sr-users July 2024

sr-users@lists.kamailio.org

42 participants
62 discussions

Incoming call with one one way audio and cutting off after thirty seconds

by palany

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

3 months, 2 weeks

http_async_query on carrierfailureroute changes rd back to carrieroute primary

by Maharaja Azhagiah

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 months, 2 weeks

Kamailio-IMS: After a abornamal Call, other call is not working (even after a new registration)

by deep.upreti＠gmail.com

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.

3 months, 2 weeks

How to log datetime in files

by 陈子妍

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 months, 3 weeks

Re: Possible Kamailio memmory leak - version 5.8.2

by Vanderlei Torres Batistela

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

3 months, 3 weeks

[TM] [ASYNC] : Freeing transaction after sending 302 response

by elhar.mohamed＠gmail.com

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.

3 months, 3 weeks

secfilter (and more modules) in kamailio kemi

by bazrafshan.m.h＠protonmail.com

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

3 months, 4 weeks

secfilter (and more modules) in kamailio kemi

by bazrafshan.m.h＠protonmail.com

3 months, 4 weeks

secfilter (and more modules) in kamailio kemi

by bazrafshan.m.h＠protonmail.com

3 months, 4 weeks

Job: System Engineer VoIP - Germany / HO

by Gerrit Jacobsen

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

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sr-users July 2024