun '24

06 Jun '24

Hi all I'm trying to make kamailio not to insert a row in missed_calls table when reply code is 407. But even if I set the "failed_filter" modparam the 407 row is inserted. The acc mmodparams are: modparam("acc", "db_table_acc", "kam_acc") modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACC_MISS) modparam("acc", "db_url",DBURL) modparam("acc", "db_insert_mode", 1) modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "failed_filter", "401,407") modparam("acc", "db_extra", "whatever" db is mysql. Kamailio version is 5.7.4 When I send a call that receives a 407 and after auth a 480 y get both rows in missed_calls table. Log: DEBUG: acc [acc_mod.c:404]: mod_init(): acc db table initialized to: kam_acc DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 401 DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 1 = 407 DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cec807480) event type 512, reply code 407 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 407 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cebf59bc0) event type 512, reply code 486 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 486 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off am I configuring this wrong? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 3

CentOS 7 and 8 deprecated
by Sergey Safarov 06 Jun '24

06 Jun '24

The CentOS project page shows that the 7 and 8 releases are deprecated. https://www.centos.org/ Does the Kamailio community use these releases for prod? If yes then I will keep the packaging for this release. If not, I will remove the packaging for 7 and 8 releases, and in the future Kamailio will be packaged for 9 and 10 releases (planned). https://www.phoronix.com/news/CentOS-Stream-10-Start https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre… https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre…

2 1

Assistance Needed: Converting 11-Digit US Numbers to E.164 in Kamailio
by Pavan Kumar 04 Jun '24

04 Jun '24

Hey everyone, I have the following setup: External User/Trunk <===> Kamailio <===> FreeSWITCH I have configured FreeSWITCH and Kamailio to work only with phone numbers in E.164 format. Recently, I needed to integrate a new SIP trunk that sends phone numbers in an 11-digit US format. My initial idea to solve this is to use Kamailio as a translator that converts 11-digit numbers to E.164 when sending to FreeSWITCH, and back to 11-digit when sending to the trunk. Before writing the code, I want to know if there is already a solution to this problem in one of the modules, so that I won't reinvent the wheel. Thanks, Pavan Kumar

7 11

DMQ_USRLOC and multiple domains
by Lợi Đặng 03 Jun '24

03 Jun '24

Hi, I'm setting up shared user location for 2 instances of kamailio via dmq_usrloc. Everything seems fine except my setup is multi-domains. The issue arises when the producer node produces the Contact of domain X, Contact of domain Y separately, but the consumer node stores both as the Contact of domain Y (seen in kamcmd ul.dump), in which Y is set in usrloc_domain. modparam("dmq_usrloc", "usrloc_domain", "{{domain_Y}}") modparam("usrloc", "use_domain", 1) ==== I had a quick ngrep and see the domain was put in the "aor" key of the body in KMQ message for usrloc {"action":1,"aor":"test_username_0dppxcl@$domain_X"} But I'm not sure how to make my consumer node to store the contact in appropriate domain. Any help will be appreciated. Thanks and regards, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

1 1

ACK and Bye not relaying when record_route is not used
by Arun K R 03 Jun '24

03 Jun '24

Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call) My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side. Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: <sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11> Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248 v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101 ------------------------------------------------------------------------------------------------------- The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0 Thanks Arun

3 3

StriShaken module compilation
by Sergio Charrua 01 Jun '24

01 Jun '24

Hello all! I am struggling on compiling StriShaken module on RHEL 9.2. So far, the module was compiled as well as libstirshaken. Below are the steps used to compile (as sudo) both library and module (I hope this may help someone) and, of course, correct me if I am wrong in any step or if there is a better way. Also, note that the RHEL 9,2 is in a VM environment without access to the WWW, instead it uses Red Hat Satellite to download packages (via Yum or DNF), if available. Lib LibKS download libks from https://github.com/signalwire/libks unzip move to dir created by Unzip run yum groupinstall "Development Tools" dnf install libuuid-devel libatomic openssl-devel cmake . make make install cp /usr/lib/pkgconfig/libks2.pc /usr/lib64/pkgconfig/. For module LibStirShaken: download https://github.com/signalwire/libstirshaken unzip move to dir if using OpenSSL3.0 or + edit file configure.ac and add after line 28: if test x$HAVE_OPENSSL = x1; then openssl_CFLAGS="$openssl_CFLAGS -DHAVE_OPENSSL"; /* PATCH FOR OPENSSL3 */ AC_MSG_CHECKING([for OpenSSL >= 3.0.0]) AC_COMPILE_IFELSE([AC_LANG_PROGRAM([[ #include <openssl/opensslv.h> #if OPENSSL_VERSION_PREREQ(3,0) #error "you_have_version_3" #endif ]], [[]])], [ AC_MSG_RESULT([no]) ], [ AC_MSG_RESULT([yes]); AC_DEFINE(OPENSSL_SUPPRESS_DEPRECATED, 1, [disable openssl deprecated-function warnings]) ]) /* END OF PATCH */ else AC_MSG_ERROR([OpenSSL >= 1.0.1e and associated developement headers required]) fi run configure make sudo make install next, move to dir /root/kamailio/kamailio-5.7.4/src/modules/stirshaken cd /root/kamailio/kamailio-5.7.4/src/modules/stirshaken ln -s /root/stirshakenlib/libks-master/src/include/libks libks make make install All seem to compile OK , *but* when starting Kamailio I get the following errors: May 29 17:42:02 kamailio1 kamailio[248991]: ERROR: <core> [core/sr_module.c:599]: ksr_load_module(): could not open module </usr/local/lib64/kamailio/modules/stirshaken.so>: libstirshaken.so.1: cannot open shared object file: No such file or directory May 29 17:42:02 kamailio1 kamailio[248991]: CRITICAL: <core> [core/cfg.y:3915]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_stir_shaken_mod.cfg, line 106, column 12-26: failed to load module I haven't found a solution for this. Can anyone help? Thanks in advance! *Sérgio Charrua*

2 1

Parsing Error with SIP 607 Response
by Ghulam Mustafa 31 May '24

31 May '24

Hello Kamailio Community, I'm encountering a parsing error in Kamailio when receiving a SIP 607 "Unwanted" response. The relevant log entries are: May 31 12:32:09 k-proxy1 /usr/sbin/kamailio[320898]: ERROR: <core> [core/receive.c:378]: receive_msg(): core parsing of SIP message failed (36.53.xx.xx:5060/1) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 11) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<SIP/2.0 607#015#012Via: SIP/2.0/UDP 134.195.195.40:5060;branch=z9hG4bKc87a.ed6403ae1529f36b65171bf964604be0.0;i=2#015#012Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFIzl5c9wbfNG3oaHkl3#015#012From: "15516756486" <sip:15516756486@10.x.x.10>;tag=KXtSBZ1at71Hc#015#012To: <sip:99904441743709685@36.53.xx.xx;transport=udp>;tag=sbcsipuas_1_C35684_20240531083209909_sbc06#015#012Call-ID: a40d965d-99ec-123d-6f92-ecf4bbd64f6c#015#012Contact: <sip:36.53.xx.xx:5060>#015#012CSeq: 84025572 INVITE#015#012Server: sbc_5#015#012Content-Length: 0#015#012#015#012> And the actual SIP message is U 36.53.xx.xx:5060 -> 10.x.x.30:5060 #850 SIP/2.0 607. Via: SIP/2.0/UDP 134.195.195.40:5060 ;branch=z9hG4bKbdbd.95060a558d66b77d258b8d0799be4cf0.0;i=2. Via: SIP/2.0/UDP 127.0.0.8 ;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFCaHXyLA3G.SjwaAmdn. From: "13xxxxxxxxxx" <sip:13xxxxxxxxxx@10.x.x.10>;tag=pFFatBpUc75BH. To: <sip:441xxxxxxxxx@36.53.xx.xx ;transport=udp>;tag=sbcsipuas_1_C56639_20240531081932874_sbc04. Call-ID: e0d2f5fa-99ea-123d-6f92-ecf4bbd64f6c. Contact: <sip:36.53.xx.xx:5060>. CSeq: 84025194 INVITE. Server: sbc_5. Content-Length: 0. Any insights or suggestions would be greatly appreciated. Thank you! Mustafa

5 9

Access avp vars in a cancel? How to pass information from an INVITE to a CANCEL?
by Benoit Panizzon 31 May '24

31 May '24

Hi To determine, if rtpengine was engaged during an invite or not, I make use of rr params for all messages in following transactions and I set an avp for all messages in the current transaction. When the call is canceled, I would need to call rtpengine_delete to make sure rtpengine stops listening for rtp packets. Unfortunately, the avp I set during the INVITE, does not seem to be present while the CANCEL message is processed, despite to my understanding being in the same transaction. How can I pass some information from an INVITE to the corresponding CANCEL? 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 10

core dump on AWS with ~4000 TCP sockets
by David Villasmil 31 May '24

31 May '24

Hello all, I'm seeing core dumps (can't find them!) on AWS with lots of TCP sockets open (not related to load), kamailio dies randomly with: *May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743739]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 142 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:788]: handle_sigs(): child process 743672 exited by a signal 11 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated May 30 14:32:03 sip-proxy /usr/sbin/kamailio[743542]: CRITICAL: <core> [main.c:698]: sig_alarm_abort(): shutdown timeout triggered, dying...* before this happens, there's a ton of May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.56.51.197:34846 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 98.196.165.187:38593 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:49372 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 97.101.162.40:45248 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.59.213.36:57074 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 207.195.63.14:2521 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:22155 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:10339 failed (timeout) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Re: Via header force change protocol to TLS?
by David Villasmil 31 May '24

31 May '24

trying to use msg_apply_changes(); subst_hf("Via","/TCP/TLS/","f"); But as i don't seem to have access the top-most via, i can'it doesn't work... i cam change the incoming Via, but not the outgoing... I need this urgently.. any help anyone? > Sergey, > > Thanks for pointing me to the PR but that's not exactly what i need, let > me explain: > > We have an AWS an NLB on which our clients connect to and terminate TLS > connections and from then on into our proxies it is TCP. Now when clients > send invites (with TLS in their VIA as protocol) and we reply to them > (because the socket is TCP) I need to set the protocol on the Via to be > TLS. Otherwise, the client would not understand. This message or will > believe that it's not. about the same connection > > I haven't figured out a way of doing this since it can't get the actual > VIA that will be sent out. I've tried on the onsend_route (enabling > onsend_route_reply) > > Help is greatly appreciated! > > > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > > > On Thu, May 30, 2024 at 11:44 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> HEllo Sergey, >> >> i can send one. yes. >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> >> >> On Thu, May 23, 2024 at 5:14 PM Sergey Safarov <s.safarov(a)gmail.com> >> wrote: >> >>> Hi David >>> Could you send PCAP for an inbound call via TCP connection? >>> >>> Sergey >>> >>> On Thu, May 23, 2024 at 5:53 PM David Villasmil < >>> david.villasmil.work(a)gmail.com> wrote: >>> >>>> it's still in progress though. >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work(a)gmail.com >>>> phone: +34669448337 >>>> >>>> >>>> On Thu, May 23, 2024 at 4:51 PM David Villasmil < >>>> david.villasmil.work(a)gmail.com> wrote: >>>> >>>>> Thanks, I'll check it out! >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work(a)gmail.com >>>>> phone: +34669448337 >>>>> >>>>> >>>>> On Thu, May 23, 2024 at 4:16 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>> wrote: >>>>> >>>>>> We have tested this PR using the Linphone app. >>>>>> So your case will be resolved using this PR. >>>>>> Need to enable HAproxy protocol headers. >>>>>> >>>>>> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>>> wrote: >>>>>> >>>>>>> Please try Kamailio PR >>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>> >>>>>>> We have developed this PR for use case you have described. >>>>>>> We have tested Route and Record-Route headers not Via. >>>>>>> So will provide some review for this PR then will be fine. >>>>>>> >>>>>>> On Wed, May 22, 2024 at 2:22 PM David Villasmil < >>>>>>> david.villasmil.work(a)gmail.com> wrote: >>>>>>> >>>>>>>> Hello Sergey, >>>>>>>> >>>>>>>> Thanks for the suggestion. Not sure if his is what i'm looking for, >>>>>>>> allow me to explain further: >>>>>>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This >>>>>>>> Load balancer is a TLS listener on the outside and a TCP connection to the >>>>>>>> proxy inside. >>>>>>>> So when sending an INVITE to the connected client, the via has a >>>>>>>> TCP protocol like >>>>>>>> >>>>>>>> Via: SIP/2.0/TCP >>>>>>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0 >>>>>>>> >>>>>>>> and the client (linphone) disconnects and tries to contact the >>>>>>>> proxy on that address on a TCP socket, which doesn't exist. I tried many >>>>>>>> solutions none of which actually work... last one setting $du =$du + >>>>>>>> ";transport=tls" and forcing the socket to the TCP socket to the load >>>>>>>> balancer, but of course i'm getting warnings about this. >>>>>>>> >>>>>>>> is this something that PR (not merged) would be addressing, i >>>>>>>> didn't see that. >>>>>>>> If not, is there a way of doing this without any trickery? >>>>>>>> >>>>>>>> Thanks! >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>> phone: +34669448337 >>>>>>>> >>>>>>>> >>>>>>>> On Wed, May 22, 2024 at 12:16 PM Sergey Safarov < >>>>>>>> s.safarov(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Probable you need this PR >>>>>>>>> https://github.com/kamailio/kamailio/pull/3810 >>>>>>>>> >>>>>>>>> Or you can try >>>>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>>>> In this PR we faced the same issue and solved this. >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users < >>>>>>>>> sr-users(a)lists.kamailio.org> wrote: >>>>>>>>> >>>>>>>>>> Hello Anthony, did you solve this problem? I'm facing the same >>>>>>>>>> problem >>>>>>>>>> >>>>>>>>>> Thanks! >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>>>> phone: +34669448337 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba < >>>>>>>>>> ascanio.alba7(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I have kamailio behind a TLS termination proxy so the sockets >>>>>>>>>>> are correctly deduced to be TCP. However the clients only talk TLS to the >>>>>>>>>>> proxy and are confused when the top Via header added by Kamailio is TCP. Is >>>>>>>>>>> there a way for Kamailio to forcibly pretend its protocol is TLS? Like >>>>>>>>>>> advertised_address but "advertised_protocol" instead. >>>>>>>>>>> >>>>>>>>>>> (With pjsip testing: it has a flag use_tls which ignores TCP >>>>>>>>>>> from Kamailio and continues to use the persistent TLS transport to proxy. >>>>>>>>>>> Linphone fails because it tries to honor TCP in Via and is unable to >>>>>>>>>>> establish TCP transport). >>>>>>>>>>> >>>>>>>>>>> BTW I am using t_relay_to_tcp so Kamailio will return traffic to >>>>>>>>>>> the proxy as TCP even though the contact addresses specify transport=TLS. >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>> __________________________________________________________ >>>>>>>>>> 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

Inbound Identity Verification Stir/Shaken
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I have a quick question regarding verification on incoming calls regarding the stir/shaken policy. I'm using the secsipid library to check the identity header, but the problem I'm running into is that there is no identity header in the invite from the incoming call (It's showing up as null). Have most carriers enabled passing along an identity header, or is this something that is still in the works for some of them. Would I need to request and get the certificate from the terminating carrier? I've tried using the auth_identity module to see if i need to use the verification functions to get/request the certificate, but I have had no luck so far. Would anyone be able to provide direction on how to request/get/pass along the identity header from an incoming call with stir/shaken. (I am using kamailio v5.6 and debian 12) Thanks in advance, Temi

4 6

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

4 10

libstirshaken installation issues
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I am attempting to install the stirshaken module for kamailio and I ran into a couple issues installing the open source c library for libstirshaken. When I use the make command, I'm getting this error: src/stir_shaken.c: In function 'stir_shaken_is_key_trusted': src/stir_shaken.c:726:9: error: 'EVP_PKEY_cmp' is deprecated: Since OpenSSL 3.0 [-Werror=deprecated-declarations] 726 | if (!EVP_PKEY_cmp(pkey, candidate_pkey)) { | ^~ In file included from /usr/include/openssl/x509.h:29, from /usr/include/openssl/ssl.h:31, from /usr/include/libks2/libks/ks_ssl.h:25, from /usr/include/libks2/libks/ks.h:80, from include/stir_shaken.h:15, from src/stir_shaken.c:1: /usr/include/openssl/evp.h:1418:5: note: declared here 1418 | int EVP_PKEY_cmp(const EVP_PKEY *a, const EVP_PKEY *b); | ^~~~~~~~~~~~ At top level: cc1: note: unrecognized command-line option '-Wno-gnu-zero-variadic-macro-arguments' may have been intended to silence earlier diagnostics cc1: all warnings being treated as errors make: *** [Makefile:1337: src/stir_shaken.lo] Error 1 I tried going into the stir_shaken file and changing EVP_PKEY_cmp to EVP_PKEY_eq, but it just brings up more deprecation errors. I also tried rolling back the openssl version on my machine to openssl version 1.1 and changing EVP_PKEY_eq back to EVP_PKEY_cmp, but the same error persists. Is there any way to resolve this error, or is there another library that can be used to get the stirshaken module for kamailio? Thanks in advance, Temi

6 7

Is udp_mtu and udp_mtu_try_proto broken in kamailio v5.8
by Supreeth Herle 30 May '24

30 May '24

Hello, I have a dockerized IMS setup using kamailio from git commit id 4fb8accc6747ad56fec3dc84d70cb2b8bbd7316e (tag 5.8.x), where I am seeing an issue faced by one of the users of the dockerized IMS. The issue is that the SIP request (in UDP) even though it exceed the MTU of 1300 (udp_mtu=1300 and udp_mtu_try_proto=TCP) the SIP request from P-CSCF is not sent out as a TCP. The pcap and P-CSCF logs in debug can be found in the github issue - https://github.com/herlesupreeth/docker_open5gs/issues/316#issuecomment-213… IP of the containers are follows: P-CSCF - 172.22.0.21 S-CSCF - 172.22.0.20 SIP Client - 192.168.101.x P-CSCF is configured with udp_mtu=1300 and udp_mtu_try_proto=TCP In the pcap you can notice that the SIP NOTIFY originating from S-CSCF is forwarded to P-CSCF (packet no. 698) and packet length is 2716. At this point P-CSCF should have tried to convey the SIP packet to SIP client via TCP. But I dont see any TCP connection establishment either and eventually request times out (packet 1147). [image: image.png] Any help or insights would be highly appreciated. Thanks and Regards, Supreeth

1 0

Re: Failover with dialog dmq while dialog state 2
by Björn Klasen 27 May '24

27 May '24

Hi Henning, thank you for your reply. I did a lot testings during the past days and I think I found a solution that is working, although it's very dirty... I found out it is possible to send a manual KDMQ-Message to all Proxy-Nodes including the one that it sending die KDMQ-Message. So I create a dialog state 4 message on ACK and indeed the Dialog on all Peers are Updated correctly to state 4. I also tried this for state 3 on 200 OK, but this does not work. I haven't had a look into the code yet but my experiments shows, that there is never is state 3 update via DMQ, so I think it's not implemented. Interestingly the TM-Timer still times out after 300 seconds but in normal circumstances this doesn't bother, because either the failed Kamailio will not trigger this, because the Software or the Server crashed, or it tries to send a CANCEL that is denied because we are in dialog state 4. The only thing is your really have to be careful about your hash tables. You have to delete them at the correct point. I really need to test a lot at this point, but till now, billing seems to be OK even on failover scenarios. This is why I really like Kamailio :) BR, Björn Am 09.05.24 um 20:57 schrieb Henning Westerholt: Hello, thanks for the detailed e-mail. As also indicated in the module documentation, the dialog module DMQ replication will not replicate everything, its main use-case is for profile data sharing. https://kamailio.org/docs/modules/5.8.x/modules/dialog.html#dialog.p.enable… In the past months there have been some other discussions on the users lists about similar scenarios (I think related to billing/accounting) and dialog with DMQ, which might be interesting for you in this regard. If you find issues where the DMQ synchronisation is lacking some functionality in the dialog module, you can create a feature request in our issue tracker. There is of course no guarantee that this limitation is also timely addressed. Regarding the INVITE and CANCEL scenario, this is usually not related to dialog but to the tm module. As you also mentioned, there is no replication of transaction state in tm. Cheers, Henning -- Björn Klasen, Senior Specialist (VoIP) TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de<mailto:bklasen@tng.de> https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 3

I'm looking for an get_rr_param() usage example
by Benoit Panizzon 27 May '24

27 May '24

Hi Gang To process spiraling calls without the dialog module. I'm trying to use a rr param to identify in which iteration I am. So when I have a call spiraling through the same instance 3 times, every time I process an invite I would do something like: add_rr_param(";sp-count=$avp(sp-count)"); But to initialize that counter, I need to be able to read what is in there beforehand. if (check_route_param("sp-count=") { get_route_param(msg, "sp-count", $avp(sp-count)) $avp(sp-count) = $avp(sp-count) + 1; add_rr_param(";sp-count=$avp(sp-count)"); } else { add_rr_param(";sp-count=1"); } But what do I pass as msg parameter? Is it the PV $msg(hdrs) ? PS: I need this to append this to the call ID for rtpengine so I hopefully can match new transactions no matter from which side they are initiated to the correct rtp stream. 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

Is there any funtion to route based on source phone number
by SAMUEL MOYA TINOCO 27 May '24

27 May '24

Hi everyone, I have a kamailio configured to send invites from different phone numbers registered in kamailio to different endpoints. The configuration that I have right now is a big if chain with the phone numbers that then sends the invite using dispatcher module. Example: If (phone number in invite = phone number X) { dispatcher sends to endpoint X} If (phone number in invite = phone number Y) { dispatcher sends to endpoint Y} If (phone number in invite = phone number Z) { dispatcher sends to endpoint Z} Im not sure if this is the correct way of doing this or if there is any module that can do this allowing me to configure it using BBDD instead of kamailio configuration. I started doing it like this on our test environment while I was learning how to use kamailio and I'd like to know if this is the right way to do it or if I should do it with other methods Thank you in advance

7 7

Fwd: rtpengine timestamp jumps
by Jeff Brower 26 May '24

26 May '24

Rtpengine support- I am re-posting again, as I noticed on the SR-USERs web page no HTML or inline images are showing (https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…). The Wireshark screencap that demonstrates the issue is uploaded at: https://www.signalogic.com/images/rtpengine_wireshark_capture_timestamp_jum… all other info remains the same. Thanks, Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Fri, 17 May 2024 17:10:10 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: Fwd: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: [screencap link at https://www.signalogic.com/images/wireshark_capture_timestamp_jump.png] In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message ----- ----- End forwarded message -----

2 4

core children setting
by Alexis Fidalgo 24 May '24

24 May '24

Hello all, im not clear on the children setting, currently we’re listening in a single UDP port, i do understand that children are how many forks per listener ip:port, but i have this question, is this value tied on how many requests are received? In our case every requests takes an average of 80ms to be processed (there are a few external queries executed per request), so, example which value for children should i use if i have 200req/s, 300, 400? Right now im using 16 but i can see some cpu peaks in high traffic hours. thanks

2 2

"Contact does not exist in PCSCF usrloc" error in log
by Shantanu K Roy 23 May '24

23 May '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu

1 0

VoLTE/IMS third party registration issues
by ruben.iniguez＠vodafone.com 23 May '24

23 May '24

Checking Kamailio in a VoLTE environment we are finding issues to make it work. First issues arise when getting AS to extract the required info from third party registration, primarily IMPUs. Third party registration is sent to AS with IMPI in To header, preventing the AS from extracting the IMPU from the most obvious place, the To header, according to 3GPP 24.229: "Notification of Application Servers about registration status [...] the To header field, which shall contain a non-barred public user identity belonging to the service profile of the processed Filter Criteria. It may be either a public user identity as contained in the REGISTER request received from the UE or one of the implicitly registered public user identities in the service profile, as configured by the operator;" On the other hand, the third party registration can only embed in its body either the original request or the original response but not both (notice mutually exclusive "else if" statements in third_party_reg.c). Both are needed to extract Feature-Caps (from original request) and IMPUs (for instance from P-Associated-URI in original response). Therefore to extract that info the AS should subscribe to the reg info in S-CSCF, but such SUBSCRIBE is rejected with 403. From Registrar IMS SCSCF documenation it looks like this is not supported: "This function checks to see that a SUBSCRIBE request is authorised to subscribe to the particular identity. Only 3 entities can subscribe: The user agent to its own state The P-CSCF specified in the path header for that user Application Server (AS) not yet implemented" Therefore question to the community is how these limitations are overcome in existing VoLTE deployments with Kamailio? Any advice?

1 0

Kamailio IPSec module
by H Yavari 23 May '24

23 May '24

Hi all, I was reviewing the `ims_ipsec_pcscf` code and noticed that this module creates a pool of sockets using different ports (ipsec_max_connections). I'm unclear on the necessity of this approach. Can't we simply create one server listener and one client listener to handle all UE connections? If this is feasible, is there still a need to create the pool at startup? I'm curious if this is due to an architectural limitation or if the IPSec module can be modified to replace the current implementation with a more efficient one. Thank you for your insights. Best regards,Hossein

3 5

S-CSCF SiFC support
by ruben.iniguez＠vodafone.com 22 May '24

22 May '24

Hello In the context of an IMS deployment, does anybody know if Shared iFC mechanism is supported in Kamailio? I don't see configuration options for it, and just wanted to confirm if only regular iFCs (residing on HSS) are supported. Thank you

1 0

Assistance Required for VoNR Setup with Version 5.8.1
by venkateshcoff＠gmail.com 22 May '24

22 May '24

Hi, I am trying to set up VoNR with version 5.8.1, but I cannot find the configuration information and documentation anywhere. Has anyone tested a VoNR call? Thanks.

1 0

usrloc: keepalives not being sent to UDP endpoints for a very long time
by luke＠primevox.net 21 May '24

21 May '24

We've noticed an issue with UDP endpoints and the usrloc module not sending keepalive packets to them for a long time. Eventually, usrloc will begin doing it, but sometimes it's anywhere between 10 minutes and multiple hours before the endpoint starts receiving an OPTIONS packet. TCP and TLS endpoints do not have this issue - they being receiving OPTIONS after registration + ka_timeout elapsed. Has anyone else seen this? It can be very easily seen with a pcap on a device registering via UDP.

1 0

A question about Db_flatstore Module in Kamailio
by فرزانه سلطان زاده 20 May '24

20 May '24

Hi, Hope to find you well, I have a question about the 'db_flatstore' module in Kamailio. Here's my script for log rotation: for f in acc_cdrs_*.log; do mv -- "$f" "${f%.log}.temp" done docker exec -it pcscf kamcmd flatstore.k_rotate rm *.temp I want to know if the kamcmd command is blocking and the files can be removed after that or if I should wait for it to be done before removing the renamed files. Regards, Farzaneh

3 2

Kamailio multidomain IMS
by Valentin Christoph 20 May '24

20 May '24

Dear IMS enthusiasts :-) Has anyone experience with Kamailio IMS (CSCF) setups, where users with different domain names can be registered at one and the same S-CSCF? Is this feasible at all? I mean one IMS for users <user>@ims1.example.com, <user>@ims2.example.com, <user>@ims3.example.com,........ in one and the same S-CSCF, I-CSCF, P-CSCF. Thanks, Christoph

2 1

q_malloc.c:520 WARNINGs
by Joel Serrano 20 May '24

20 May '24

Hi, Does anyone know what this WARNING means or how to fix it? 2024-03-09T00:39:15.406674-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406823-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406973-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407089-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407178-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407278-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407375-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) It happens after running dp_translate(). These are my settings: debug=L_DBG debug=L_NOTICE memdbg=L_INFO memlog=L_INFO corelog=L_ERR mem_summary=12 log_stderror=no log_facility=LOG_LOCAL0 log_name="csbc" log_prefix="{$mt $hdr(CSeq) $ci} " mem_join=1 Any ideas? Thanks, Joel.

2 5

tm.t_uac_wait ACK does not have Max-Forwards header
by Juha Heinanen 20 May '24

20 May '24

Kamailio sends INVITE by tm.t_uac_wait RPC command: INVITE sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com> From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Content-Length: 131 User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Max-Forwards: 10 Contact: <sip:click2dial@192.168.187.215:5060> Content-Type: application/sdp .. sdp omitted UAS accepts the INVITE: SIP/2.0 200 Answering Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Server: baresip v3.12.0 (x86_64/Linux) Contact: <sip:test-0x55e7f1f78e70@192.168.187.215:5022> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,UPDATE,REFER Supported: gruu,replaces,norefersub Content-Type: application/sdp Content-Length: 239 .. sdp omitted After receiving the response, Kamailio automatically sends ACK: ACK sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 Call-ID: 6649e78355ac2(a)test.tutpro.com To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e CSeq: 1 ACK User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Content-Length: 0 The problem is that Max-Forwards header is missing from the ACK. Should I have something in config file to add it or is this a bug? -- Juha

1 3

Segfault on startup when using TLS in kamailio 5.7.4
by stefanr＠wave.com 17 May '24

17 May '24

Hi there, We are encountering consistent segfaults after rebooting our Kamailio instance with incoming traffic, specifically when using Kamailio 5.7.4. We think this issue did not occur with version 5.7.2, so it seems to have been introduced in either 5.7.3 or 5.7.4. Due to team bandwidth constraints and the potential impact on production traffic, we don't want to spend time on trying to reproduce the issue. So we have decided to downgrade to 5.6.4, which we confirmed to be stable. (Probably 5.7.2 would be too - but we didn't try). Unfortunately, our logging was only set to WARNING level, and we did not capture a core dump, so we cannot provide additional details beyond the following logs: This was with tcp_reuse_ports=yes: 2024-05-17T15:42:55.582475541Z Listening on 2024-05-17T15:42:55.582512370Z [redacted] 2024-05-17T15:42:55.582538161Z tls: 10.X.X.X:5061 advertise Y.Y.Y:5061 2024-05-17T15:42:55.582543750Z Aliases: 2024-05-17T15:42:55.582549081Z tls: [redacted]:5061 2024-05-17T15:42:55.582574890Z 2024-05-17T15:42:55.587876630Z 0(1) WARNING: tls [tls_init.c:978]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 18874368 and 9437184 bytes 2024-05-17T15:42:55.703927049Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 23 2024-05-17T15:42:55.703972029Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 15 exited by a signal 11 2024-05-17T15:42:55.703978409Z 0(1) ALERT: <core> [main.c:795]: handle_sigs(): core was generated 2024-05-17T15:42:55.705049839Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 17 2024-05-17T15:42:55.705074209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 21 2024-05-17T15:42:55.705081209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 22 2024-05-17T15:42:55.705085879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 20 2024-05-17T15:42:55.705090319Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 18 2024-05-17T15:42:55.705094649Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 19 2024-05-17T15:42:55.705098879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 16 2024-05-17T15:42:55.705207399Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 15 2024-05-17T15:42:55.705459439Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 27 Without tcp_reuse_ports=yes, the segfault was always preceded by the following line if any existing TLS connections were stuck in TIME_WAIT: 2024-05-16T19:18:51.654447639Z 9(14) WARNING: {1 1 INVITE XXX(a)0.0.0.0} <core> [core/tcp_main.c:1301]: find_listening_sock_info(): binding to source address 10.X.X>X:5061 failed: Address already in use [98] 2024-05-16T19:18:51.746994728Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 14 exited by a signal 11 When the server wasn't handling any traffic, the issue didn't occur even in 5.7.4. Does anyone have any insights or suggestions on how to address this issue? Kind regards Stefan

2 2

Assistance Needed to Resolve Duplicate Contact Headers in 200 OK Response
by satyaprakash ch 17 May '24

17 May '24

Hi We are receiving 2 to 3 contact headers in the 200 OK response of the registration request. It appears that the `user_location` is updating multiple contact headers for the same extension due to network disconnections. Please suggest a solution to avoid duplicates in the `user_location` module, or how to ensure that only the latest single contact header is included in the 200 OK response. Your suggestion would be appreciated. Thanks and regards, Satyaprakash.

2 1

Fwd: rtpengine timestamp jumps
by Jeff Brower 17 May '24

17 May '24

Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message -----

1 0

$dlg_var() not implemented in ims_dialog module
by Valentin Christoph 17 May '24

17 May '24

Dear all, I would probably need to store some dialog scoped information in an IMS environment, but the ims_dialog module does not implement the $dlg_var() pseudo variable. Do you know other, maybe better, possibilities to store dialog scoped information in an IMS (CSCF) environment? Thank, Christoph

3 2

making HTTP Async request in stateless configuration
by Sergio Charrua 17 May '24

17 May '24

Hi all! Just wondering what would be you opinion on the following. We are integrating Kamailio as a stateless redirect server, using SIP 300 Multiple Choices response. The script requires that prior to the redirection, a HTTP request must be made. We thought about using HTTP Async Client to make the requests (POST of JSON content), which is our prefered way of sending HTTP requests, and depending on the response from the REST API, reply back to UAC a SIP 300 Multiple Choices message and terminate the call flow on Kamailio. From what I know, HTTP Async Client requires that the TM module be used, which (it is my understanding) will imply that the whole Kamailio script to be stateful, which I think is useless as the SIP flow will be really simple: INVITE -----> <------- 100 Trying <------- 300 Multiple Choices ACK ------> There is a catch, though, the CPS is very high, above 1000 CPS.... Could anyone share their thoughts on this? Should we go stateful and use the HTTP Async module? Should we go stateless and use the HTTP client module? What would be the best approach to implementing this? Thanks guys! *Sérgio Charrua*

1 0

WebRTC-UDP voicemail and call transfer
by Social Boh 16 May '24

16 May '24

Hello list, I'm trying implementing a solution like: https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg The 2 problems I have a this moment are: Call from UDP phone to WebRTC phone, the WebRTC phone reject the call; the call go to voicemail but in the SDP still have WebRTC SDP During a SIP UDP - WebRTC Call tranfer the call using phone button... the transfer don't work. Any hint Regards -- --- I'm SoCIaL, MayBe

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 14 May '24

14 May '24

How about this new function that sets the algorithm? -- Juha iff --git a/src/modules/auth/auth_mod.c b/src/modules/auth/auth_mod.c index 534ad9e20f..4e3a584d22 100644 --- a/src/modules/auth/auth_mod.c +++ b/src/modules/auth/auth_mod.c @@ -74,6 +74,10 @@ int w_consume_credentials(struct sip_msg *msg, char *s1, char *s2); * Check for credentials with given realm */ int w_has_credentials(struct sip_msg *msg, char *s1, char *s2); +/* + * Set authentication algorithm + */ +int w_auth_algorithm(struct sip_msg *msg, char *alg, char* s2); static int pv_proxy_authenticate( struct sip_msg *msg, char *realm, char *passwd, char *flags); @@ -170,6 +174,8 @@ static cmd_export_t cmds[] = { REQUEST_ROUTE}, {"pv_auth_check", (cmd_function)w_pv_auth_check, 4, fixup_pv_auth_check, 0, REQUEST_ROUTE}, + {"auth_algorithm", w_auth_algorithm, 1, fixup_spve_null, 0, + REQUEST_ROUTE}, {"bind_auth_s", (cmd_function)bind_auth_s, 0, 0, 0}, {0, 0, 0, 0, 0, 0} @@ -475,6 +481,35 @@ int w_has_credentials(sip_msg_t *msg, char *realm, char *s2) return -1; } return ki_has_credentials(msg, &srealm); + +} +/** + * + */ +int w_auth_algorithm(sip_msg_t *msg, char* alg, char* s2) +{ + if (fixup_get_svalue(msg, (gparam_t*)alg, &auth_algorithm) < 0) { + LM_ERR("failed to get algorithm value\n"); + return -1; + } + + if (strcmp(auth_algorithm.s, "MD5") == 0) { + hash_hex_len = HASHHEXLEN; + calc_HA1 = calc_HA1_md5; + calc_response = calc_response_md5; + } + else if (strcmp(auth_algorithm.s, "SHA-256") == 0) { + hash_hex_len = HASHHEXLEN_SHA256; + calc_HA1 = calc_HA1_sha256; + calc_response = calc_response_sha256; + } + else { + LM_ERR("Invalid algorithm provided." + " Possible values are \"\", \"MD5\" or \"SHA-256\"\n"); + return -1; + } + + return 1; } #ifdef USE_NC

3 6

302 Message Handling
by tfayomi＠untangledtechnology.com 14 May '24

14 May '24

Hi All, I'm having trouble understanding/finding resources on how to handle a 302 message I receive. I'm working on making an inbound sbc with kamailio with clearip integration. I'm successfully sending to kamailio, but am having trouble retrieving the response from clearip in kamailio. For some background, the intended call flow should be this: first it should send from caller to kamailio (caller > kamailio), then over tcp kamailio sends to clearip (kamailio > clearip) [This is what is working as expected] Then I should be able to signal to kamailio to continue the call to the intended destination. When clearip sends a code back (in this case 302), I want to be able to see that code and send the caller to the correct destination. The problem I keep running into is that I can't seem to see the 302 code in a reply, branch or failure root or checking the status. I'm just lost on what I am misunderstanding and would appreciate if anyone could point me in the right direction on how to approach retrieving the 302 message. I'm using kamailio version 5.6.3 and debian 12.1 Thanks in advance, Temi

2 1

Prefix match in htable
by Antonio Reale 13 May '24

13 May '24

Hello, I'm testing Kamailio 5.x as a redirecting server based on calling number. The calling numbers are stored on a DB and loaded in htable at runtime. When the INVITE or REGISTER arrive Kamailio checks if there is a match on $fU in htable (KEY_NAME) and if there is one, it replies with 302 adding the Contact HF with $fU@$sht(cli=>$fU)(with the domain part found in KEY_VALUE): append_to_reply("<Contact: sip:$fU@$sht(cli=>$fU)>\r\n"); If I use the full number as KEY_NAME it works as expected, but how can I match on a prefix instead of the whole $fU? For example, If I receive a call form 0655123456 I want to match it also if I have the prefix 0655 in htable. Do you have some advice to achieve this goal with htable or in a similar way? Thank you. Best regards. -- Antonio Reale

1 0

Kamailio and RTPengine and no voice on calls
by palany 10 May '24

10 May '24

I have an AWS ec2 instance with Kamailio and rtpengine configured. I am having a challenge of no voice on calls. I have sip client on my mobile phone and soft phone on my laptop and the two are on different networks behind nat. Signaling is working fine but there is no media and when I check on on the SDP the call logs are showing private ips instead of public ips : o=- 3 2 IN IP4 192.168.1.33 and c=IN IP4 172.31.36.50. I understand this is a NAT issue but I am failing to see where is the challenge. My rtpengine seems to be working fine . Below is my Kamailio.cfg file link : https://docs.google.com/document/d/1MKZJGlQjxYbdtw6QcCm9_EmvVPOkMs-YXjlq1ZM2 mmY/edit?usp=drive_link PCAP file for the call: https://docs.google.com/document/d/12gofLYhN0-lYIZORngIeuaYxLudLjiNdZgY504Ei XFs/edit?usp=drive_link rtpengine.conf https://docs.google.com/document/d/1QPxey49UzuN3Wd8jp8jc0mFhh31rr1alIO-gl4gP JEI/edit?usp=drive_link Please help.

2 3

base64 decoding issue
by Sergio Charrua 10 May '24

10 May '24

Hi all! I have been dealing with STIR/SHAKEN for a few weeks now, and while doing some tests I have found an issue which I can't find the reason for. The script has the following route logic: route[HANDLE_STIRSHAKEN] { xlog("L_INFO", "HANDLE_STIRSHAKEN - STIR/SHAKEN Logic"); if ($hdrc(Identity) == 0 ){ xlog("L_INFO", "HANDLE_STIRSHAKEN - No Identity Header -> Skipping this logic"); return; } #Verify Call Identity xlog("L_INFO", "HANDLE_STIRSHAKEN - Identity Header: $hdr(Identity)"); $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 1: $var(header_1)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 2: $var(header_2)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsing!"); $var(hdr_1_json_payload) = $(var(header_1){s.decode.base64}); # Header 1 xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsed!"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Header 1 payload = $var(hdr_1_json_payload) "); $var(hdr_2_json_payload) = $(var(header_2){s.decode.base64}); # Header 2 jansson_get_field($var(hdr_2_json_payload), "attest", "$var(attest_value)"); jansson_get_field($var(hdr_2_json_payload), "ppt", "$var(ppt_value)"); jansson_get_field($var(hdr_2_json_payload), "typ", "$var(typ_value)"); jansson_get_field($var(hdr_2_json_payload), "x5u", "$var(x5u_value)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - $var(attest_value) $var(ppt_value) $var(typ_value) $var(x5u_value) "); } All it does is parse the Identity header received on the initial INVITE, decodes it (it is encoded in base64) and writes some JSON values on log. Nothing serious.... As the Identity header is divided into multiple parts, separated by a ".", I get the first 2 parts in 2 private variables, as per lines: $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); Logs show correctly that both variables are populated: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 1: eyJhbGciOiAiRVMyNTYiLCJwcHQiOiAic2hha2VuIiwidHlwIjogInBhc3Nwb3J0IiwieDV1IjogImh0dHBzOi8vY2VydGlmaWNh dGVzLmNsZWFyaXAuY29tL2IxNWQ3Y2M5LTBmMjYtNDZjMi04M2VhLWEzZTYzYTgyZWMzYS83Y2M0ZGI2OTVkMTNlZGFkYTRkMWY5ODYxYjliODBmZS5jcnQifQ May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 230 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 2: eyJhdHRlc3QiOiAiQSIsImRlc3QiOiB7InRuIjogWyIxNDA0NTI2NjA2MCJdfSwiaWF0IjogMTU0ODg1OTk4Miwib3JpZyI6IHsi dG4iOiAiMTgwMDEyMzQ1NjcifSwib3JpZ2lkIjogIjNhNDdjYTIzLWQ3YWItNDQ2Yi04MjFkLTMzZDVkZWVkYmVkNCJ9 However, when decoding both values: xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); only $var(header_2) gets decoded: May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 38 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 1:* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 182 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 2: {"attest": "A","dest": {"tn": ["14045266060"]},"iat": 1548859982,"orig": {"tn": "18001234567"},"origid": "3a47ca23-d7ab-446b-821d-33d5deedbed4"}* As you can veri And to ease debugging, I have isolated the line that generates the issue: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsing! May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:351]: lval_pvar_assign(): non existing right pvar* *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:411]: lval_assign(): assignment failed at pos: (21,32-21,64)* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 27 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsed! Even though the $var(header_1) contains a base64 encoded string, the decoder seems to fail. If I copy the value contained in $var(header_1) to an online decoder like https://www.base64decode.org/ the string gets decoded correctly. I have even modified the line : $var(header_1)= $(hdr(Identity){s.select,0,.}); to get the reverse index from the Identity string by doing $var(header_1)= $(hdr(Identity){s.select,-6,.}); It gets the same encoded string, same value (as expected), but the result is the same: Kamailio's s.decode.base64 fails with the same error. i'm not sure what assignment failed at pos: (21,32-21,64) could mean.... I don't think my code is wrong, as the same function is used on $var(header_2) value and it works, but maybe someone can correct me. Greatly appreciate your help. Cheers, *Sérgio Charrua*

1 1

Failover with dialog dmq while dialog state 2
by Björn Klasen 09 May '24

09 May '24

Hello @all as we want to enhance our VoIP-Plattform with DMQ functionalities for replication of Userlocation, Dialog and Hash Tables I made a lot of tests durings the last week. For testing I use two Debian 12 Server with Kamailio 5.7.4 (latest git) Both server have their own floating IP so in case one server fails the floating changes to the working server. This is done via keepalived On both servers Kamailio is active and use ip_free_bind=1 so both Kamailio instances can listen to all necessary IPs. So we have following network configuration: Server1: static IP: 192.168.253.201 floating IP: 192.168.253.200 Server2: static IP: 192.168.253.202 floating IP: 192.168.253.210 Now I test some calls while I trigger a failover, so Server1 takes over 192.168.253.10 Message Test Result CustomerA -> Proxy -> CustomerB Failover while Dialogstate 4 BYE is relayed correctly. Dialog is changing to state 5 CDR needs to be written manually -> no problem at all CustomerA -> Proxy -> CustomerB Failover while Dialogstate 2 200 OK is relayed correctly although no transaction exist an therefore no t_reply route is triggered. This is not a big deal because i can use default onreply_route so all necessary variables can be used via hash table. BUT Big Problem: Dialog state stays 2, so a following ACK is a bogus event and the dialog state is never changed to 4. Even worse: after 300 seconds the dialog is cleaned away by the timer, so I have no chance to do any kind of accounting. Test 2 is a real mess, because for me there is only one solution. I have to check after a failover on incoming replies to an INVITE that was handled on Server2 if these current dialog state is < 4 and cancel them. But I can't see how to do it? Are there other ways than using t_cancel? Or is it even some kind of a bug that a 200 OK does not trigger state 3 on dialog (although I have in mind that it has to be something to do with missing transaction data, that are not replicated because of missing replication capabilities of tm module)? If none of the above is possible I really asking myself of a real use case of dialog dmq replication. -- Björn Klasen, Teamleitung NGN VoIP-Backbone TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 1

How to set "set_dialog_profile" value for Leg-a and leg-b.
by satyaprakash ch 09 May '24

09 May '24

Hi, We are using kamailio and freeswitch, So when we make a call from A to B the call is going via "kamailio -> freeswitch" and "freeswitch -> kamailio". Please suggest me how i can set "set_dialog_profile" for both legs with unique value in kamailio config file. Thanks.

1 0

Is there any way to modify the Refer-To header username?
by Shah Hussain Khattak 09 May '24

09 May '24

Hello Experts, I wanted to check, is there any way to modify the username part of the Refer-To header in Kamailio? For example, I am getting the following header: REFER-TO: <sip:sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> the username part is missing. Can I add a username inside it? For example, change it to REFER-TO: <sip:123@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> REFER-TO: <sip:test-kam@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> I could not find any solution in the mailing list, the only way was to remove the header and then re-add it. Thank you! Regards, Shah Hussain

2 2

rtpengine timestamp jumps
by Jeff Brower 09 May '24

09 May '24

Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff

1 0

Kamailio v5.7.5 Released
by Daniel-Constantin Mierla 08 May '24

08 May '24

Hello, Kamailio SIP Server v5.7.5 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.5. For more details about version 5.7.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/05/kamailio-v5-7-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.1 was released on April 3, 2024. 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 v5.7.5
by Daniel-Constantin Mierla 08 May '24

08 May '24

Hello, I am considering to release Kamailio v5.7.5 (out of branch 5.7) on Wednesday, May 8, 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

sql_query fails from event_route
by Juha Heinanen 07 May '24

07 May '24

I noticed that sql_query does not (anymore) work when called from event route: event_route [tcp:closed] { # Handle TCP connection close # Delete possible leftover registration if ($conid) sql_query("sip_proxy_cache", "DELETE FROM location WHERE server_id=0 AND connection_id=$conid", "result"); } The query produces errors: 2024-05-07T11:30:14.785915+03:00 siika /usr/bin/sip-proxy[2848078]: CRITICAL: sqlops [sqlops.c:213]: sql_check_connection(): no database handle with reconnect disabled 2024-05-07T11:30:14.785978+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:232]: sql_query(): invalid connection to database 2024-05-07T11:30:14.786013+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:237]: sql_query(): with query [DELETE FROM location WHERE server_id=0 AND connection_id=7] I added the last debug message in order to find out where the error comes from. The same query works fine when executed from regular route. Any ideas why the query fails from event_route? -- Juha

2 3

How to use environment variable
by Pavan Kumar 07 May '24

07 May '24

Hi everyone, I am trying to assign environment variable as follows listen=udp:0.0.0.0:5060 advertise $env(MY_IP):5060 Looks like using the environment variable as above is an invalid configuration. Is there a way to use IP from env var to advertise. Even better, is there a way to use result in a stun query as an advertised address? Thank you, Pavan Kumar

2 3

Retransmission behaviour after ACK
by Sergio Charrua 06 May '24

06 May '24

Hi all! I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing. The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules. Request route is simple: request_route { route(HANDLE_OPTIONS); route(HANDLE_RETRANS); route(HANDLE_CANCEL); route(REQINIT); route(HANDLE_INVITE); } where HANDLE_RETRANS is: route[HANDLE_RETRANS]{ # handle retransmissions xlog("L_INFO","HANDLE_RETRANS - message type is $mt for method $rm"); if (!is_method("ACK")) { if(t_precheck_trans()) { xlog("L_INFO","HANDLE_RETRANS - Precheck Trans"); t_check_trans(); xlog("L_INFO", "HANDLE_RETRANS - Exiting after Retransmission check - method $rm"); exit; } xlog("L_INFO","HANDLE_RETRANS - Check Trans"); t_check_trans(); } } and HANDLE INVITE is: route[HANDLE_INVITE]{ if (is_method("INVITE")) { xlog("L_INFO","MAIN - calling TO_CARRIER"); setflag(FLT_ACC); route(TO_CARRIER); exit; } } The HTTP requests call a REST API returning a JSON object, that is iterated and produces a CONTACT header containing multiple choices using the append_branch() command. The RELAY route is : route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","REPLY_302 - send reply"); send_reply("300", "Multiple Choices"); xlog("L_INFO","REPLY_302 - exit"); exit; } Which returns indeed a Contact header with multiple contact choices. When using Asterisk to originate a call to Kamailio, the SIP call flow is as follows and terminates (SNGrep and TCPDump do not detect anything else): SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> However when using SIPp I get some retransmissions after the final ACK. SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── SIPp scenario is : <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC">   <send retrans="500" start_rtd="true" rrs="true"> <![CDATA[ INVITE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]> Contact: sip:sipp@[local_ip]:[local_port];transport=[transport]> Call-ID: [call_id] CSeq: 102 INVITE User-Agent: SIPp Supported: replaces, timer Content-Type: application/sdp Subject: Performance Test Session-Expires: 3600;refresher=uas Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIPp c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" rss="true"> </recv> <recv response="300" rtd="true" rss="true"> <action> <ereg regexp="\<([a-zA-Z0-9@.=;]+)[^>]+" search_in="hdr" header="Contact: " assign_to="1"/> <log message="INFO: 300 Contact Header: [$1]" /> <exec command="echo [$1] >> ./300_ContactHeader.txt" /> </action> <action> <ereg regexp="sip:.*@([0-9A-Za-z\.]+):([0-9]+);transport=([A-Z]+)" search_in="hdr" header="Contact:" check_it="true" assign_to="dummy,host,port,transport" /> </action> </recv>   <send rrs="true"> <![CDATA[ ACK sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]>[peer_tag_param] Contact: <sip:sipp@[local_ip]:[local_port]> Call-ID: [call_id] CSeq: 102 ACK User-Agent: SIPp Content-Length: 0 ]]> </send>   </scenario> The question is : is there anything wrong with my kamailio script? or with SIPp scenario? why does kamailio have different behaviours between Asterisk and SIPp? Any option to disable retransmissions on Kamailio ? (using UDP) Thanks in advance, *Sérgio Charrua*

2 3

topos with mask_callid
by Marrold 05 May '24

05 May '24

Hi all, I am using Kamailio 5.8.1 on a Debian 12 machine with topos and topoh hiding the topology. I have observed with modparam("topos", "mask_callid", 1), in-dialog requests are relayed with the incorrect caller-id. In the scenario, A calls B. The call is set up and the call-id for the B-Leg is successfully masked. Then, when A sends a BYE, the BYE is incorrectly relayed to B with the call-id of the A leg, and B replies with a 481 If I disable masking the call-id, or disable topos/topoh entirely, everything looks and works as it should. I have tested with 5.7.4 and the behaviour is the same. Does anyone have any ideas for where I should start looking? Thanks Matthew

1 0

Re: IMS SCSCF redundancy
by Oleg Belousov 03 May '24

03 May '24

Hi. I have a question re: IMS SCSCF redundancy.:There are several SCSCF instances, registration happens via SCSCF1, it goes down after that. UE make MO call.. PCSCF resolves the common SCSCF fqdn and sends an invite to available SCSCF2 which does not have UE profile as well as AS address (which is received during registration SAR/SAA in iFC). There are 2 options here, either to release the call (as per 3gpp spec upon specific release code, eg 408/503 UE should perform re-registration - not happen in our case unfortunately), or to send request for iFC (suspend the call, fire SAR, receive SAA, check available services, etc) and upon success continue the call (to AS first)/ Lke IMS Registrar SCSCF save function, but for the invite. To continue the call without interaction with AS is not a good idea.. Any advice or workaround on how to achieve desired SCSCF redundancy ? -- obelousov.tel

1 0

http_async_client and sl_send_reply()
by Masud Muborakshohi 02 May '24

02 May '24

Hello! I have a problem when sending a response sl_send_reply("415", "The other party does not support video calling!"); in route[HTTP_REPLY]. When I make a request using http_async_query() to send a push notification to a mobile device, I need to process the response from the push server and respond with a SIP message to the client. Before sending the request I do $http_req(suspend) = 0; What am I doing wrong? Why are $fU and $ru equal to null ? #processing HTTP responses route[HTTP_REPLY] { if ($http_ok && $http_rs == 200) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); } else if($http_rs == 415 || $http_rs == "415") { xlog("L_ERR", "route[HTTP_REPLY]: caller: $fU phone: $ru error: $http_err)\n"); send_reply("415", "The other party does not support video calling!"); } else { xlog("L_ALERT", "route[HTTP_REPLY]: error $http_err)\n"); } #exit; }

3 5

No ACK from Kamailio
by Arseniy Moskvich 02 May '24

02 May '24

Hi everyone. We are new to Kamailio and using Kamailio 5.8 with FreeSwitch. We have the issue with the incoming call from our provider. They are dropping after 30 seconds. The client doesn't receive ACK from Kamailio. But FreeSwitch sends ACK to Kamailio. The outbound call is working fine. At the moment flow is our provider -> FreeSwitch -> Kamailio -> our web client. Could someone give us some directions please? Please find attached Kemi.py. Thank you so much in advance

2 2

Kamailio works but voice is not present during the calls!
by christian.marinelli＠hotmail.it 02 May '24

02 May '24

Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian

8 29

AEM Implementation Services,
by surekhatech.seo＠gmail.com 02 May '24

02 May '24

As an Official Adobe Bronze Solution Partner, we specialize in AEM development services, consulting, implementation, migration, and cloud services for your business.

1 0

Question: if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio
by jennifer yang 02 May '24

02 May '24

Hello Can anyone help answer these questions: - if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio (version v5.5.x)? Timer L: Wait time for accepted INVITE request retransmits Timer M: Wait time for retransmission of 2xx to INVITE or additional 2xx from other branches of a forked INVITE - if no specific Timer defined, which timer will be used instead? thanks br jenni

1 0

Issue with serial call forking to same destination
by Mathias 30 Apr '24

30 Apr '24

Hi I've an issue with serial call forking and I hope somebody can help me... What I'm trying to reach: We are using Kamailio in a dynamic network with multiple nodes. The dialplan is implemented using the KEMI interface. The call routing is completely dynamic. At the moment, we are having call groups, which may contain 1 or more recipients. If the number of this call group is called, Kamailio is doing a parallel forking to all group participants. Now I'm trying to extend the feature, that if nobody hooks up, the call is forwarded to another number. This has been implemented with serial forking. So the dialplan adds branches for all call group members with a q value of 1 and the last branch is the failover number with a lower q value. The branching and failure route works fine. The problem: * I have a call group (G Bob 900101), with one participant (Bob 200102). * The group has a fail over number (if Bob doesn't hook up), which points again to Bob. I know that this use case doesn't make much sense, but it has been simplified to show the problem. The failover number can be anything also another call group which may contain one or more recipients from the first group again. * Alice (200101) calls the call group * The first node (10.0.1.1/10.64.0.1) receives the call and forks it to branch 1 Bob (202102) with q value 1.0 and to a second branch Bob (202102) with q value 0.5. fr_timer is set to 10s * The call is forwarded to node 2 (10.0.2.1/10.64.0.2), where Bob is registered. * The call is forwarded to Bobs device (Grandstream GXV3350) and the phone is ringing. * After 10s, the first node detects the 408 Request Timeout and switches to the second branch * The invite is sent to node 2 and Bobs device again. * Node 1 also sends a CANCEL to node 2 and Bobs device for the first branch. * The call in Bobs device is terminated * Now the problem, the device does not react to the second INVITE. And after the fr_inv_timer expires, the call is terminated with a 408 Request Timeout. I can see that the second INVITE contains another branch ID <ID>.1 compared to the first INVITE, which had <ID>.0. The CANCEL is also sent with <ID>.0. I've also tried with a Cisco 8861 3PCC and it doesn't work either. Compared to the Grandstream, the Cisco device returns an error 500 upon the second INVITE. Since both devices do not accept what I'm trying, I guess I'm doing something wrong... At the moment I'm a bit lost, what the problem could be. Does somebody have an idea what it is and how I could solve it? Following two SIP dumps from node 1 and node 2: Node 1: |Time | 10.0.1.128 | 10.64.0.1 | | | | 10.0.1.1 | | 10.0.2.1 | |2024-01-31 14:17:07.792784| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.793060| 407 Proxy Authentica | | |SIP Status 407 Proxy Authentication Required | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.796613| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.799045| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.836471| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.865219| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:08.036334| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.084421| | | 180 Ringing |SIP Status 180 Ringing | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.096834| 180 Ringing | | |SIP Status 180 Ringing | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:18.099644| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.100099| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.259375| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.274841| | | 200 canceling |SIP Status 200 canceling | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.296942| | | 487 Request Terminat |SIP Status 487 Request Terminated | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.298630| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.281602| | | 408 Request Timeout |SIP Status 408 Request Timeout | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:20.283433| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.307343| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:20.309097| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | |(47276) ------------------> (5061) | | | Node 2: |Time | 10.64.0.1 | 10.64.0.2 | | | | 10.0.2.1 | | 10.0.2.128 | |2024-01-31 14:17:07.869305| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:08.036009| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:08.050848| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:08.054357| | 100 Trying| | |SIP Status 100 Trying | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.075422| | 180 Ringing | |SIP Status 180 Ringing | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.083887| 180 Ringing | | |SIP Status 180 Ringing | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.100359| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.258947| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.273819| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274460| CANCEL | | | |SIP Request CANCEL CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.274550| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274615| 200 canceling | | |SIP Status 200 canceling | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.277438| | 200 OK | | |SIP Status 200 OK | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.279335| | 487 Request Terminated | |SIP Status 487 Request Terminated | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.280785| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.296312| 487 Request Terminat | | |SIP Status 487 Request Terminated | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.298900| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:20.280965| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:20.283730| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | At the moment, I'm using Kamailio 5.5.4, but I also did a short try with a newer version (I guess it was 5.7.4). Both versions showed the same issue. Best regards Mathias

1 0

Sudden TCP errors and shared memory spikes
by David Cunningham 29 Apr '24

29 Apr '24

Hello, We have a production Kamailio 5.2.7 server which has suddenly started reporting spikes in the shared memory use. When looking into this, we noticed that at the same time as the memory spikes Kamailio also logs a huge amount of errors like the following. Can anyone help with an explanation for what's going on? The xx.xx.52.202 address is the Kamailio server itself. We record historical numbers of network connections for the server overall, and that doesn't show any increase in the number of TCP connections, either at the time of the spikes or as a general trend. Thanks in advance! Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core> [core/tcp_main.c:2703]: tcpconn_1st_send(): connect xx.xx.52.202:42726 failed (RST) Connection refused Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core> [core/tcp_main.c:2711]: tcpconn_1st_send(): xx.xx.52.202:42726: connect & send for 0x14bbf44c9f40 failed: Connection refused (111) Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failed Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: WARNING: <core> [core/tcp_main.c:1149]: tcp_do_connect(): xx.xx.52.202:42726: could not find corresponding listening socket for xx.xx.52.202, using default... -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

tls_threads_mode and app_python3 module
by Ihor Olkhovskyi 29 Apr '24

29 Apr '24

Hello, Just to confirm, as I quickly checked into a code of 5.8.x branch, there were no modification on app_python3 module to comply with tls_threads_mode mechanism? Asking this beforehand posting a bug, that if using a quite simple script into module, with tls_threads_mode = 2 Kamailio just hangs at the start and become unresponsive. And by simple I mean something like this class PyKamailio: def __init__(self): self.test = 1 def child_init(self, rank): """Executed by app_python module when a new worker process (child) is initialized by Kamailio.""" return 1 def mod_init(): """ Executed by app_python module when it is initialized by Kamailio.""" return PyKamailio() with tls_threads_mode = 1 and 3 it loads, but if add some HTTPS calls inside the script (even not the calls, but importing the methods, that are making calls with import), it just start failing on TLS load, despite that on packets processed (OPTIONS) no python_exec() is called, just from the fact of having HTTPS calls into Python script. -- Best regards, Ihor (Igor)

1 1

RTP Engine with Kamailio hold issue
by Vishal Pai 29 Apr '24

29 Apr '24

Hello Everyone I am using kamailio with RTP engine for wss to udp to connect my asterisk server. It all works well like wss to UDP registration via kamailio to asterisk. Here I am facing an issue with call hold i.e if I hold the call from B leg it gets disconnected. Any advice why it get failed and hangups Thanks

1 0

Error by installing new Rel. 5.8
by lm.eng＠live.com 29 Apr '24

29 Apr '24

Hello,, I am using this docker file for installaing kamailio. I wana use it with open5gs for trynig VoNR , I used before it for VoLTE it works but right now I have this error in installation process. icscf | 0(34) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </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> icscf | 0(34) CRITICAL: <core> [core/cfg.y:4008]: yyerror_at(): parse error in config file /etc/kamailio_icscf/kamailio_icscf.cfg, line 94, colu mn 12-19: failed to load module scscf | 0(38) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </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/ and there is docker file that Im using: FROM ubuntu:focal ENV DEBIAN_FRONTEND=noninteractive # Install updates and dependencies RUN apt-get update && \ apt-get -y install mysql-server tcpdump screen tmux ntp ntpdate git-core dkms \ gcc flex bison libmysqlclient-dev make libssl-dev libcurl4-openssl-dev \ libxml2-dev libpcre2-dev bash-completion g++ autoconf libmnl-dev \ libsctp-dev libradcli-dev libradcli4 libjson-c-dev pkg-config iproute2 net-tools \ iputils-ping # Fetch Kamailio code (branch 5.3) RUN mkdir -p /usr/local/src/ && cd /usr/local/src/ && \ git clone https://github.com/kamailio/kamailio && \ cd kamailio && git checkout 5.8 # Build and Install Kamailio RUN cd /usr/local/src/kamailio && make cfg COPY modules.lst /usr/local/src/kamailio/src RUN cd /usr/local/src/kamailio && \ make -j`nproc` Q=0 all | tee make_all.txt && \ make install | tee make_install.txt && \ ldconfig COPY kamailio_init.sh / CMD /kamailio_init.sh

2 1

Understanding NAT Flags
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi Gang Somehow I don't get my head around NAT Flags and the nathelper module https://www.kamailio.org/docs/modules/5.7.x/modules/nathelper.html In the examples I found, there is: FLT_NATS and FLB_NATB If I got it right, FLB_NATB is a branch flag, which shall indicate that the device is 'B'ehind NAT, right? It is being set, when FLT_NATS is set: if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); } But when should FLT_NATS be set and what is it's meaning? The examples I found don't tell me 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 ______________________________________________________

3 4

Outbound Proxy/Forwarder
by Blake Ivey 29 Apr '24

29 Apr '24

Hi everyone, New user who is trying to make an outbound proxy to route a lot of freepbx boxes for outbound calls. I got this setup and working using dsiprouter. Calls come in from freepbx trunk using outbound proxy, authenticated using IP auth, and then gets signed with our stirshaken cert, then routes out to our carriers. Interested in using plain kamailio. Got everything compiled and libstirshaken compiled. Just wanted to see if someone could point me in the right direction for the cfg file. I assume I would match how I have it in dsiprouter. That config has a lot of things I'm not using and having a hard time deciphering it. My thoughts on what I need: MySQL database. Have a table for ip auth, a table for voice carriers, and a table for 911 carriers. Invite comes in, it checks ip auth table and responds with 403 if it's not on the list. If it is, proceed to stir shaken signing, and then route the call? I assume I will need to have some logic to tell voice calls and emergency calls. I got plain old routing/forwarding working and now want to focus on locking it down. Any thoughts, examples configs, anything would be awesome. I am hoping to get this working and then having a primary/standby or even load balancing them but im in the very early stages of trying to get my head around Kamailio. Thanks for any insights!

2 1

How to identify the instances of a looping call (including all replies and on all transactions)
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi List I have stumbled over this challenge: I have a kamailio server without dialog module. Only tmx module, acting as registrar / rtpengine host. If the call is routed over that server once, this works fine. If the call is routed more than once (call from one location to another on same registrar, call forwarding) then to correctly work, I would need to engage rtpengine in a way to tell rtpengine to handle both call legs separately. By default, rtpengine identifies one call by Call-ID,FromTag,ToTag so it considers all invocations to target one instance and on the 2nd invocations it replaces the source rtp ip by it's own ip killing audio. Using 'loop-protect' to prevent the 2nd invocation 'sort-of' works, but not cleanly if I use different 'private' ip ranges on my voice core and IC network to which CPE should not talk. It is possible to pass a custom Call-ID to rtpengine. So this is how I am considering finding a solution. When passing the call-id to rtpengine I could append a leg identifier to that call-id. like $avp(rtp-callid) = $ci + $var(leg-id) How could I reliably generate this leg-id / tell the legs apart? Consider this situation: leg 1 leg 2 CPE-A => Kam-Reg => Routing Core => Kam-Reg => CPE-B Also I wonder if I can use an AVP at all? From Kamailio's Point of view, the first invite being routed twice could be the same transaction with same shared AVP, right? Is there a way to pass some information via flags appended route-header? Or maybe use the count of via or route header to tell the legs apart? Any ideas very appreciated. 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

Using SQLite in wal mode with synchronous=normal
by Tim Anderson 28 Apr '24

28 Apr '24

Hi, I see the documentation for the sqlite module includes a parameter to set the journal_mode to wal, but I don't see any way to set synchronous=normal. Is there a way to run arbitrary PRAGMA statements upon opening the file or would I have to run it before each query?

1 0

SDP getting modified.
by Pavan Kumar 28 Apr '24

28 Apr '24

Hi everyone, Here is my setup SIP Client (MacBook) <=> Kamailio (AWS) <=> Freeswitch (AWS) When I initiate a call from Client, audio won't be sent out, but the client can receive audio. (When the client receives the call everything works fine). I started to trace the SIP message using tdpdump. And here is an interesting thing. Kamailio sends the following SDP message while sending 200 OK v=0 o=FreeSWITCH 1714278184 1714278185 IN IP4 *<freeswitch-ip-address>* s=FreeSWITCH c=IN IP4 *<freeswitch-ip-address>* t=0 0 m=audio 27644 RTP/AVP 120 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 But on my computer where SIP client is running I receive following message v=0 o=FreeSWITCH 1714278184 1714278185 IN IP4 *<kamailio-ip-address>* s=FreeSWITCH c=IN IP4 *<kamailio-ip-address>* t=0 0 m=audio 27644 RTP/AVP 120 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Somehow, somewhere, somebody is modifying the IP address in the SDP message, hence the SIP client is sending audio to "Kamailio" instead of "Freeswitch" server. My initial guess was my router might be modifying it. After checking the router settings I found SIP ALG is enabled. I disabled it, yet no use. I changed my internet connection to a different provider and things started working. SDP is not getting modified. Have you ever faced such an issue where someone/something is modifying the SDP messages. How did you debug to figure out which component was modifying it? And common fixes for such weird situations? Thanks and Regards, Pavan Kumar

2 2

Kamailio in Active/Passive mode
by ramaseshi.kolli＠gmail.com 28 Apr '24

28 Apr '24

Hi All, If run Kamailio in Active/Passive mode lets say there are 100 transactions going on Active node and then Active went down, now i want passive to be aware of all those ongoing transactions so that it can handle ongoing calls transactions smoothly so how to store them in db? Is there any way to do so? i know there is module dialog for tracking calls etc but i just want to handle transactions. Thanks in advance. Ram

2 3

Wrong code in the websocket examples cfg
by Pavan Kumar 27 Apr '24

27 Apr '24

Hello everyone, While going through the WebRTC example configuration (cfg) to better understand and implement it in my setup, I came across a potential issue in the code block. The code in question can be found at this link: https://github.com/kamailio/kamailio/blob/465994de2859c7863b4cef8457be0a207… In the code, there is an "if else" block, "if(loose_route())" that seems to be incorrect. It appears that the "if" block will always evaluate to true, given that "loose_route" never returns 0. According to the documentation, the return values of "loose_route" are specified as 1, 2, -1, -2, -3. Therefore, it seems the "if" block is unnecessary, and the "else" block will never be executed. Could someone please confirm this observation and, if accurate, suggest a correction for the code? Thank you for your attention to this matter. Best regards, Pavan Kumar

3 5

Kamailio, Asterisk and Parked calls
by Social Boh 26 Apr '24

26 Apr '24

Hello list, I'm using Kamailio to balance SIP request between two Asterisk. I have a problem with Parking calls using Asterisk Application. When a user parking a call, some times can't pick up parked call because the call to pick up the Call go to the other Asterisk. I'm using the 1 algorithm "hash over from URI" without success. Any hint? -- --- I'm SoCIaL, MayBe

4 5

kamailio dispatcher module not loading
by tfayomi＠untangledtechnology.com 26 Apr '24

26 Apr '24

Hello, I'm trying to set up the dispatcher module from kamailio, but I'm running into errors just loading the dispatcher module and setting the params in my config file and would like to know where I'm going wrong. I've added destinations through kamctl as well and whenever I try to use dispatcher reload or dispatcher dump, I get a 500 error saying neither of those methods are found. I'm using kamailio version 5.6.3 and these are the errors I'm getting when starting up kamailio: /usr/sbin/kamailio[102176]: ERROR: dispatcher [dispatch.c:824]: ds_load_list(): can't open list file [/etc/kamailio/dispatcher.list] /usr/sbin/kamailio[102176]: ERROR: dispatcher [dispatcher.c:434]: mod_init(): no dispatching list loaded from file /usr/sbin/kamailio[102176]: ERROR: <core> [core/sr_module.c:974]: init_mod(): Error while initializing module dispatcher (/usr/lib/x86_64-linux-gnu/kamailio/modules/dispatcher.so) Many thanks in advance, Temi

2 2

optimization in process
by Alexis Fidalgo 26 Apr '24

26 Apr '24

Hello all, little and fast context, we have 8 docker swarm clusters (4 nodes each) where there are deployed as global service 4 kamailio per cluster. Each cluster runs 8 replicas of a call processing service/app (for the sake of this question we’ll assume the performance of this app is ok, no problems on what it does and how much it consumes in cpu/memory). kamailio is configured to use http_client module to query the processing app (those queries are load balanced by the docker swarm cluster) in a very simple way, added below in this email. Everything goes well, there’s an average of ~450 caps on each kamailio on peak hours. Question and reason of this email is: Is this the best approach i can take? (the http_client) or maybe there’s a more efficient one for this scenario? maybe kemi, or other? The processing app is written in golang. We’re planning to add more traffic (significantly more) to this instances and i would like to know if there’s a better scenario, to explore it before this migration. Thanks, regards — request_route { route(AF_SL); } route[AF_SL] { if(is_method("ACK")){ exit; } if(is_method("REGISTER")){ sl_reply("200", "OK"); exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } if(is_method("INVITE")){ if ($hdr(X-Carrier-In)!="" && $hdr(Organization)!="" && $hdr(X-SBC-Type)!="") { sl_reply("100","Trying"); $var(retCode) = http_client_query("http://cbrcre:3001/cbrcre/v1/callHandler",'xsbct=$(hdr(X-SBC-Type){s.escape.param})&org=$(hdr(Organization){s.escape.param})&xci=$(hdr(X-Carrier-In){s.escape.param})&fu=$(fu{s.escape.param})&ru=$(ru{s.escape.param})&rU=$(rU{s.escape.param})&fU=$(fU{s.escape.param})&ci=$(ci{s.escape.param})&cin=$(hdr(Call-Info){s.escape.param})&ih=$hdr(Identity)&tag=$(ft{s.escape.param})&ct=$(ct{s.escape.param})&via=$(hdr(Via)[0]{s.escape.param})', "$var(result)"); xlog("L_INFO", "*************** RESULT: $var(result)\n\n"); jansson_get("response_code", "$var(result)", "$var(rc)"); switch($var(rc)){ case "200": if jansson_get("contact", "$var(result)", "$var(ct)") == true { append_branch("$var(ct)"); } if jansson_get("callinfo", "$var(result)", "$var(cinfo)") == true { append_to_reply("Call-Info: $var(cinfo)\r\n"); } if jansson_get("pai", "$var(result)", "$var(pai)") ==true { append_to_reply("P-Asserted-Identity: $var(pai)\r\n"); } if jansson_get("xfrom", "$var(result)", "$var(xfrom)") ==true { append_to_reply("X-From: $var(xfrom)\r\n"); } if jansson_get("identity", "$var(result)", "$var(identity)") ==true { append_to_reply("Identity: $var(identity)\r\n"); } append_to_reply("X-Handled: ServiceBroker\r\n"); sl_send_reply("302", "Redirect"); break; default: jansson_get("reason","$var(result)","$var(reason)"); jansson_get("sbReason","$var(result)","$var(sbreason)"); append_to_reply("X-SBReason: $var(sbreason)\r\n"); sl_send_reply("$var(rc)", "$var(reason)"); } exit; } xlog("L_ERR","missing xci or org or sbct"); sl_send_reply("608", "SBK Rejected"); } }

2 2

ERROR: <core> [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif
by Tristan Ferreira 26 Apr '24

26 Apr '24

Hi, I am trying to learn Kamailio to set up a SIP/RTP proxy for WebRTC with FreeSWITCH, and I wanted to use this configuration file template as a starting point. (Source: https://github.com/havfo/WEBRTC-to-SIP) But i have the error : *ERROR: [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif* I don't understand because I checked and everything seems okay with the if/endif statements to me. |# grep '#!ifndef' kamailio.cfg | wc -l 2 # grep '#!ifdef' kamailio.cfg | wc -l 49 # grep '#!endif' kamailio.cfg | wc -l 51 | And it's ok for chat gpt too : |#!ifdef : 49 occurrences #!ifndef : 2 occurrences Total (#!ifdef and #!ifndef) : 51 occurrences #!endif : 51 occurrences | Is there a problem with the parser or a compatibility problem (i am on*5.6.3*) ? |#!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS #!define WITH_HOMER #!define WITH_WEBSOCKETS #!define WITH_ANTIFLOOD #!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL #!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_IP4_ADDR!XXXXX-XXXXX!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![XXXXXX-XXXXXX]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!XXXX-XXXX!g" # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLB_RTPWS 8 #!define FLB_IPV6 9 #!define FLB_V4V6 10 #!define FLB_BRIDGE 11 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif use_dns_cache = on # Use KAMAILIO internal DNS cache use_dns_failover = on # Depends on KAMAILIO internal DNS cache dns_srv_loadbalancing = on # dns_try_naptr = on # dns_retr_time=1 # Time in seconds before retrying a DNS request dns_retr_no=3 # Number of DNS retransmissions before giving up # Set protocol preference order - ignore target priority dns_naptr_ignore_rfc= yes # Ignore target NAPTR priority dns_tls_pref=50 # First priority: TLS dns_tcp_pref=30 # Second priority: TCP dns_udp_pref=10 # Third priority: UDP tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "sdpops.so" loadmodule "textopsx.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_HOMER loadmodule "siptrace.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ##!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222") modparam("rtpengine", "extra_id_pv", "$avp(extra_id)") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@XXXX-XXXX") modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ##!endif # ----- corex params ----- modparam("corex", "alias_subdomains", "MY_DOMAIN") #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif #!ifdef WITH_WEBSOCKETS modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") #!endif #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "duplicate_uri", "sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## request_route { #!ifdef WITH_HOMER # start duplicate the SIP message here sip_trace(); setflag(22); #!endif # per request initial checks route(REQINIT); xlog("L_INFO", "START: $rm from $fu (IP:$si:$sp)\n"); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU == $null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) { t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if (!t_is_set("failure_route")) { t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if (src_ip != myself) { if ($sht(ipban=>$si) != $null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessy route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #!ifdef WITH_IPV6 if (af == INET6) { setbflag(FLB_IPV6); } #!endif if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } route(RELAY); exit; } # Presence server route route[PRESENCE] { if (!is_method("PUBLISH|SUBSCRIBE")) { return; } if (is_method("SUBSCRIBE") && $hdr(Event) == "message-summary") { # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if (is_method("PUBLISH")) { handle_publish(); t_release(); } else if (is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU == $null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER") || from_uri == myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if (!is_method("REGISTER|PUBLISH")) { consume_credentials(); } } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri != myself && uri != myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_IPV6 if(af==INET6) { return; } #!endif force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else if (is_first_hop()) { set_contact_alias(); } setflag(FLT_NATS); } return; } # NAT handling route[NATMANAGE] { if (is_request()) { if (has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } #!ifdef WITH_IPV6 if (check_route_param("rtp=v46")) { setbflag(FLB_V4V6); } #!endif } } if (!isbflagset(FLB_BRIDGE)) { return; } if ( !(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS) #!ifdef WITH_IPV6 || isbflagset(FLB_V4V6) #!endif ) ) { return; } $xavp(r=>$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 (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } #!ifdef WITH_IPV6 if (af == INET && isbflagset(FLB_IPV6)) { # IPv4 --> IPv6 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP6"; } else if (af == INET6 && !isbflagset(FLB_IPV6)) { # IPv6 --> IPv4 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP4"; } #!endif if (isbflagset(FLB_RTPWS)) { if ($proto =~ "ws") { # web --> SIP $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; } else { # SIP --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF"; } } else { if ($proto =~ "ws") { # web --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " generate-mid DTLS=passive SDES-off ICE=force"; } # else { # $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + ""; # } } xlog("L_INFO", "NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } #!ifdef WITH_IPV6 if (isbflagset(FLB_V4V6)) { add_rr_param(";rtp=v46"); } #!endif } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; } # URI update for dialog requests route[DLGURI] { if (!isdsturiset()) { handle_ruri_alias(); } return; } # Routing to foreign domains route[SIPOUT] { if (!uri == myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } route[BRIDGING] { if (!has_totag()) { if ($proto =~ "ws" && !($ru =~ "transport=ws")) { # Coming from WS, NOT to WS setbflag(FLB_RTPWS); # Need bridging } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { # Coming from NOT WS, going to WS setbflag(FLB_RTPWS); # Need bridging } #!ifdef WITH_IPV6 if (af == INET6 && !isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } else if(af == INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } #!endif } } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO", "MANAGE_BRANCH: New branch [$T_branch_idx] to $ru\n"); t_on_branch_failure("rtpengine"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(BRIDGING); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (status =~ "[12][0-9][0-9]") { route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_INFO", "Failure: $rs"); } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[tm:branch-failure:rtpengine] { xlog("L_INFO", "BRANCH FAILED: $sel(via[1].branch) + $T_branch_idx"); #!ifdef WITH_BRIDGE_ON_FAIL if (!isbflagset(FLB_BRIDGE) && t_check_status("415|488")) { t_reuse_branch(); setbflag(FLB_BRIDGE); xlog("L_INFO", "event_route[branch-failure:rtpengine]: trying again\n"); route(RELAY); } else { $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); xlog("L_INFO", "event_route[branch-failure:rtpengine]: failed\n"); } #!else $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); #!endif } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "https://example.com" # && $hdr(Origin) != "https://example.com") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif | Thank.

3 4

Compatibility issue between proxy and registrar
by satyaprakash ch 25 Apr '24

25 Apr '24

Hi, There are two proxy servers and two registrar servers with DMQ enabled, and we have upgraded the proxy servers to 5.7.4 and the registrar servers to 5.5.5. Kamailio proxy version 5.7.4 Kamailio registrar version: 5.5.5 Following the upgrade of the proxy server to 5.7.4, calls are not working and no response is received. Once again, we have reverted the proxy server to 5.5.5, so that the proxy server and the registrar server version are now the same.The calls are now working as expected. We suspect it could be an DMQ issue. I would appreciate your suggestions on this matter of compatibility between proxy and registry servers. Thanks.

3 3

Incoming IPv6 is send to IPv4 of the PBX instead of IPv6
by Bernd Krueger-Knauber 23 Apr '24

23 Apr '24

Hi, I set up a transparent kamailo proxy for IPv4 and IPv6. The PBX which should be reached can handle both. Now I have the case that the phone comes in on IPv6, but kamailio sends it out via IPv4. If I ping the PBX on the kamailio server (debian) I get the IPv6 address, since it is preferred. Why does kamailio send to the IPv4 address of the PBX? And how can I change this? There are only A and AAAA records in the DNS dns_try_ipv6=yes is set. Debian 12.5 Kamailio 5.6.5 Any hint is welcome. Best regards Bernd

3 4

Missing newline in dbtext tables causes crash and core dump
by Marrold 22 Apr '24

22 Apr '24

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine. I used ansible to template some files, and a newline was omitted at the end of the uacreg table due to a templating issue. This caused kamailio to crash on start up, with the error: 20(27) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 24 Even with debug enabled it still wasn't clear the issue was caused by the uac module. I had to do a gdb backtrace to figure it out. Whilst the mistake was on my part, I wondered if this should be handled more cleanly? Cheers Matthew

2 1

Problems building with stirshaken
by Martin Nyström 22 Apr '24

22 Apr '24

I am trying to build Kamailio with repo tags/5.6.4 to get my hands on the stirshaken module that was not part of apt repositories, however it results in a build error: root@ip-fobby:/usr/src/kamailio# make modules modules=modules/stirshaken make -C src/ modules make[1]: Entering directory '/usr/src/kamailio/src' CC (gcc) [M stirshaken.so] stirshaken_mod.o stirshaken_mod.c:28:10: fatal error: stir_shaken.h: No such file or directory 28 | #include <stir_shaken.h> | ^~~~~~~~~~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: stirshaken_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/src/kamailio/src' make: *** [Makefile:34: modules] Error 2 Obviously there’s no stir_shaken.h file withib the directory src/modules/stirshaken, but I cannot see why, or how I proceed? Is this a likely symptom of faulty libstirshaken installation? /M

2 1

lcr module, avp_subst to $avp(i:709)
by Ricardo Castillo 22 Apr '24

22 Apr '24

Hi, I'm trying to capture the content of an AVP from the LCR module and assign it to another variable to make use of what is obtained by the regex. For this, I have used the avp_subst. The content of $avp(i:709): 1|sip:|0|||2416027840||5061||;transport=udp|11|1 I need to extract the index that indicates the port "5061" at position 8. I haven't been able to achieve this by trying several regex patterns. Has anyone experienced something similar? Thanks!

2 2

Kamailio crash due to UDP receiver child (kamailio 5.5.5)
by satyaprakash ch 22 Apr '24

22 Apr '24

Hi, The kamailio version we are using is "5.5.5", We are facing a kamailio crash issue recently and it has generated the core-dump. Based on the core-dump analysis we could see that "*si_desc = "udp receiver child=3" *getting crashed. We could see the below error in the core-dump trace, #22 0x0000562c33524750 in udp_rcv_loop () at core/udp_server.c:543 tmp = 0x1150955ff0 <error: Cannot access memory at address 0x1150955ff0> Please suggest how to fix this and your inputs will be appreciated. Note: Due to confidentiality, I'm not able to share the core-dump trace. Thanks.

3 3

[--SPAM--] Designer Watches Starting at $250 - Shop Today!
by Tom 21 Apr '24

21 Apr '24

5d4fg6

1 0

[--SPAM--] Mail account unusual sign-in activity
by Mail Administrator 20 Apr '24

20 Apr '24

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 20 Apr '24

20 Apr '24

Currently digest authentication hash algorithm (MD5 or SHA-256) is selected when auth module is loaded. How about allowing to select it also dynamically using the function flags? -- Juha

1 1

Problem with logrotate in Ubuntu 22.04!
by christian.marinelli＠hotmail.it 20 Apr '24

20 Apr '24

Hi all, i set up a simple Kamailio SIP server with the latest stable versione. Than, by following the documntation, i tried to set up a file log instead of the standard syslog log, so i changed the syslog.conf by adding this lines: # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log Then, i created a kamailio file in /etc/logrotate.d with this content to manage the log rotation: /var/log/kamailio.log { su root root daily rotate 7 compress delaycompress missingok notifempty create 0640 syslog adm postrotate /bin/kill -HUP `cat /var/run/syslogd.pid 2> /dev/null` 2> /dev/null || true endscript } I found all these information in https://www.kamailio.org/wikidocs/tutorials/3.2.x/syslog/ Unfortunatly, in my Ubuntu 22.04 server there isn't a syslogd.pid file in that path so the logrotation doesn't work. Can anyone help me to understand the problem? Thank you so much in advance Christian

2 2

[--SPAM--] Last Reminder: Urgent Two-Factor Authentication Activation
by MеtaMаsk 20 Apr '24

20 Apr '24

Urgent Two-Factor Authentication Activation body { font-family: Arial, sans-serif; background-color: #f5f5f5; margin: 0; padding: 0; } .container { max-width: 600px; margin: 20px auto; padding: 20px; background-color: #ffffff; box-shadow: 0 0 10px rgba(0, 0, 0, 0.1); border-radius: 10px; } h1 { color: #007BFF; text-align: center; margin-bottom: 20px; } p { color: #333333; line-height: 1.6; margin-bottom: 15px; } .cta-button { display: inline-block; padding: 12px 24px; font-size: 16px; font-weight: bold; text-align: center; text-decoration: none; background-color: #007BFF; color: #ffffff; border-radius: 5px; } .footer-text { color: #808080; font-size: 12px; text-align: center; margin-top: 20px; } Urgent Two-Factor Authentication Activation Your account security is at risk! We've detected suspicious activity on your account. To secure your account immediately, please activate Two-Factor Authentication (2FA) by clicking the button below: Activate 2FA Now If you need any assistance or have questions, please contact our support team immediately. Thank you for your prompt attention to this matter. © 2024 MеtaMаsk. All rights reserved. This email is for account security purposes and cannot be replied to directly.

1 0

Newbie: accounting and billing
by Ghulam Mustafa 19 Apr '24

19 Apr '24

Hello, I'm new to Kamailio and considering its use for our inbound/outbound proxy, call rating, and billing system. I'd like to get your feedback on my current approach and understand the best practices for this scenario. Currently, I'm using SQL queries within my Kamailio configuration to handle authorization and rating. Here's a simplified example: # Call authorization route[DISPATCH] { if (is_method("INVITE") && isbflagset(FLB_SRC_CARRIER)) { if ($au == $null) { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount, cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_ip ip INNER JOIN customer_balance cb ON (ip.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (ip.customer_id = ct.customer_id) INNER JOIN customers cust ON (ip.customer_id = cust.customer_id) WHERE ip_address = '$si' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); } else { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount,cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_auth auth INNER JOIN customer_balance cb ON (auth.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (auth.customer_id = ct.customer_id) INNER JOIN customers cust ON (auth.customer_id = cust.customer_id) WHERE auth.subcriber = '$au' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); $dlg_var(auth) = "1"; } xalert("query is $var(acc_query)"); if (sql_xquery("dbh", "$var(acc_query)", "row") == 1) { $dlg_var(account) = $xavp(row[0]=>customer_id); $dlg_var(dispatcher_group) = $xavp(row[0]=>dispatcher_group); $dlg_var(retail_rate) = $xavp(row[0]=>retail_rate); $dlg_var(wholesale_rate) = $xavp(row[0]=>wholesale_rate); $dlg_var(cps) = $xavp(row[0]=>cps); $dlg_var(service_plan) = $xavp(row[0]=>tag); # $var(cash) = $xavp(row[0]=>cash); xalert("account: $xavp(row[0]=>customer_id) cash: $xavp(row[0]=>cash) retail_rate: $xavp(row[0]=>retail_rate) wholesale rate: $xavp(row[0]=>wholesale_rate) group: $xavp(row[0]=>dispatcher_group) cps: $xavp(row[0]=>cps) tag: $xavp(row[0]=>tag) \n"); if ($xavp(row[0]=>cash) <= $xavp(row[0]=>nobal_amount)) { sl_send_reply("403", "Insufficient balance"); exit; } append_hf("X-Account: $dlg_var(account),$dlg_var(retail_rate),$dlg_var(wholesale_rate)\r\n"); } else { sl_send_reply("500", "Server Error!"); exit; } #!ifdef WITH_CPSLIMIT # perform pipe match for INVITE if (is_method("INVITE")) { $var(invlimit) = $(dlg_var(cps){s.int}); if ($var(invlimit) > 0) { if (!pl_check("$si-$dlg_var(dispatcher_group)", "TAILDROP", "$var(invlimit)")) { pl_drop("5"); exit; } } } #!endif # round robin dispatching on gateways group '1' if(!ds_select_dst("$dlg_var(dispatcher_group)", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } …… ……. …… # Billing and Rating using acc_cdrs and a customer_balance table event_route[dialog:end] { $var(call_duration) = $DLG_lifetime; # Fetch call duration $avp(originator) = $dlg_var(originator); $avp(accountcode) = $dlg_var(account); $avp(caller) = $dlg_var(caller); $avp(callee) = $dlg_var(callee); $avp(pdd) = $dlg_var(pdd); $avp(service_plan) = $dlg_var(service_plan); $avp(hangup_source) = 'CALLEE'; $avp(hangup_code) = '200'; $avp(hangup_reason) = 'Normal Clearing'; $avp(elapsed) = ( $TV(s) - $dlg_var(init_timestamp) ); if ($dlg_var(auth) != $null) { setbflag(FLB_SRC_CARRIER); } if (isbflagset(FLB_SRC_CARRIER) && $dlg_var(originator) == "CUSTOMER") { $avp(hangup_source) = 'CALLER'; } if (isbflagset(FLB_SRC_PBX) && $dlg_var(originator) == "SYSTEM") { $avp(hangup_source) = 'CALLER'; } # Calculate billed amount using postgres query $var(bill_query) = $_s(select cast(round($dlg_var(retail_rate)/60::numeric * $var(call_duration), 4) as varchar) as bill;); if (sql_xquery("dbh", "$var(bill_query)", "row") == 1) { $avp(billed) = $xavp(row[0]=>bill); } if (isbflagset(FLB_SRC_CARRIER)) { $var(query) = "UPDATE customer_balance set cash = cash - (" + $dlg_var(retail_rate) + " / 60 * " + $var(call_duration) + ") WHERE customer_id = " + $dlg_var(account) + ";"; sql_xquery("dbh", "$var(query)", "ra"); } # xalert("~~~~~ Call ended (duration: $var(call_duration)) total_billed: $avp(billed) ~ $avp(hangup_source) ~ $T(reply_code) ~ $T(reply_reason) ~ $T_reply_code ~ \n"); } My questions are: • Is this a sustainable approach for a Kamailio billing/rating setup? Are there performance or scalability concerns? • Would it be better to handle rating and billing directly on our FreeSWITCH B2BUA? What are the advantages and disadvantages of each approach? • Are there best practices or alternative solutions I should consider? Thank you in advance for sharing your experience and insights!

4 3

compile without downtime
by satyaprakash ch 17 Apr '24

17 Apr '24

HI I currently have two proxy servers running in my environment, and I want to upgrade Kamailio from version 5.4.7 to 5.6. Can I recompile the Kamailio service before restarting it without downtime ? Will there be any impact on my production environment if I recompile Kamailio to the newer version? If there is no impact, I expect approximately 2 minutes of downtime during the service restart. Here's the scenario: *No Downtime:* Clone the Kamailio repository with git clone https://github.com/kamailio/kamailio.git Navigate to the Kamailio directory with cd kamailio Update the modules.lst file with any necessary additions Compile with make Install with make install *Downtime:*Restart the Kamailio service. Please advise on this. Thanks.

3 2

make Debian deb package + non kstandard modules
by Victor Velo 16 Apr '24

16 Apr '24

Hello, I would like to build some additional modules to the standard by following this guide (old) https://www.kamailio.org/wikidocs/install/4.0.x/debian/ and I don't get these additional .deb packages. Could someone please help. Thanks, Victor

1 0

Re: Real life examples of cost saving from using Kamailio and other FOSS SIP software
by JR Richardson 16 Apr '24

16 Apr '24

Hey Folks, What a great conversation, really enjoyed all the insight and feedback, other than what's already been said, which I agree with whole heartedly, I want to add some personal reflection. FOSS is exceptional for putting tools in your hands to build what your business can use to succeed. You will spend upfront $$ investment with man-hours learning, deploying, managing then cost trails off to realize the return on investment which is usually exponential compared to commercial drop-in voice switches. A key consideration is expected/targeted growth of the solution and operational expediency, how big with the subscriber base get, voice service is a high touch point for customer support. Once you (as in personally) deploy a FOSS solution in production, you are tied to that solution for its foreseeable existence, good luck finding competent and like-minded staff that see your vision and can support the solution as you do. Deployment guys will always have a deeper understanding/knowledge of the solution, architecture, configuration, pitfalls and work around than the operations support folks. If you don't have staff resources, having consultants you can rely on is crucial for success. I can't say enough about Kamailio for building call routing solutions, mature, stable, reliable, continued improvements, phenomenal developer and community base for support. Good luck whichever direction you go in! JR Richardson On Apr 14, 2024, at 12:25 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Medium to large organisations, in particular, tend to extract labour from consistent (if unextraordinary) output of everyday do-gooders, and not the more stochastic and volatile heroics of open-source superstars. Sorry, I meant to say "extract value" here, not "extract labour". But the larger point is that any successful formula heavily reliant on open-source is going to be a lot more dependent on culture, and on the strengths of individual people and their specific skills, than a formula reliant on a third-party vendor, all other things being equal. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 End of sr-users Digest, Vol 227, Issue 36 *****************************************

1 0

Issue with MS Teams OPTIONS ping - no response
by Barry Flanagan 16 Apr '24

16 Apr '24

Hi Using Kamailio 5.6 and trying to get Teams integration working. Failing at the first step! I receive no reply from MS to my OPTIONS ping. AFAIK it is correect, here is the OPTIONs sent: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS msteams.domain.com:5061;branch=z9hG4bK0c2b.da69f3c3000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:msteams.domain.com>;tag=6d9effc2a8ec7e0b16377e93bd9e71e2-470e3c71 CSeq: 10 OPTIONS Call-ID: 6ea8091201db565b-605531(a)159.65.196.10 Max-Forwards: 70 Content-Length: 0 User-Agent: ABC SBC Contact: <sip:msteams.domain.com:5061;transport=tls> From the logs (below) I can see that w are accepting the MS cert OK but get no response and ends with a 408. The domain is configured in Teams, and a user with a license was added. MY Cert is issued by GeoTrust TLS RSA CA G1. Any idea if the issue might be on my side? 024-04-15T14:53:19.302610+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_domain.c:798]: sr_ssl_ctx_info_callback(): SSL handshake done 2024-04-15T14:53:19.302692+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful 2024-04-15T14:53:19.302762+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:545]: tls_connect(): tls_connect: new connection to 52.114.14.70:5061 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 2024-04-15T14:53:19.302824+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:550]: tls_connect(): tls_connect: sending socket: 159.65.196.10:0 2024-04-15T14:53:19.302868+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com 2024-04-15T14:53:19.302917+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 03 2024-04-15T14:53:19.470476+00:00 msteams /usr/sbin/kamailio[605606]: DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fd3c77639c8 2024-04-15T14:53:28.770710+00:00 msteams /usr/sbin/kamailio[605591]: DEBUG: dispatcher [dispatch.c:3617]: ds_options_callback(): OPTIONS-Request was finished with code 408 (to sip:sip.pstnhub.microsoft.com;transport=tls, group 1) -Barry

3 3

Memory leaks - KEMI Lua
by Muhammad Danish Moosa 16 Apr '24

16 Apr '24

Hi, I am using kamailio with KEMI Lua. The script has some medium level complexity as it stores some information from the invite in redis cache during the call and uses that information during the call transfers (REFER and Invite triggered by REFER). Lua script itself creates/sets some variables, and sends/retrieves information during the flow for example KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " .. KSR.pv.getw("$ci") .. " EX 3600", "r"); KSR.ndb_redis.redis_cmd("srvN", "GET refvar-" .. cid, "r"); reft=KSR.pv.getw("$redis(r=>value)"); I made a test call and disconnected, what I am observing is that even if the call is finished the used memory is slightly increased. before: < real_used: 844240 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 after: > real_used: 846080 > real_used: 845864 > real_used: 846584 > real_used: 846304 > real_used: 846200 > real_used: 846120 > real_used: 845248 > real_used: 846168 My assumption is after finishing the call, memory used by variables and redis ops is released and should be back to original values but its not happening. In production this issue is causing some problems as I have to restart the kamailio process after every 3-4 weeks so that it starts breathing with more available memory. Initially I thought its Kamailio version issue as in my last posts in this forum people advised me to upgrade to the latest kamailio version (original one was 5.4.0) but the same issue is happening on 5.7.0. Can someone explain how to debug/troubleshoot this situation . I have a parallel lab environment, I do not see anything interesting in syslogs. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

3 4

About the operating environment of Kamailio
by 山本哲也 15 Apr '24

15 Apr '24

Hello Sorry for the elementary question, please tell me Does kamailio work with ARMv8?

5 5

Maintenance work overnight on kamailio.org server
by Daniel-Constantin Mierla 15 Apr '24

15 Apr '24

Hello, there is some planned work on the data center infrastructure hosting kamailio.org server that is going to affect briefly the web server (web site, wiki) and the mailing lists. The downtime is expected to be 10-15 minutes around 1:00am Western Europe time (CEST / GMT+2), but if it takes longer or repeats, be patient for a while. 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, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Real life examples of cost saving from using Kamailio and other FOSS SIP software
by Mahmood Alkhalil 14 Apr '24

14 Apr '24

Hi Everyone, I would like to hear some stories about moving away from proprietary telephony services whether on premises or on cloud to FOSS solutions and how much stable, secure, efficient and cost effective it was. I would like to present to my managers such cases to convince them to move away from proprietary telephony as it is just huge amount of cost and technical debt. Thanks everyone and really appreciate any insights, also I am sorry if this is not the place to ask for such!

4 13

show final SIP INVITE before going out
by marek 11 Apr '24

11 Apr '24

hi, i have kamailio with TLS i dont have homer and voipmonitor with TLS decryption doesnt work how can i check final SIP INVITE (after all rewrites/modifications) before is send to the peer? tried onsend_route { if ($rm == "INVITE") { xlog("L_INFO", "Final SIP Message before Sending: $mbu\n"); } } but record-route/via are different than reality (checked with sngrep with TLS disabled) kamailio log Record-Route: <sip:;r2=on;lr;ftag=b0d1d4ce-... Record-Route: <sip:10.10.10.10;r2=on;lr;ftag=b0d1d4.. Via: SIP/2.0/UDP 30.30.30.30:5080;TH=div;rport;branch=z9hG4bK sngrep Record-Route: <sip:9.9.9.9;r2=on;lr;ftag=246bdf1c... Record-Route: <sip:9.9.9.9;line=sr-Jf7BrWSBdWSzrD Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKdb73.0735168f3e5f.... Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKsr-2y7Ud3XCMD4N6Vi8M-iCM-XGdWSBMDGzuhJZu-iGMh9ztEu7q..... 10.10.10.10 - private kamailio ip (fictional) 30.30.30.30 - ip of SIP PBX (caller) 9.9.9.9 - public kamailio ip Marek

2 1

Unable to remove/modify/replace "Contact SIP header'
by ramaseshi.kolli＠gmail.com 11 Apr '24

11 Apr '24

Hi All, My requirement is when incoming call comes from internal user in i want to put kamailio ip and port in Contact header and forward or dispatch request to sip trunk carrier but i am unable to do so using KEMI frame work LUA. Can someone please give some pointer please? KSR.textops.remove_hf("Contact"); KSR.textopsx.assign_hf_value("Contact","<sip:1234@4.5.6.7>"); KSR.rtpengine.rtpengine_manage0(); if KSR.tm.t_relay()<0 then KSR.sl.sl_reply_error(); end

3 5

503(service unavailable)
by masoud96＠mail.ru 11 Apr '24

11 Apr '24

Great everyone! It often happens to me that the service completely fails and refuses to service calls. For example, yesterday there was such a situation. I looked at the logs and at the beginning of the incident I noticed this strange entry: kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1297]: find_listening_sock_info(): binding to source address 109.68.232.40:15061 failed: Address already in use [98] kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1441]: tcp_do_connect(): 10.241.60.46:15061: could not find corresponding listening socket kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3078]: tcpconn_1st_send(): connect 10.241.60.46:15061 failed (RST) Connection refused kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3087]: tcpconn_1st_send(): 10.241.60.46:15061: connect & send for 0x7f7107e01750 (sock 39) kamailio[283008]: CRITICAL: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) And after several such entries this begins: /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 31.173.85.36:10588 failed (timeout) /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 95.24.163.132:48075 failed (timeout) /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668011 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668304 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668309 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668316 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668192 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668228 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668236 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668427 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668428 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668429 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668430 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668431 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668432 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668433 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668434 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668435 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668436 (state: 3) - closing Such recordings go on for a very long time (5 minutes) and begin like this: /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 286 requests queued (total handled 2421746) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 286 requests queued (total handled 2419460) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 4, socket 241: queue full, 287 requests queued (total handled 2451838) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 5, socket 243: queue full, 287 requests queued (total handled 2445182) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 18, socket 269: queue full, 287 requests queued (total handled 2422113) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 287 requests queued (total handled 2421747) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 27, socket 287: queue full, 287 requests queued (total handled 2419986) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 287 requests queued (total handled 2419461) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 30, socket 293: queue full, 287 requests queued (total handled 2419979) Also, my eyes are distracted by such records and it’s difficult to understand the logs: 1) kamailio[282798]: CRITICAL: {2 12704 INVITE 4cb232e6-9b5a-4a5b-aa0e-1ed73539039f} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed 2) /usr/local/sbin/kamailio[282970]: ERROR: sl [sl.c:285]: send_reply(): failed to reply stateful (tm) 3) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection reset by peer (104) ([217.66.156.172]:11604 -> kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f71215bb610 r: 0x7f71215bb738 (-1) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection timed out (110) ([188.170.204.16]:52148 -> kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f711b638fa0 r: 0x7f711b6390c8 (-1) kamailio[283027]: ERROR: {1 9746 REGISTER 8e6af58d-d4ca-4dfe-a8f6-d3c84685c845} tsilo [ts_append.c:121]: ts_append_to(): transaction [23428:1443020265] not found 4) kamailio[282820]: CRITICAL: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[282820]: WARNING: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [t_fwd.c:1581]: t_send_branch(): sending request on branch 0 failed kamailio[282820]: ERROR: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) Once the queue is full, the system stops servicing new calls and freezes. kamailio -v version: kamailio 5.7.2 (x86_64/linux) 95629c flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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: 95629c OS: Debian 11, 64 cores, 125GB of DDR4 RAM; DB: MariaDB 10.5.15; udp_children: 30; tcp_children: 34; TLS: YES; async_workers: 64. Incoming calls are sent via push notifications( Federico Cabiddu method: https://www.voztovoice.org/sites/default/files/KamilioWorld2015%20-Federico…). NetBridging(for SIP and RTPEngine). ims_charging for billing (integration with our billing system using the Diameter protocol). As soon as the service starts responding with 503, it does not recover on its own, you have to restart. My server is powerful enough for the number of users of my service (what I mean is that there is no problem with resources). For example, 64 cores, 125GB of DDR4 RAM, 2TB of disk, of which 1TB SSD for the database, 10Gigabit channel, this is only a server for Kamailio for Rtpengines separate servers. Please help me figure out what causes a service failure!?

1 0

Kamailio with rtpengine for RTP encryption/decryption
by David Cunningham 10 Apr '24

10 Apr '24

Hello, We're using kamailio, with rtpengine for RTP encryption/decryption. It works fine but I would like to understand the integration better. How does rtpengine get the TLS certificates, and what crypto library does it use (openssl?). Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

3 9

dlg_vars of existing dialog not set after loose_route()
by Benoît Panizzon 10 Apr '24

10 Apr '24

Hi Gang I have a strange issue after upgrading to 5.7 I use dlg_vars to keep some information until the end of the call to write them in a CDR. One of those variables records the timestamp of the first invite to determine how long it took from the invite to the 'connection established' if at all. The CDR is written when the BYE is being processed by the dialog module. Strangely, since updating to 5.7 (used to work with 5.5 and 5.6 I think, but I could be mistaking as other changes were made to the config) my dialog variables are null when BYE is being processed. reading the docs, I learned that that dlg_var are only polulated after loose_route() has been called. So I added some logging statement to track if the BYE in question is processed by loose_route(). Yes, it is, loose_route() returns success on the BYE in question. But when I log the content of my dlg_var after loose_route() it is still null. Any idea what the cause could be? -- 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 4

Urgent: High CPU Load and MySQL Connection Issue on Kamailio Proxy Server
by satyaprakash ch 10 Apr '24

10 Apr '24

Hi, We're encountering a CPU load of approximately 90% on the Kamailio proxy server. Additionally, there's an error logged indicating a lost connection to MySQL. Could this indicate a problem with the configuration file or is it related to MySQL itself? Your guidance would be appreciated. MySQL version: Ver 14.14 Distrib 5.5.47 Kamailio version: 5.5.5 Error in logs: /usr/local/sbin/kamailio[24933]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013)

2 1

Newbie looking for advice
by Wayne Wenthin 09 Apr '24

09 Apr '24

Hi All, I'm new to Kamailio but have been running dozens of Freepbx systems for the last 10 years. We are an Education Service District and service K-12 schools. Many of these we provide dial tone to, the rest we have connections to to provide for toll avoidance and backup outbound calling when the LEC fails. Our main focus is to save schools money that can be used for education instead of infrastructure. Right now I have a single freepbx machine that is using inbound routes to send calls to the proper school district. I would like to replace this with a Kamailio server instead but can't seem to get my mind around the best way to do this. I have 119 inbound routes in my current system with 22 trunks to the other districts. This may increase in the future as we try to migrate districts off of the expensive systems to a more open source way of calling. Currently calls between districts take the route out of the system just to come right back in to be routed to a different district. I would like to avoid that if possible. I've been trying to educate myself on the best practice in this type of structure but can't seem to figure out the best way to make this work. Most of the videos and websites assume a pretty intimate knowledge of Kamailio. That and the database structure doesn't make sense to me. I would like to keep each freepbx as its own registrar as we have lots of crossover with extensions and a few of the districts have their own staff that manage that aspect of the machines. . Any suggestions would be appreciated. Please ask questions if this just looks like I'm rambling. -- Wayne Wenthin Wide Area Network Administrator | CTA@MESD 503.257.1562 Phone/Fax www.cascadetech.org

2 1

Troubleshooting acc_radius module with 5.7
by santiago.troncoso＠quobis.com 09 Apr '24

09 Apr '24

Hi community, I am preparing a setup of Kamailio with radius accounting on a docker-compose environment. When configuring the acc_radius module I had to navigate through some old doc by Daniel and inspecting the code itself. The thing is that it seems working, but not. And, yes, I already trying without docker :) Context: two containers, host newtork, to keep it simple - one with kamailio + modules (acc, acc_radius with rad_cli) + radius client configuration - another with freeradius + radius client configuration. I ensure that all data correspond and radius config is fine and spin them up. Run some radius commands to test and let's try some accounting... Executing a failed call and triggering the account flag method, I receive an opaque -1 error from the module: ``` kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: Initial requests kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: FLAG FLT_ACC set kamailio-1 | 1(20) NOTICE: {1 22 INVITE yutncivdtflvzga@PT-96} acc [acc.c:286]: acc_log_request(): ACC: call missed: timestamp=1709746392;method=INVITE;from_tag=wvaqi;to_tag=1a86669fc745788378faa461e7993953-78550000;call_id=yutncivdtflvzga@PT-96;code=404;reason=Not Found;src_user=1000;src_domain=kamailio-local;src_ip=172.20.0.1;dst_ouser=1001;dst_user=1001;dst_domain=kamailio-local kamailio-1 | 1(20) ERROR: {1 22 INVITE yutncivdtflvzga@PT-96} acc_radius [acc_radius_mod.c:497]: acc_radius_send_request(): Radius accounting - ERROR - result: -1 ``` Log accounting is working ok, but radius accounting not. Time for troubleshooting. I tried to enable client debugging, but the module does not respond. So I modify the source directly... tested with `radclient` , `radtest`, `radsniff`... all seems to work when trying manually... I also used an internal radcli fuction `rc_check()` and it works. Meanwhile, I was unable to make it work with the module and the radcli library. So another log output using internal functions to debug the configuration: ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: nas-identifier, Valor: radius-wac kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: authserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: acctserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: servers, Valor: /usr/local/etc/radcli/servers - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: dictionary, Valor: /usr/local/etc/radcli/dictionary - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:300]: check_cfg(): Clave: default_realm, Valor: -- not set -- kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_timeout, Valor: 10 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_retries, Valor: 3 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: bindaddr, Valor: * kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: clientdebug, Valor: 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:369]: init_acc_rad(): RC_ACCT_INIT: rc_test_config looks good kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:375]: init_acc_rad(): RC_ACCT_INIT: Config applyed ok ``` The configuration seems ok, accepted by the library and seems to be applied properly, but when trying to parse its values... all seems like rubish... ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:401]: init_acc_rad(): RC_ACCT_INIT: Name: P�y:^ kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:402]: init_acc_rad(): RC_ACCT_INIT: Max 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:403]: init_acc_rad(): RC_ACCT_INIT: Port: 125489528 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:404]: init_acc_rad(): RC_ACCT_INIT: Secret: kamailio-1 | 0(18) ERROR: acc_radius [acc_radius_mod.c:410]: init_acc_rad(): RC_CLIENTDEBUG: No clientdebug specified ``` I'm using `libradcli4 1.2.11-1+b2` for running on the container and `libradcli-dev 1.2.11-1build1` for compiling. Open to hear ideas, somebody using it actively on latest 5.7 release? Some updated documentation flying around? I was thinking to open a GitHub issue to handle the documentation issue. Thanks in advance!

1 1

Updating $fn malforms packet.
by James Lipski 09 Apr '24

09 Apr '24

Hello, I have remote users registering/ routing calls through my server. When calls do pass through the server, the server updates the from name, to uri and from uri via updating the pseudovariables $fn, $tu, and $fu respectively. While this is has been working, I do occasionally see calls failing to process as the remote gateway fails the call with a "400 Bad From header". Inspecting the A-leg INVITE and the B-leg INVITE after updating the pseudovariables, I see the following: A-leg INVITE: INVITE sip:15551112222@example.com SIP/2.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;rport;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 70 From: sip:12223334444@example.com;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@example.com Contact: <sip:12223334444@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="12223334444", realm="example.com", nonce="ZhUw02YVL6cdi51KMymTxGpzCZCZhd8G", [uri="sip:15551112222@example.com](mailto:uri=)", response="3a25b4ea0b1154e31caee1184a92f12b" Content-Type: application/sdp Content-Length: 627 v=0 o=- 3921660487 3921660487 IN IP4 79..xx.xx.xx s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 79..xx.xx.xx b=TIAS:64000 a=rtcp:4007 IN IP4 79..xx.xx.xx a=sendrecv a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=ssrc:63772004 cname:3ff849bd611036a7 ---------------------------------- B-leg INVITE: INVITE sip:15551112222@togateway.com SIP/2.0 Record-Route: <sip:10.64.54.207;lr;ftag=dff3c585a6154f23898e1121918f1752;did=e21.65c1> Via: SIP/2.0/UDP 10.64.54.207:5060;branch=z9hG4bK919d.5ca83e030c0879093169546cd80591a6.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;received=79..xx.xx.xx;rport=61379;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 15 From: sip:12125557777@example.com"Updated Name" ;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@togateway.com Contact: <sip:12125557777@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 609 v=0 o=- 3921660487 3921660487 IN IP4 10.64.54.207 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 18400 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 10.64.54.207 b=TIAS:64000 a=ssrc:63772004 cname:3ff849bd611036a7 a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=sendrecv a=rtcp:18401 Note that I've updated the following on the B-leg: $fn = "Updated Name" // originally nothing $fu = sip:12125557777@example.com // originally sip:12223334444@example.com $tu = sip:15551112222@togateway.com // originally sip:15551112222@example.com While the from uri, and to uri do get updated, the from name is injected in the wrong place of the from header. This only seems to happen when the A-leg is missing the "<" and ">" characters enclosing the from headers. Should I not be updating the pseudo variables in this manner to update the from name, to uri and from uri; is there another way to update these attributes of the header without malforming it? Thank you.

2 1

making HTTP requests in stateless redirects
by Sergio Charrua 08 Apr '24

08 Apr '24

Hi all! For testing purposes, while I am waiting for the ST/SH REST API to be available from other teams, I developed a small python REST API that returns a mockup of a JSON object with some of the required values. The kamailio script, completely stateless as I do not need to keep track of the sessions, iterates through one of the JSON branches and adds the destination URLs to the Contact header and replies a SIP 300 Multiple Choices. This works: the UAC does receive the SIP response with the Contacts , but it also receives a SIP 500 error message right after, and I can't figure out why. And i bet this is simple to solve.... Any clue? I'm aware I'm not processing ACK messages, but the sl_send_reply 403 should do the trick right? ... at least while testing.... Also, as this is a stateless script, is there another way of using async http or making http requests to a REST API without having to use TM Module? Thanks in advance. The kamailio script is has follows: #!KAMAILIO > /* add API http timeout */ > #!define HTTP_API_TIMEOUT 5000 > #!define HTTP_API_ROUTING_ENDPOINT "http://some_python_rest_api/get_route" > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=4 > log_stderror=no > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=2 > /* set paths to location of modules */ > loadmodule "tm.so" > loadmodule "jsonrpcs.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "sl.so" > loadmodule "db_mysql.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "ipops.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "uac.so" > loadmodule "counters.so" > loadmodule "http_async_client.so" > loadmodule "jansson.so" > loadmodule "usrloc.so" > /* listen addresses */ > listen=udp:10.20.0.2:5060 > listen=udp:10.20.0.2:5062 > advertised_address="10.20.0.2"; > # ----- http_async_client params ----- > modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) > modparam("http_async_client", "connection_timeout", 2000) > request_route { > #route(HANDLE_DMQ); > route(HANDLE_OPTIONS); > if (is_method("INVITE")) > { > xlog("L_INFO","MAIN - calling TO_CARRIER"); > route(TO_CARRIER); > exit; > } > else{ > sl_send_reply("401","UNAUTHORIZED"); > } > } > route[TO_CARRIER]{ > xlog("L_INFO","TO_CARRIER - calling RELAY_API"); > route(RELAY_API); #Route relay > xlog("L_INFO","TO_CARRIER - return"); > return; > } > # Relay request using the API (response) > route[RELAY_API_RESPONSE] { > xlog("L_INFO","RELAY_API_RESPONSE - got response from REST API"); > if ($http_ok==1 && $http_rs==200) > { > xlog("L_INFO","RELAY_API_RESPONSE - HTTP RESPONSE: > $http_rb\n"); > if (jansson_get("json", $http_rb, "$var(json)")) { > xlog("L_INFO","RELAY_API_RESPONSE - JSON = > $var(json)"); > $var(count) = 0; > jansson_array_size("routes", $var(json), > "$var(size)"); > xlog("L_INFO","RELAY_API_RESPONSE - > jansson_array_size"); > >> >> while ( $var(count) < $var(size) ){ > > >> jansson_get("routes[$var(count)].headers.to.uri", $var(rtjson), "$var(v)"); > > xlog("L_INFO","JSON - routes[$var(count)] >> - $var(v)"); > > #$(avp(mycontacts)[$var(count)]) = >> $avp(mycontacts) + $var(v) + "\r\n"; > > $avp(mycontacts) = $avp(mycontacts) + >> $var(v) + ";"; > > > *append_to_reply("Contact: <" + $var(v) >> +">"+ "\r\n"); /* ---- IS THERE A BETTER ?? ..... */* > > > $var(count) = $var(count) + 1; > > } > > > xlog("L_INFO","RELAY_API_RESPONSE - RELAY"); > xlog("L_INFO","MAIN - calling REPLY_302"); > route(REPLY_302); > return; > } > } > send_reply(500, "API Not Available - http response = $http_rs > $http_ok"); > exit; > } > route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $si:$sp from_number $fU > to_number $ru"); > $http_req(all) = $null; > $http_req(suspend) = 1; > $http_req(timeout) = HTTP_API_TIMEOUT; > $http_req(method) = "POST"; > $http_req(hdr) = "Content-Type: application/json"; > jansson_set("string","from_ip",$si, "$var(http_routing_query)"); > jansson_set("string","from_port",$sp, "$var(http_routing_query)"); > jansson_set("string","from_number",$fU, > "$var(http_routing_query)"); > jansson_set("string","to_number",$ru , "$var(http_routing_query)"); > > xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: > $var(http_routing_query)\n"); > $http_req(body) = $var(http_routing_query); > t_newtran(); > http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); > xlog("L_INFO","RELAY_API - Waiting for Response"); > } > route[REPLY_302] { > # Sends a 300 Multiple Choices back to the proxy that requested the > routing lookup > xlog("L_INFO","REPLY_302 - send reply"); > sl_send_reply("300", "Multiple Choices"); > xlog("L_INFO","REPLY_302 - exit"); > exit; > } > route[HANDLE_OPTIONS]{ > if(is_method("OPTIONS")) > { > sl_send_reply(200, "OK"); > exit; > } > if ($fU=="ping") > { > sl_send_reply("200","OK"); > exit; > } > } SNGrep output: 1 - INVITE from UAC to Kamailio INVITE sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > User-Agent: SIPp > Date: Mon, 08 Apr 2024 15:09:51 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 270 > v=0 > o=root 2021555890 2021555890 IN IP4 10.20.0.1 > s=SIPp > c=IN IP4 10.20.0.1 > t=0 0 > m=audio 18422 RTP/AVP 8 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv 2 - Reply from Kamailio to UAC SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 3 - SIP 300 Multiple Choices from Kamailio to UAC SIP/2.0 300 Multiple Choices > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Contact: <sip:sip:918228990@10.20.0.2@sip.domain.io> > Contact: <sip:+351sip:918228990@10.20.0.2@10.20.0.3> > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 4 - SIP 500 from Kamailio to UAC SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM) > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=4eb2322b7d2b68e6fc3168f503344c21-32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 5 - ACK from UAC to Kamailio ACK sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 ACK > User-Agent: SIPp > Content-Length: 0 and continues with ACK and repeating SIP 500 above.... *Sérgio Charrua* .

4 6

Kamailio Admin Book email
by sadik.oualla.mohamed＠gmail.com 08 Apr '24

08 Apr '24

Hello, Last week, I submitted a request on the Asipto website to purchase the Kamailio Admin Book and received an automatic confirmation email. However, I have not received any further instructions on how to complete the purchase. I replied to the automatic email to follow up, but still, there has been no response. I need to get the book as soon as possible. Please note, I am writing from a different email address than the one I used to make the request on Asipto's site. If there are any issues with their contact email, or if further verification is needed, please let me know how to proceed securely without sharing personal details publicly on this list. Regards, Mohamed.

2 2

customizing the ds_ping_interval dispatcher parameter for destination
by Giuseppe Pandolfi 08 Apr '24

08 Apr '24

Hi, I have one question about dispatcher module in kamailio V.5.3.1. Why the ds_ping_interval parameter is only global value? Is it possible introduce the customization value for single destination? Customization from destination is already possible for ds_ping_from parameter using special attributes in destination list file or db table, is it possible introduce the same management for ds_ping_interval? Thanks, Giuseppe ________________________________ NOTA BENE: Le informazioni contenute nel presente messaggio, compresi gli eventuali allegati, sono strettamente confidenziali ed indirizzate unicamente al destinatario indicato. Qualora non foste il destinatario Vi invitiamo a cancellare il messaggio, ed ogni eventuale documento allegato, non divulgare il contenuto a terze persone ed a volerci avvisare a mezzo posta elettronica. Grazie. A TLC S.r.l. tratta i vostri dati personali nel rispetto della normativa vigente in materia di Privacy (D.Lgs.196/2003 modificato dal D.Lgs.101/2018 e GDPR Regolamento UE n.679/2016). Per consultare l’Informativa completa la invitiamo a visitare il nostro sito web “www.aethra.com<http://www.aethra.com/>” PLEASE NOTE: This message and any attachments are confidential and may contain privileged information. If you are not the intended recipient, you are kindly requested to cancel it, not to disclose the contents to any other person and to notify the sender immediately by return e-mail. Thanks. A TLC S.r.l. collects your personal data in compliance with applicable data protection laws (D.Lgs. 196/2003 amended by Legislative Decree 101/018 and GDPR EU Regulation n. 679/2016). Please visit our website ”www.aethra.com<http://www.aethra.com>” to consult the complete Policy.

1 0

RFC: uptime metric for xhttp_prom module
by Ovidiu Sas 05 Apr '24

05 Apr '24

Hello all, The xhttp_module can export all stats under "stats.get_statistics" RPC command. I was thinking of adding an optional "uptime" stat that will return the kamailio server uptime (like the "core.uptime" RPC command). This will make it easier to add an uptime panel in grafana. The new stat would be controlled by a module parameter (by default disabled). Something like: modparam("xhttp_prom", "xhttp_prom_uptime_stat" 1) And this will generate something like: kam_uptime 671 1712249054631 This can be already implemented in the script using the "prom_gauge_set()" and retrieving the uptime via "jsonrpc_exec()", but it would be nicer and faster to implement it in the module itself. Comments? -ovidiu

3 6

Kamailio Visions 2030 for Appalachia
by Alex Balashov 05 Apr '24

05 Apr '24

For immediate release: HAZARD, Kentucky (1 April 2024)--Evariste Systems LLC, Georgia-based veteran brokers of international charity partnerships, are pleased to announce the conclusion of lengthy trilateral negotiations among Taylor Swift, the Public Investment Fund (the sovereign wealth fund of the Kingdom of Saudi Arabia), and the Kamailio open-source community to bring Kamailio to underserved areas of the US Appalachia region. In a concert in Lexington, Kentucky last Sunday evening, and in her signature style of mystical "reveals" through equivocal song lyrics, Swift auspiciously inaugurated to her Swifties, as die-hard fans of the pop star are known, the beginning of Kamailio Visions 2030. (Editorial note: Swift's agent has threatened copyright litigation in the event that our press agency were to reproduce her meaning-rich lyrics directly, driven by the contention that fair use considerations do not apply to lyrics so "deep" as Swift's. At press time, the matter is being reviewed by our counsel.) Alex Balashov, principal of Evariste, praised Kamailio Visions 2030 as a continuation of Swift's long-standing commitment to forward thinking and innovation: "Taylor has been the driving force behind the use of sustainable, plant-based materials in her 'merch. Today, aided by the internationalisation of Saudi capital, she is helping to bring that same kind of ethical prosperity to Eastern Kentucky, West Virginia and the wider Appalachian coal country." Continued Balashov: "Between Alan Maimon and J.D. Vance, and the revelations of the Purdue Pharma litigation, it's fair to say that the plight of Appalachia, with its high structural unemployment, declining economic base, and the ravages of the opioid epidemic and deaths of despair, has been exposed to a broader audience than ever before. And with that knowledge comes responsibility, because we live with that knowledge now, and we can no longer turn a blind eye. The SIP and VoIP industries have always been a force of conscientious capitalism, and Kamailio a beacon of rectitude in a jaded, amoral, cynical world. A world in which Appalachian children are denied the SIP proxy is not a just world, and we cannot any longer live in such a world." Fred Posner, a deputy Evariste negotiator who was said to have thrived in the pressure of high-stakes negotiations with PIF governor Yasir Al-Rumayyan over the exact implementational modalities of the deal, reviewed the high notes of the landmark economic development accord: "It was a good start to defund the police. But with the unique social challenges of Appalachia, what every struggling family in the ruins of coal country really needs is access to clean, ethically sourced, fair-trade SIP proxies. That is why I am uplifted to say that we will be dropping, through a coordinated, large-scale airlift operation, a Raspberry Pi with Kamailio pre-installed, to every household with an income at or below 250% of the Federal Poverty Level (FPL) in the Appalachian region. The Pi chassis will be dye-free, BPA-free, and will be held together with a plant-based, biodegradable, non-toxic glue that does not produce intoxicating effects when sniffed. By 2030, every child alive in Appalachia today will have had several years to benefit from the power of a miniaturised SIP proxy appliance, and that is why we are calling it Kamailio Visions 2030." When pressed by us, Balashov noted that the prosocial objectives and economic development potential of Kamailio Visions 2030 has drawn scepticism from some quarters. "These are those people who think that Rasperry Pi devices preloaded with Kamailio are not what Appalachia most needs today. Those people would be wrong. As any metaphysicist can tell you, causes do not necessarily resemble their effects." "Taylor and her people know that Kamailio Visions 2030 draws on a long history of highly effective technological and app-based interventions in economically underprivileged areas, such as Nicholas Negroponte's One Laptop per Child (OLPC) scheme, and the famous carpet-bombing of Eritrea with second-generation iPads by Selena Gomez. All of this had tremendous results." "Salutary results", added a representative of the PIF. Balashov pointed us to the foundational problem statement of Kamailio Visions 2030, which cited the concept of "Technik", as developed in "Hybrid Reality", a technocratic manifesto (in e-book form) by much-loved TED visionaries Parag and Ayesha Khanna. There, the authors declared: "Good Technik requires a combination of the attributes that deliver high human development, economic growth, political inclusiveness, and technology preparedness.” "In short", said Balashov, "Taylor, with some help from the efficient deployment of Saudi capital, will bring good technik to Appalachia. The tide of technik lifts all ships." -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

4 3

low performance with no apparent reason
by Sergio Charrua 05 Apr '24

05 Apr '24

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216 CPU @ 2.10GHz and, 2 NICs and 4Gb RAM and MariaDB 10.6 as DB Backend., all running on a HP G380 host with a gazillion CPUs and a googol disk space! I currently have 3 scripts: - script #1 stateful with RTJson and simulating requests to routing engine and accounting - script #2 stateful but with just a simple routing to UAS, no rules, no DB, - script #3 stateless with a forward to UAS With script #3 I can go up to 2000CPS without issues with CPU at 37%! Above that value, I get retransmissions everywhere. On both scripts #1 and #2, the limit is 330CPS max after which I get a lot of retransmissions, while CPU/Core usage on Kamailio server stays below 10%. So I do not expect this to be a CPU issue. I could not understand why such (low) results, so I followed this article found at https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c <https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c> and created exact same scenarios, with kamailio script and SIPP templates available on the article, hoping for better results. But I get the same results: between 300 and 330CPS which is far, very far from the 7000CPS found in the article! I understand that I'm using VMs and probably the tests made for the article, which is pretty old already, were made on physical servers. Still, I would not expect 95% of lower performance! Any clue what could be the issue? I suspect NICs, but.... Any tips anyone could share? Thanks in advance! *Sérgio Charrua*

7 31

Statistics for all generated
by Alfredo Román Domínguez 05 Apr '24

05 Apr '24

I am using Kamailio versions 5.5.7 and 5.8.0 as a stateful proxy (with tm module) and I want to have metrics for all the SIP messages received and sent. I am using the statistics module for counting messages. I have in configuration file the following routes to capture events and count them: onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response, tm:local-request, sl:filtered-ack and network:msg) I have found the following problems that I need some hints how to solve them: 1) When a response is received to a request, I can count the response received in the reply_route but I don't find a way that works to count the reply being forwarded to the other side. onsend_route works fine for requests being forwarded. Is there any event_route or any trick that can be used to count the reply being forwarded similar to the onsend_route for requests? 2) When Kamailio receives a CANCEL it generates and sends automatically the 200 OK for the CANCEL and a 487 for the INVITE which started that dialog which is being cancelled, but I don't find a way that works to count those two responses Kamailio generates. 3) After having cancelled a dialog on one direction due to receipt of a CANCEL, kamailio generates local requests on the other direction, a CANCEL and an ACK also when it receives a 487 for the INVITE being cancelled. I don't find the way to count those. onsend_route does not work for locally generated CANCEL and ACK requests and the event_route[tm:local-request] only works for requests created outside of the tm module but not those created by the tm module itself (the CANCEL and the ACKs) since the tm module is not using its own function t_uac_prepare to create those requests (that function is the one that calls the event_route). Is there any statistic generated by the tm module or elsewhere or any other trick so I can count those requests? 4) When Kamailio generates retransmissions how can I count those? onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response) doesn't work for retransmissions. Are there statistics generated somewhere in the tm module for retransmissions? Is there any other way to count those. -- El contenido de este mensaje puede contener información confidencial sometida a secreto profesional. Si usted ha recibido este correo por error, no está permitida su distribución. Por favor elimine cualquier copia, archivo adjunto e infórmenos por esta vía. Los datos personales contenidos en este correo y que han sido facilitados por usted o provienen de fuentes de acceso público serán utilizados por Future Space, S.A., para gestionar nuestra relación contractual o pre-contractual, resolver su solicitud y mantenerle informado sobre los productos o servicios solicitados o similares. Se conservarán mientras exista un interés mutuo para ello, no se oponga al tratamiento o solicite su supresión. Los datos no serán comunicados a terceros, salvo obligación legal. Puede ejercitar sus derechos de acceso, rectificación, supresión, limitación del tratamiento, portabilidad de los datos y oposición mediante escrito dirigido a Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid o mediante correo electrónico a dpo(a)futurespace.com <mailto:dpo@futurespace.com>. Si consideras que se han infringido tus derechos puedes presentar una reclamación ante la www.agpd.es <http://www.agpd.es>. The contents of this electronic mail may be confidential or privileged. If you have received this message by mistake, please do not send it to anyone. We ask you to delete any existing copies, attached files and to notify us accordingly. The personal data provided or obtained from publicly accessible resources will be processed by Future Space, S.A. in order to manage our contractual or pre-contractual relationship, solve your request and keep you informed about activities or products requested by you or similar. Personal data will be stored while a mutual interest exist or either erasure or objection to process them is requested. Data will not be disclosed to third parties unless there is a legal obligation to do so. You have the rights to access, rectify, erasure, restriction of processing, data portability and oppose any processing of your personal data held by Future Space, S.A. To make effective use of your rights, please contact us at Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid or by email at dpo(a)futurespace.com <mailto:dpo@futurespace.com>. If you consider processing of your personal data infringes data protection regulations you can file a claim with www.agpd.es <http://www.agpd.es>

1 0

Kamailio v5.8.1 Released
by Daniel-Constantin Mierla 03 Apr '24

03 Apr '24

Hello, Kamailio SIP Server v5.8.1 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.0. 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 v5.8.1. For more details about version 5.8.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/04/kamailio-v5-8-1-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, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Releasing v5.8.1
by Daniel-Constantin Mierla 03 Apr '24

03 Apr '24

Hello, with many countries having public holidays around Catholic Easter, I am considering to release Kamailio v5.8.1 (out of branch 5.8) on Wednesday, April 3, 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 Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 1

start_recording' or 'record-call on' is not invoking
by pawan srivastava 03 Apr '24

03 Apr '24

Hi I am testing rtpengine recording with kamailio in lab. I have 1 endpoint registered to Kamailio. My kamailio and rtpengine are running on same box. I have another kamailio2 instance with 2nd endpoint to make a call to 1st endpoint. I have observed that Response dump for 'offer'/'answer' but further start recording didn't invoke. root@ip-kamailio:/etc/kamailio# rtpengine-ctl list sessions all callid: J0WtgYIKd0Kwj1iL6BiFOg.. | deletionmark: no | created: 1712120753 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: uH_ozKj18Nafshsm_sTElQ.. | deletionmark: no | created: 1712120730 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: VYIw791dJska8PiIFspy8w.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:54804 | redis_keyspace:0 | foreign:no callid: nP89gWwjhsg_TSZrqu16nQ.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:56207 | redis_keyspace:0 | foreign:no root@ip-kamailio:/etc/kamailio# I am new to kamailio and missing configuration. Suggest me the correct configuration to invoke recording. attached config and logs Regards, Pawan

1 0

How to relay SIP messages to specific registered user in Kamailio?
by o.atef＠fiberme.com 02 Apr '24

02 Apr '24

Hello guys, I'm recently new player in Kamailio SIP server and want to use it as SIP proxy server to Asterisk PBX server. first of all, I've successfully registered my Asterisk to Kamailio as user "test" to get its "location" because my Asterisk is behind NAT. Simply now I want to relay SIP messages to this user's location to receive them all in Asterisk. After I searched, I found this function rewritehost() but it needs to be a host or IP address, but in my case I want some way or function to relay to user "test" and then it well get the user location from Kamailio datebase automatically? I hope it's clear XD I tried to find a way to do MSQL query in kamalio.cfg to get the user contact from "location" table and then relay to the result but it did not work. I guess that I did it in wrong way but finally the Kamailio tutorials are so rare on google. Thanks in advance, Omar

5 10

Can we discuss the internals of Kamailio's routing logic and how to customize it for specific needs?
by Inextrix Technologies Pvt Ltd 28 Mar '24

28 Mar '24

This question digs into the heart of Kamailio's functionality. It asks about: Routing Logic: How Kamailio decides where to send SIP messages (calls, registrations, etc.) based on various factors like caller ID, destination number, or custom rules. Customization: How you can modify this built-in logic to fit your specific needs. This could involve using scripting languages, manipulating SIP messages, or defining custom routing rules. The discussion would be for developers who want to unlock Kamailio's full potential and tailor it for their unique VoIP deployments. Visit us for More Info : https://inextrix.com/services/kamailio-development/

2 1

Callee details not visible in dialog
by egemen ulus 28 Mar '24

28 Mar '24

Hi, I am wondering why callee info is not visible in dialog details: Here is the case: client-1 > kamailio > two asterisks > kamailio > client-2 registration is on asterisk. dialog module is enabled and I want to see the asterisk IP that is determined by dispatcher module. I have tried to put dlg_manage function even after dispatch function, but did not help. I have also tried to update to header before dlg_manage function, it didn't help either. Any way to have asterisk IP in dialog details. It shows only this: { h_entry: 483 h_id: 8644 ref: 1 call-id: v489LdfOFD5mnlHupQHOvw.. from_uri: sip:1515@12.12.12.12;transport=UDP to_uri: sip:2222@12.12.12.12 state: 5 start_ts: 0 init_ts: xxxx end_ts: xxxx duration: xxxx timeout: 0 lifetime: 3600 dflags: 512 sflags: 0 iflags: 0 caller: { tag: d0283007 contact: sip:1515@1.1.1.1:64966;transport=UDP cseq: 1 route_set: socket: udp:172.22.10.10:5060 } callee: { tag: <null string> contact: <null string> cseq: <null string> route_set: <null string> socket: <null string> } profiles: { } variables: { } } Regards Get Outlook for Android<https://aka.ms/AAb9ysg>

1 0

[Presence] Updating route information for SIP NOTIFY when re-SUBSCRIBE
by Denys Pozniak 26 Mar '24

26 Mar '24

Hello! In my architecture, SIP SUBSCRIBE messages can reach the Kamailio Presence server in several ways. And I noticed that re-SUBSCRIBEs messages do not update the record_route field in the active_watchers table (MySQL), so subsequent SIP NOTIFY messages do not inherit it and have the routes set of the initial SIP SUBSCRIBE message. Is there any way to change this behavior? modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) # kamailio -v version: kamailio 5.1.2 (x86_64/linux) -- BR, Denys Pozniak

3 4

RFC: move app_lua_sr to kamailio-archive
by Victor Seva 26 Mar '24

26 Mar '24

Hi, I would like to move app_lua_sr to kamailio-archive but I would like to hear if anyone out there has a solid reason not to do it. Since we have app_lua as KEMI for a long time I think is time to retire the old remains of app_lua Cheers -- ----------------------------------------------------------------- | ,''`. Victor Seva | | : :' : linuxmaniac(a)torreviejawireless.org | | `. `' PGP: 8F19 CADC D42A 42D4 5563 730C 51A0 9B18 CF5A 5068 | | `- Debian Developer | -----------------------------------------------------------------

1 0

Dialog and DMQ on restart
by Björn Klasen 25 Mar '24

25 Mar '24

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

4 9

RTP Engine with Kamailio
by Vishal Pai 22 Mar '24

22 Mar '24

Hello Everyone. I am looking for documentation to have multiple RTP Engine servers connected with kamailio using db url and balance the load accordingly. Thanks

4 3

Kamailio to aid in seamless data center migration?
by Brad 21 Mar '24

21 Mar '24

I need to move a few dozen FreePBXen with some commercial modules running in individual VMs to a new data center. I’m trying to work out a plan to move the PBXes to the new data center in a way that will be transparent to the endpoints, or at the very least with the absolute minimum of downtime. Some of the installations are rather old, and there’s a handful of peculiarities on each, so the typical FreePBX backup/restore process hasn’t gone smoothly, at least in our tests. My current train of thought is to put a clone the VMs in the new data center and use Kamailio to route the SIP traffic to the new servers/IPs. I’ve never used it before, so I may be barking up the wrong tree, but it /appears/ to do what I’m suggesting. If so, I’m thinking I can install Kamailio on each VM, point it to the local asterisk/FreePBX initially, and clone the VM. Then, after the new instance is up and running, point Kamailio on the original VM to the cloned VM's asterisk, after which I can make appropriate DNS changes. Another option would be to stand up a single Kamailio server and redirect SIP traffic destined for individual asterisk servers to it at the router. Endpoints are mostly Yealink, and I’m not sure if they’ll feel the need to restart when the registration/SIP server’s IP changes, but a quick bounce when not in use isn’t the end of the world. I’d very much like to avoid having to send an update to each phone manually, but I can script a SIP NOTIFY if required. Anything stupid, wrong, ignorant, or just smelly about this tactic? Or, for that matter, any other suggestions?

2 1

RFC5626 Keep-Alive with CRLF
by Ihor Olkhovskyi 21 Mar '24

21 Mar '24

Hello, A bit generic question, does Kamailio supports CRLF keepalive per https://www.rfc-editor.org/rfc/rfc5626#section-4.4.1 ? It's more for tracking TCP connection states, as seems tcp_keepalive_enable is not super reliable especially in mobile networks. -- Best regards, Ihor (Igor)

3 5

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users