- sr-users - kamailio.org

Enable switch redirect to other SIP server source
by bayu p 21 Feb '22

21 Feb '22

Dear Kamailio Expert, Could you help me switching redirect sip server to another sip server ? As described below architecure. I have some uac_redirect module to implement , but kamailio cannot perform running states. Pleas help to reditect sip server to another sip server. Thanks & Best regards,

2 1

kamailio success story
by Valentin Christoph 21 Feb '22

21 Feb '22

Hi all, We had to connect to an SBC, which was not compliant to RFC 3261. It was urgent, because management was waiting for first call. Within minutes, I could implement a workaround in kamailio routing configuration: ## CV experimental 2022-02-17: fake Supported HF if (is_present_hf("Require") && !is_present_hf("Supported")) { append_hf("Supported: $hdr(Require)\r\n", "Require"); msg_apply_changes(); } Thank you, kamailio! All the best Christoph

2 1

SIP server port service cannot be changed from default value (5060)
by bayu p 18 Feb '22

18 Feb '22

Hello Experts, Please help me to change port service from default 5060 port into other port due to security reason. Since I have change port in kamailio.cfg file, but services cannot fly normally that shown client unable login online (status not online). [image: image] <https://user-images.githubusercontent.com/76675592/154027407-24ab423b-302b-…> Besides, I need to run the kamailio services after install default kamailio, I have not found where procedure found in this installation process on github.com/kamailio. Could you please share if there another steps of compiling Kamailio source, install, deploy process, and running command for this specific installation process? Thanks Gb

4 9

Kamailio v5.5.4 Released
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, Kamailio SIP Server v5.5.4 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.4. For more details about version 5.5.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/02/kamailio-v5-5-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Releasing Kamailio v5.5.4
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, I am considering to release Kamailio v5.5.4 later this week, likely on Thursday, Feb 17, 2022. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

Info: sipexer v1.0.0 - sip cli tool
by Daniel-Constantin Mierla 16 Feb '22

16 Feb '22

Hello, I want to announce the availability of sipexer v1.0.0 - a sip cli tool that can facilitate testing and monitoring of SIP signalling systems. It tries to have a modern approach, with a flexible templating system, supporting both IPv4 and IPv6 with all the common transport layers, respectively UDP, TCP, TLS and WebSocket (for WebRTC). The project can be found at: * https://github.com/miconda/sipexer It is written in Go language for better portability, binaries for Linux, MacOS and Windows are made available for download in the release page: * https://github.com/miconda/sipexer/releases/tag/v1.0.0 Among its features: * send OPTIONS request (quick SIP ping to check if server is alive) * do registration and un-registration with customized expires value and contact URI * authentication with plain or HA1 passwords * set custom SIP headers * template system for building SIP requests * fields in the templates can be set via command line parameters or a JSON file * variables for setting field values (e.g., random number, data, time, environment variables, uuid, random string, …) * simulate SIP calls at signalling layer (INVITE-wait-BYE) * respond to requests coming during SIP calls (e.g., OPTIONS keepalives) * send instant messages with SIP MESSAGE requests * color output mode for easier troubleshooting * support for many transport layers: IPv4 and IPv6, UDP, TCP, TLS and WebSocket (for WebRTC) * send SIP requests of any type (e.g., INFO, SUBSCRIBE, NOTIFY, …) One usage example that could ease the testing of Kamailio is initiating registrations or simulating calls over WebSocket without the need of having a JavaScript soft phone application running in a web browser. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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allow_source_address_group in failure route
by Marrold 16 Feb '22

16 Feb '22

Hello, I'm using Kamailio as a proxy between several voice gateways and if the call fails for whatever reason (far end is busy etc) I am forwarding to an Asterisk box to play an announcement. Within the failure route I am using allow_source_address_group() to ascertain if the failure response was from a Voice Gateway or Asterisk itself, however it appears to return the group for the first branch when the reply is from the second branch to Asterisk. To handle the failure I am adding some headers, and setting a new destination indirectly with ds_select_dst before t_relay. I am using 5.5.3 on Debian 11, using KEMI / Python. Does anyone have any ideas? Thanks Matthew

2 1

Missing contact in 180 ringing causes wrong routing
by Xuo Guoto 15 Feb '22

15 Feb '22

Hello all, I have a kamailio ( kamailio 5.5.3 (x86_64/linux)) with asterisk (Asterisk 16.2.1~dfsg-2ubuntu1) with following architecture ┌─────────────┐ ┌─────────────┐ ┌─────────────────────┐ │ Asterisk ├───────┤ Kamailio │ 188.xxx.59.171 ───────── 103.xxx.xx.134 │ Linphone │ └─────────────┘ └─────────────┘ └─────────────────────┘ 10.10.1.13 10.10.1.9 In case the diagram is not displayed correctly its pasted at https://pastebin.com/b0UtE4Lj Registrations are handled by Kamailio and calls from linphone are routed to Asterisk. Similarly all calls originating in asterisk are routed over to Kamailio which in turn looks up AOR and routes the calls over to Linphone. Along with Linphone I also have a WebRTC endpoint at Kamailio. When calls are sent to WebRTC from asterisk, it gets correctly routed over to Asterisk, but if the client is Linphone the ACK of 200OK is not routed correctly to Linphone. I get this error in the logs: [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced tls:188.xxx.59.171:5061, to udp:188.xxx.59.171:5060) Digging into the difference between WebRTC and Linphone, I could see that in 1800 Ringing packet from WebRTC has a Contact headers while its missing in the packet from Linphone This is the 180 ringing from WebRTC: tag: rcv pid: 695615 process: 17 time: 1644860126.209155 date: Mon Feb 14 23:05:26 2022 proto: wss ipv4 srcip: 103.xxx.xx.134 srcport: 52508 dstip: 188.xxx.59.171 dstport: 443 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/WSS 188.xxx.59.171:443;branch=z9hG4bK730a.2cc7461b72f0c895e7cce5a0cedce4f7.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Content-Length: 0 While this is from Linphone tag: rcv pid: 695614 process: 16 time: 1644860097.970935 date: Mon Feb 14 23:04:57 2022 proto: tls ipv4 srcip: 103.xxx.xx.134 srcport: 25843 dstip: 188.xxx.59.171 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/TLS 188.xxx.59.171:5061;branch=z9hG4bKb5b1.d3231367383d07b1063098dcb35cc2b0.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPjd16f1379-1be7-448e-894a-c42d96f63146 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=cf7a7770-529d-4450-8bed-880382dce1dd To: <sip:user_jbjp@10.10.1.9>;tag=2ooYDia Call-ID: 8a025999-b096-4fe0-a623-9f6e6a69b8ae CSeq: 27838 INVITE User-Agent: Linphone Desktop/4.3.2 (Ubuntu 20.04.3 LTS, Qt 5.12.12) LinphoneCore/5.0.49 Supported: replaces, outbound, gruu Record-route: <sip:188.xxx.59.171:5061;transport=tls;r2=on;lr=on;did=3b9.b1b2> Record-route: <sip:10.10.1.9;r2=on;lr=on;did=3b9.b1b2> Content-Length: 0 When the 180 ringing is passed on to the Astrerisk it has the contact header modified with alias. tag: snd pid: 695615 process: 17 time: 1644860126.210189 date: Mon Feb 14 23:05:26 2022 proto: udp ipv4 srcip: 10.10.1.9 srcport: 5060 dstip: 10.10.1.13 dstport: 7000 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;alias=103.xxx.xx.134~52508~6;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Content-Length: 0 Now, I am not sure if this is the reason for routing the packets to wrong destination, but this is all I could figure out. Any clue to get this resolved will be much appreciated. My Kamailio config is pasted at https://pastebin.com/qyRS0usw with regards, X. Sent with ProtonMail Secure Email.

1 0

get kamailio sip processes id
by Patrick Karton 14 Feb '22

14 Feb '22

hello, as we are all aware kamailio create a lot of processes for many tasks. im using kamailio in kubernetes and i would like to monitor only sip processes with top command. i only need a way to get only kamailio sip processes id (udp receiver that handle sip request) . but i dont knwo how to get it. top -c -p id1,id2,id3,id4 Thanks

3 5

SEMS license with kamailio and rtpengine
by Patrick Karton 14 Feb '22

14 Feb '22

Hello im using kamailio and rtpengine to build a custom SBC. i was using freeswitch as my b2bua but i noticed a lot of issues when cps and simultaneous increased so i tried sems and i have better results. But i do not really understand sems license so i want to know if i can use it. i read from sems github "It is licensed under dual license terms, the GPL (v2+) and proprietary license. This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed. For a license to use SEMS under non-GPL terms, please contact FRAFOS GmbH at info(a)frafos.com" so my question is if i want to use sems as b2bua with kamailio and rtpengine and make profit of it do i need to purchase a license each time i want to sell it to a customer ? Thanks.

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Kamailio server behind NAT
by Tihomir Culjaga 13 Feb '22

13 Feb '22

Hi all I'm wondering what is best practice to configure kamailio for a setup shown below. Carriers only support interconnection over a private network so I need to terminate every separate network on kamailio machine (and on RTPEngine) . That will turn very quickly into an administration nightmare as we add more and more carriers. My plan is to terminate carrier private networks on a router and NAT the traffic to kamailio where it leads me to a kamailio behind NAT setup. Well, in theory set_advertised_address and set_advertised_port per message can do the trick... at least I hope. Also, some carriers require a SIP registration to activate SIP trunks. I do that using UAC module (remote registration) but here again I got a problem with contact address as you can set only one contact address (modparam("uac", "reg_contact_addr", "10.1.1.11:5060")). /--RTP 10000-19999-- RTPEngine1 -----RTP------+-- IPPBX1 carrier1 --- NAT ---| | | | | | carrier2 --- NAT ---+-----SIP 5060------ kamailio ----SIP---- Dispatcher | | | carrier3 --- NAT ---| | | \--RTP 20000-29999-- RTPEngine2 -----RTP------+-- IPPBX2 So bottom line, I'm wondering what would be the best practice for such a scenario ? That would spare me from hitting many pitfalls. Would really appreciate a hint. Regards, Tihomir.

3 3

Vo5G with open5gs
by Hornbuckle, Mark A 11 Feb '22

11 Feb '22

Hello, I have been trying to find documentation on Kamailio IMS and Vo5G support for open5gs. I wanted to ask you directly if Kamailio IMS does support Vo5G with open5gs? Thank you

1 0

resending same message from UE after receiving RP-ACK
by Eugene B 11 Feb '22

11 Feb '22

Hi, I trying to use smsops module as SMSC. When I sending message from UE I see re-sending same message after each RP-ACK(~10 attempt). What could be the problem? Thanks! pcap in attach. kamailio 5.3.2 -- *Yours faithfully,* *Eugene B.*

2 1

kamilio open5gs 5G SA compatible?
by Sandland, Seth P 11 Feb '22

11 Feb '22

Were looking to test a kamilio on Open5gs 5g stand alone ( non lte, non NSA) can you conform or provide documents or install instructions on how this is accomplished. I can not find much of any details if Kamilio support Open5Gs 5g SA and what features it does support on a 5g SA core. Seth Sandland Wireless Communications IT Engineer Pacific Northwest National Laboratory Tel: (509) 375-6842 Email: seth.sandland(a)pnnl.gov<mailto:seth.sandland@pnnl.gov> www.pnnl.gov<http://www.pnnl.gov/> [A close up of a logo Description automatically generated]

1 0

Publish XML and xpath
by Igor Olhovskiy 10 Feb '22

10 Feb '22

Hello, I'm trying to get state of endpoint from PUBLISH message in a most simple way using xmlops module Trying to make this $xml(body=>doc) = $rb; log("L_ERR", "$rU is now in state $xml(body=>xpath:/dialog-info/dialog/state)\n"); XML in PUBLISH itself <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:88881@dev.company"> <dialog id="v6qmEYf5yI" direction="initiator"> <state>confirmed</state> </dialog> </dialog-info> Actual text in log is ... <script>: 88881 is now in state Means it's empty. xmlops module is loaded with no parameters. Tried also string $xml(body=>xpath:/dialog-info/dialog/state/text()) -> same result What am I missing here? kamailio 5.4.6 -- Thanks in advance, Igor

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KEMI KSR.async Python3
by Ahmed Moubayed 10 Feb '22

10 Feb '22

Hi, I'm trying to use the async module from a python3 Kemi script but it conflicts with the keyword async. Any suggestions on how it can be used? Is there an alternative to KSR.async.route()? Thanks, Ahmed

1 0

CDR preserve original call direction
by Ahmed Fouad 10 Feb '22

10 Feb '22

Hi, I've enabled ACC and CDR logging and it's working fine for the logging but I noticed something wrong in the CDR table , the call source and destinations are depending on the BYE message not the initial invite dialog ACC table +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | id | method | callid | time | src_ip | src_domain | src_user | dst_ouser | dst_domain | +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | 216 | BYE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:57:51 | x.x.x.x | domain.com | DST_CLID | SRC_CLID | domain.com | | 215 | INVITE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:55:17 | x.x.x.x | domain.com | SRC_CLID | DST_CLID | dst_domain.com | The INVITE is the correct one with SRCIP x.x.x.x and domain.com but the BYE reversed the order since the the DST_CLID who sent the BYE so it's now logged as the source and the final CDR log looks like +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | cdr_id | src_username | src_domain | dst_username | dst_domain | call_start_time | call_end_time +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | 195 | DST_CLID | domain.com | SRC_CLID | domain.com | 2022-02-09 06:55:12 | 2022-02-09 06:57:51 | the CDR as shown is reversed regarding the source/dst username I tried to enable the direction ACC direction flag but didn't help and here is my current paramters modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "cdrs") Can you please advise how to configure ACC/CDR to log the CDR with the correct initial INVITE call flow regardless of the BYE direction ?

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REGISTER with http_async_client ?
by Cyril Ramière 10 Feb '22

10 Feb '22

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); } } ##### END SNIP Best regards!

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Kamailio doc request
by Mustafa Rahman 10 Feb '22

10 Feb '22

2 1

Failure route with t_relay()
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I am trying to make a re-transmission mechanism from the failure route, I've tried to use $dp and $du as a relay, but variables are not accepted in t_relay(), When using ip and port straight forward t_relay("144.0.0.0", "1111"), I am getting: [t_fwd.c:1752]: t_forward_nonack(): no branches for forwarding [tm.c:1753]: _w_t_relay_to(): t_forward_noack failed Is there a way to re-transmit the call to the same carrier again? Thanks in advance, and have a great day, Max Kemel

2 1

Failure route mechanism with counter
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I would like to use a failure route with a retry mechanism, which provides a MAX_NUM_OF_RETRIES (max number of times reaching the failure route), Which can be achieved by a simple global counter, from what I've seen, the Counters module doesn't provide a GET function in script, What is the easiest solution for this kind of behavior? Thanks a lot, Max Kemel

2 1

Timers firing too soon
by Alberto Izquierdo 04 Feb '22

04 Feb '22

Hi all, I'm using Kamailio timers to execute a series of calls with a certain delay. I have found out that from time to time, the timers are triggered too soon, sometimes they are triggered 2/3 seconds before they should. I have read in the documentation that it is normal to have a bit of delay in some of the timers, but it doesn't mention anything about this behavior. Is this normal? Thank you.

1 0

Debian kamailio.init vs. kamailio.service
by Juha Heinanen 04 Feb '22

04 Feb '22

By comparing pkg/kamailio/deb/bullseye dir kamailio.init and kamailio.service files, it appears to me that they are not equal alternatives. Is this correct interpretation? -- Juha

1 0

Homer7 and Kamailio 5.3
by Angelo Sipper 03 Feb '22

03 Feb '22

Hi to all, We have a Kamailio 5.3 that sends all traffic to a homer7 using siptrace module with parameters force_send_sock and hep_mode_on,1 and hep_version,3 and trace_mode,1. Everything works fine getting all traffic to Homer7 but, in case Homer7 is not reachable, Kamailio then no longer accepts traffic or registrations and it is like it crashes. When we revive Homer7 and is it again reachable accepting traffic, Kamailio then works again fine without any restart or actions. Any ideas why this may be happening? Is there anything we need to add to the config to avoid this behavior as having Homer7 now is a single point of failure for Kamailio? Thank you! Kind Regards, Angelo

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uac problem with 5.3.8
by Richard Robson 03 Feb '22

03 Feb '22

we have been running 5.3.4 with the uac_auth_mode() function ans it has been working we have upgraded to 5.3.8 and with the same script we are getting an error with the uac module saying it was unable to find the cmd auc_auth_mode I have tried this with uac_auth as well with the same result. it looks like the uac module is not loaded or is not loading but running a core.modules reports the uac module can anyone shed any light on this problem? is it a version pecific problem?

3 4

Updating $du for uac REGISTERs in event_route[tm:local-request]
by Raffaella Seuß 03 Feb '22

03 Feb '22

Hello all, I tried to update $du for REGISTERs sent by the uac module in event_route[tm:local-request]. (I want to overwrite the auth_proxy under some conditions.) However, the requests are still sent to the auth_proxy defined in uacreg. Am I missing something here, why doesn't it work? It's basically just: event_route[tm:local-request] { if (is_method("REGISTER")) { # ... $du = "sip:PROXY_IP:PROXY_PORT"; } } My Kamailio version is 5.5.2. With kind regards, Raffaella Seuß foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: raffaella.seuss(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

1 0

Siremis configuration
by Tim Mastarone 02 Feb '22

02 Feb '22

Hi, I am using Siremis v 5.3 with kamailio. My users can log into the Siremis site just fine but the field to enter their email address is not appearing. My sql queries seem to be working besides that. Does this feature need to be enabled? I would like to use email to confirm the user login with a confirmation code. (mysql Ver 15.1 Distrib 10.1.48-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2) Thank you, [cid:image001.png@01D81830.52C21EA0]<http://www.vikingelectronics.com/> TIM MASTARONE Associate Engineer 715-386-8861 x208 vikingelectronics.com<http://www.vikingelectronics.com/>

1 0

two SIP proxies
by Uwe Sigleur 31 Jan '22

31 Jan '22

2 1

tel2sip is giving unpredictable result
by Jyoti Bansal 31 Jan '22

31 Jan '22

hello, we are using kamailio 5.3 . while establishing a call the tel uri is getting converted into sip uri sometimes incorrectly . In the sip uri some extra characters are getting appended in the invite message. Can somebody please provide some guidance on this, attaching the pcap for reference(packet no. 8). Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

kamailio log dialog correlation
by marek 31 Jan '22

31 Jan '22

hi, is it possible create log of SIP dialog like https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging ? it looks like its possible like this xlog("L_INFO", "D$dlg(h_id) something to log \n"); but i must copy $dlg(h_id) in every xlog tried this modparam("xlog", "prefix", "D$dlg(h_id)") but it not works with variables any other tips/ideas? goal is to correlate logs for one dialog (call) thanks Marek

7 16

REGISTER with http_async_client?
by Cyril Ramière 31 Jan '22

31 Jan '22

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. It seems that my AUTH_REPLY route is called but considered as a "failure_route" 19(27) exec: *** cfgtrace:dbg_cfg_trace(): failure_route=[AUTH_REPLY] c=[/etc/kamailio/kamailio.cfg] l=665 a=16 n=if 17(23) NOTICE: <script>: [AUTH_REPLY] FAILURE ROUTE If I remove the http_async_client and try with route(AUTH_REPLY) then it is considered as request_route (but this is not a solution since I need to do this http request) Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); if(t_is_failure_route()) { xlog("L_NOTICE", "[AUTH_REPLY] FAILURE ROUTE \n\n"); } } } ##### END SNIP Best regards!

1 1

Use specific expires per User-Agent
by mayamatakeshi 31 Jan '22

31 Jan '22

Hi, I have a requirement to specify a REGISTER max_expires based on the User-Agent. I checked the docs and it seems there is no pseudo-variable or something like that would permit this. I would like to be able to do something like this: if($ua ~= "Some-UA") { $max_expires=100; } else { $max_expires=150; } save("location"); Is there some way to achieve this?

2 2

failed to find string from message body
by Juha Heinanen 29 Jan '22

29 Jan '22

I tried to check if message body contains a string using function call str_find("$rb", "string")) but it failed like this: Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1491]: pv_printf_mode(): no more space for spec value - printed:0 token:6880 buffer:4096 Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1503]: pv_printf_mode(): buffer overflow -- increase the buffer size... Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/sr_module.c:1698]: get_str_fparam(): Could not convert the PV-formated string to str Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: textops [textops.c:4457]: str_find_f(): cannot get first parameter Any suggestions on how to get it done properly? -- Juha

1 0

Question about siptrace destination/source port capture in HEPv3 mode
by Karsten Horsmann 29 Jan '22

29 Jan '22

Hello List, I observe an interesting behavior. We use siptrace as an active capture agent for QXIP HOMER/HEPIC installations. The destination port and the source port in the HEPv3 Header are not correct if you use transport tcp for example. My Question: is that intended or a bug and i should fill up an issue? On udp, it's filled with 5060 or whatever your listening directive says. I tested that with kamailio 5.5 and kamailio master. I attached an HEPv3 capture from two kamailios (master) speaking with each other. You need the https://github.com/sipcapture/hep-wireshark lua dissector to read the capture. 10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode) 192.168.50.4 (kamailio as registrar) Frame 1 shows an tcp transport register Request from 10.0.2.15 to 192.168.50.4 in HEPv3. The Source Port in the HEP3 Protocol is not correct (its an highport / 45419). Frame 4 shows the 401 answer and the Destination port is 0. Frame 7 also had this behavior. ----- params (nearly at the end): modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture agent id Limitation: 32-bit for HEPv3. modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need call siptrace flag or sip_trace() -- Mit freundlichen Grüßen / Kind Regards *Karsten Horsmann*

3 2

tel2sip is not working correctly
by Jyoti Bansal 29 Jan '22

29 Jan '22

Hello all, I am using kamailio code to establish calls with the real ue's. Our calls are getting established with an issue , whenever invite is send from A party to B party with teluri that tel uri is converted to sip uri. While converting to sip uri some extra bytes are getting added after user=phone and the call ends. Can someone help us on this issue of getting some extra bytes added in the sip uri? for example :- sip:405874112224411@ims.mnc123.mcc456.3gppnetwork.org;user=phone�� Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

1 0

Comunication between devices RTP/SRTP and viceversa
by Social Boh 28 Jan '22

28 Jan '22

I'm trying to configure Kamailio to permit communication between devices: some use TLS and SRTP some UDP and RTP with RTPEngine in the middle. A priori I don't know if the device support SRTP or RTP so in the routing I have to detect on the fly how to act. The problem is this: Call from RTP to SRTP device: 488 Not Acceptable here Call from SRTP to RTP device: 415 Unsupported Media Type Any hint? Thank you Regards -- --- I'm SoCIaL, MayBe

4 5

200 OK re transmission only on re-SUBSCRIBE
by Lewis Hutchinson 27 Jan '22

27 Jan '22

First of all I should start by saying this is my first post so go easy on me!! : ) and thank for everyone's efforts to share knowledge it has been invaluable for my kamailio learning. Build Details Version: kamailio 5.5.3 (x86_64/linux) Kamailio kemi python Scenario Relevant config def ksr_route_api_query(self, msg): KSR.log("info", "SUBSCRIBE API Query Starts") # Create new Transaction KSR.tm.t_newtran() async def main(): url = 'https://api.hidingtheurl.com/subscribe' payload = { "ruri": KSR.pv.get("$ru"), "from": KSR.pv.get("$fu"), "from_tag": KSR.pv.gete("$ft"), "to": KSR.pv.get("$tu"), "callid": KSR.pv.get("$ci"), "cseq": KSR.pv.get("$cs"), "contact": KSR.pv.gete("$ct"), "supported": KSR.pv.get("$hdr(Supported)"), "event": KSR.pv.get("$hdr(Event)"), "expires": KSR.pv.get("$hdr(Expires)"), "diversion": KSR.pv.get("$di"), "body": KSR.pv.get("$rb") } async with aiohttp.ClientSession() as session: async with session.post(url, json=payload) as response: if response.status == 0: KSR.xlog.xinfo(f"SUB Manager Timeout: {response.status}") KSR.tm.t_reply(408, "SUB Manager Timeout") elif response.status >= 500: KSR.xlog.xinfo(f"SUB Manager Down: {response.status}") KSR.tm.t_reply(500, "SUB Manager Down") elif response.status >= 400: KSR.xlog.xinfo(f"SUB Manager Error: {response.status}") KSR.tm.t_reply(400, "SUB Manager Error") else: if response.status == 200: KSR.xlog.xinfo(f"SUB Manager Success: {response.status}") KSR.tm.t_reply(200, "OK") KSR.log("info", "SUBSCRIBE API Query Ends") asyncio.run(main()) # API Query Ends Scenario Initial SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get receive 200OK response from API and send a 200OK on to Device SUBSCRIBE --------> <------------ 200 OK After Subscription Expiry re-SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get 200OK response from API and send a 200OK on to Device but also another 200OK a fraction of a second later... SUBSCRIBE --------> <------------ 200 OK <<<--------- 200 OK This happens for all following re-SUBSCRIBE's The 200 OK are identical, and it is a re transmission, but I can't work out why.. Relevant DEBUG is below Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: INFO: <script>: SUB Manager Success: 200 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_reply Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:287]: sr_apy_kemi_exec_func_ex(): params[2] for: t_reply are int-str: [200] [OK] Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (80.111.111.111, 80.111.111.111, 0) Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1763]: cleanup_uac_timers(): RETR/FR timers reset Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out - buf=0x7eff0cd50938: SIP/2.0 200 OK#015#012Via:... shmem=0x7eff09190ca8: SIP/2.0 200 OK#015#012Via: Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:648]: _reply_light(): finished Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_precheck_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_check_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7eff0cb18220: SIP/2.0 2..., shmem=0x7eff09190ca8: SIP/2.0 2 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:112]: sr_kemi_config_engine_python(): execution of route type 1 with no name returned 1 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/receive.c:514]: receive_msg(): request-route executed in: 38258 usec Everything works well as is but I want to clean up this issue or at least understand what is causing it. Lewis

2 6

Some problems with SIP TRACE
by Arnaud 27 Jan '22

27 Jan '22

Hi, I'm trying to use siptrace to getting copy of my SIP transactions, especially in TLS. Here's my configuration : *loadmodule "siptrace.so" modparam("siptrace", "trace_mode", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "duplicate_uri", "sip:MY_IP:5065") modparam("siptrace", "trace_local_ip", "MY_IP:5061") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_capture_id", 1) modparam("siptrace", "trace_flag",22); * And in my Routing Logic : *request_route {* *sip_trace();setflag(22);* *(...)* But I didn't receive anything on my listening port 5065 according my TCPDUMP. Status confirm me siptrace is working : *root@debian:/# kamcmd siptrace.status checkEnabled* I think I made a mistake in the configuration, but I don't know where. My second question is when I try to use sip_trace_mode("t"); I getting : failed to find command sip_trace_mode. Is it in an other module ? Many thanks for your help. Regards, Arnaud

2 1

Re: [SR-Users] sr-users Digest, Vol 200, Issue 25
by surendra tiwari 27 Jan '22

27 Jan '22

Hi Aditya, There are 2 options to achieve this: Option-1 use below param in auth_db module and while doing authentication change table name to your table user_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> domain_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> password_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> Option-2 create a new http service that will connect to kamailio via evapi and will be responsible for authentication and location service. On Wed, 26 Jan 2022 at 16:31, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Kamailio as a SIP registrar (Aditya Uppuluri) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Jan 2022 16:21:33 +0530 > From: Aditya Uppuluri <uppuluriaditya(a)gmail.com> > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Kamailio as a SIP registrar > Message-ID: > <CAKWpYb= > 9EABvpmbPhvCyt-roQaOkiJnZSh17vVu6Wk26PiBbeg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hello Team, > Thank you for the amazing product. I am a newbie to kamailio and currently > trying to use kamailio as a SIP Registrar (for a multidomain). Here is the > requirement > > 1. We already have a mysql database (name: VOIP) with a table named > VoipSubscribers which contains the userid, password and domain information > of all the users. > 2. We want the ability in kamailio to use the same VoipSubscribers table > for user registrations and also store the location information in the same > database (VOIP) > > May I know how to achieve this without altering the table schema? > Thank you in advance. > > Regards, > Aditya Uppuluri >

1 0

Kamailio as a SIP registrar
by Aditya Uppuluri 26 Jan '22

26 Jan '22

Hello Team, Thank you for the amazing product. I am a newbie to kamailio and currently trying to use kamailio as a SIP Registrar (for a multidomain). Here is the requirement 1. We already have a mysql database (name: VOIP) with a table named VoipSubscribers which contains the userid, password and domain information of all the users. 2. We want the ability in kamailio to use the same VoipSubscribers table for user registrations and also store the location information in the same database (VOIP) May I know how to achieve this without altering the table schema? Thank you in advance. Regards, Aditya Uppuluri

2 1

Append HF don't work in event_route
by Arnaud 24 Jan '22

24 Jan '22

Hello, I'm trying to connect kamailio to Microsoft Teams. Dispatch is correctly configured and sended. Following this tutorial : https://skalatan.de/en/blog/kamailio-sbc-teams, I added an Event_Route to add contact header to SIP OPTION from dispath to MS : event_route[tm:local-request] { xlog("*** tm:local-request *** "); if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { xlog("L_INFO", "Append new contact !\n"); append_hf("Contact: <sip:MY_SBC_NAME.COM:5061 ;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } When I check the log to see my sended frame, I don't see a contact header field : INFO: <script>: Append new contact ! INFO: <script>: Sent out tm request : OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS MY_SBC_NAME.COM:5061 ;branch=z9hG4bK4497.f3e5e754000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:MY_SBC_NAME.COM.fr>;tag=a76e006ace3802951ed860e08df964af-7971 CSeq: 10 OPTIONS Call-ID: 2e1278ea77796160-9837@MY_SBC_IP Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.2.1 (x86_64/linux)) According the documentation about textop, append_hf(txt[, hdr]) : This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE. So my question is append HF can be used in Event_route ? Is there any other way to add contact header ? Regards, Arnaud

1 0

g_hash_table_insert_internal: assertion 'hash_table != NULL' failed
by Muhammad Danish Moosa 23 Jan '22

23 Jan '22

Hi, I am running stress tests on kamalilio , at times i see this on screen (process:30959): GLib-CRITICAL **: 21:43:56.961: g_hash_table_insert_internal: assertion 'hash_table != NULL' failed version: kamailio 5.5.3 Kamailio does not crash and keeps accepting calls though. Is it anything alarming. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

1 0

dispatcher lb
by Gokan Atmaca 22 Jan '22

22 Jan '22

Hello I'm load balancing with Dispatcher. The call comes in but hangs up without picking up the phone. What could be the problem? The settings are as follows. dispatcher.list 1 sip:18.X.X.X.X:5060 0 0 maxload=20 1 sip:67.X.X.X.X:5060 0 0 maxload=20 # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 4.130000 STD: 0.398000 EST: 4.006000 MAX: 6 TIMEOUT: 0 } } DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 22.282000 STD: 2.345000 EST: 23.075000 MAX: 42 TIMEOUT: 0 } } } } } } -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

2 3

Effect of GRUU on the behavior of loose_route()
by Denys Pozniak 21 Jan '22

21 Jan '22

Hello! I ask for help in explaining the different behavior of loose_route() function in two similar cases (depending on GRUU parameter on the end device). The call path goes through a series of Proxies: Proxy Registrar (use Path) -> Proxy Edge -> PhonerLite (GRUU on/off). If the GRUU parameter is enabled on the PhonerLite, then the in-dialog SIP ACK (on 200 OK/SDP) message is routed on the Proxy Edge successfully (loose_route() deletes Route and handle_ruri_alias() sets $du). When GRUU is disabled - loose_route() for some reason replaces RURI (but lr=on ?) and sets $du. The configuration of the [WITHINDLG] route section is similar to the default script. IP: 1.1.1.1 is the Proxy Registrar IP: 2.2.2.2 is the Proxy Edge IP: 3.3.3.3 is the Phone (Phoner Lite) Successful case (GRUU enabled): 2022/01/21 20:08:20.745859 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:20.746952 2.2.2.2:5060 -> 3.3.3.3:5065 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK0a34.2b5076a5473cc42d883bf38969aa7dcf.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 3(11755) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: rr [loose.c:856]: after_loose(): Topmost route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa' is me 3(11755) DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header 3(11755) DEBUG: rr [loose.c:181]: find_next_route(): No next Route HF found 3(11755) DEBUG: rr [loose.c:886]: after_loose(): No next URI found 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: nathelper [nathelper.c:1202]: handle_ruri_alias(): setting dst_uri to <sip:3.3.3.3:5065> 3(11755) DEBUG: nathelper [nathelper.c:1224]: handle_ruri_alias(): rewriting r-uri to <sip:userid@2.2.2.2:5060 ;gr=009356C7-5A79-EC11-8468-538E88103B96> Unsuccessful case (GRUU disabled): 2022/01/21 20:08:47.851170 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:47.851963 2.2.2.2:5060 -> 2.2.2.2:5060 ACK sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK110e.af605388ece54d48387c1de53615d894.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 1(11753) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 1(11753) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 1(11753) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 1(11753) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 1(11753) DEBUG: rr [loose.c:676]: after_strict(): Next hop: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' is loose router 1(11753) DEBUG: rr [loose.c:726]: after_strict(): The last route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: rr [record.c:977]: add_rr_param(): rr_param_buf=<;did=3f8.6a83> 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: tm [t_lookup.c:1329]: t_newtran(): msg (0x7efe343f4998) id=11/11753 global id=10/11753 T start=0xffffffffffffffff 1(11753) DEBUG: tm [t_lookup.c:498]: t_lookup_request(): start searching: hash=57361, isACK=1 1(11753) DEBUG: tm [t_lookup.c:456]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0] 1(11753) DEBUG: tm [t_lookup.c:675]: t_lookup_request(): no transaction found 1(11753) DEBUG: tm [t_funcs.c:286]: t_relay_to(): forwarding ACK statelessly -- BR, Denys Pozniak

1 0

Experiences with Kamailio on Debian 11 Bullseye
by Daniel-Constantin Mierla 20 Jan '22

20 Jan '22

Hello, has anyone experienced issues with Kamailio running on Debian 11 Bullseye? It's now more than 2 months since Debian 11 release and I wonder if people started to migrate to it for production systems. So far I did only basic tests on it, although I have a Debian Testing that I used for compilation and simple tests during the past several months before Debian 11 release. If someone discovers issues or changes in behaviour with various libraries or applications/database servers, report them to the tracker to try to address them as soon as possible. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 08-11, 2021 (Europe Timezone) - Nov 22-25, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

8 14

Handling special characters - KEMI lua redis_ndb
by Muhammad Danish Moosa 20 Jan '22

20 Jan '22

Hi Guys, I am trying to do following code KSR.ndb_redis.redis_cmd("srvN", "SET var " .. testvar, "r"); it works fine but if testvar has special character like SPACE or % , it does not work. For example when testvar="asdsada%Dasda"; it starts considering %D as place holder and does not add key to redis (does not raise error as well) With Space I tried to use delimeters like "" or '' but none of them work. For now I have got it working through , search and replace but thats not a clean solution. Any suggestions? -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 1

Does cfgutils usleep block the process?
by mayamatakeshi 20 Jan '22

20 Jan '22

Hi, it seems execution of usleep eventually reaches this code: https://github.com/kamailio/kamailio/blob/master/src/core/ut.h#L508 so it seems it would block the process and so it might cause delay of packet processing if many processes do this at the same time. Can someone confirm this?

2 2

use $var or $avp in ds_list_exists & ds_select dispatcher function
by Patrick Karton 19 Jan '22

19 Jan '22

Hello im using kamailio 5.5.3 and im trying to use pseudo vars in ds_list_exists and ds_select function but it does not work. ds_list_exists return -2 with ds_list_exists("6") it works $var(dset)=6 or $avp(dset)=6 it does not work ds_list_exists("$var(dset)") --> return -2 ds_list_exists("$avp(dset)") --> return -2 Thanks.

1 0

Fwd: Possible memory leak on 5.5.x (new)?
by George Diamantopoulos 19 Jan '22

19 Jan '22

Forwarding my reply to the list, using gmail's reply button set Henning as the sole recipient :-\ ---------- Forwarded message --------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: Sat, 26 Jun 2021 at 02:25 Subject: Re: [SR-Users] Possible memory leak on 5.5.x (new)? To: Henning Westerholt <hw(a)skalatan.de> Hello Henning, Thanks for your reply. Here's what has come up after a few hours: shm55: https://pastebin.com/h9JCePmc shm54: https://pastebin.com/Nx5xEEnA It seems to me htable is the culprit? Are you seeing anything different? 54 has been running for 77020 seconds, 55 for 28521 (significantly less). I'm going to turn it off until we figure something out... BR, George On Fri, 25 Jun 2021 at 18:17, Henning Westerholt <hw(a)skalatan.de> wrote: > Hello, > > > > Good observation. Please run the memory statistics CLI commands to get > more hints about the module that might cause it (as per below link). Then > please report more details. If you can point to a particular module, you > can also open an issue on our tracker. > > > > https://www.kamailio.org/wiki/tutorials/troubleshooting/memory > > > > Cheers, > > > > Henning > > > > *From:* sr-users <sr-users-bounces(a)lists.kamailio.org> *On Behalf Of *George > Diamantopoulos > *Sent:* Friday, June 25, 2021 4:53 PM > *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > *Subject:* [SR-Users] Possible memory leak on 5.5.x (new)? > > > > Hello all, > > > > I'm still investigating the (most likely non-kamailio-related) memory leak > of my previous message to the list, there have been no developments so far. > I'll update if anything changes. > > > > This concerns a new finding which seems to affect kamailio 5.5.x. I have > two kamailio instances receiving the same traffic via round-robin. I > upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared > of any issues arising. I was lucky to do so, because with identical > configuration, 5.5.x seems to run out of SHM very quickly. Here are links > to graphs produced by our monitoring system: > > > > Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png > > New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png > > > > The configuration uses mtree, htable, vars and vns extensively. Has anyone > come across anything similar? Let me know if I can provide any further > information to help disect this. Thanks! > > > > BR, > > George >

2 8

Issues running mongodb command from kamailio
by Edo 18 Jan '22

18 Jan '22

Hi All, Please help! I am getting an error when trying to start kamailio after making some changes to the configuration file. Error is : *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 47-57: '('')' expected (function call)* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad expression* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad command* *atus=255/EXCEPTION* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 103: bad command* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 105: bad command* * used inside params of another function: xlog* *rse error in config file /etc/kamailio/kamailio.cfg, line 879, column 28: use of function execution inside params not allowed* *Server.* Changes made is : # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; #Start of change if (mongodb_find_one("mongodbsrv1", “telephone", “cXXX", " { \'did\': \'2XXXXXXXXXXXXXXXXX\' } ", "mgr1")) { xlog("response from mongodb is [[$mongodb(mgr1=>value)]]\n"); } #End of change append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } Please can someone tell me what I am doing wrong. Error says syntax but I have tried many permutations of the suggested syntax with no luck. Please help Thanks -- ----- Ekunwe EDO Network Services, Inc. Tel: 601.497.3932

2 4

REGISTER for MS Teams
by Lucian Romi 18 Jan '22

18 Jan '22

Hi, can register send from ms teams or I have to use Kamailio to send to my PBX.

2 1

JS Kemi TypeError: undefined not callable (property 'ifdef' of [object Object])
by Huseyin Kaan Koyukan 18 Jan '22

18 Jan '22

Hello Everyone, First of all, this is my first ever e-mail to the group so I'd like to thank all the developers who built this great product and thank everyone for the kind and helpful community. I've been using Kamailio for the past 8 months, I've had lots of fun with it. My issue is I'm trying to set-up JS Kemi interpreter for my Kamailio-Asterisk box. I translated the config file to JS kemi, which wasn't too bad but I'm getting a type error for the "KSR.kx.ifdef("WITH_ASTERISK")" statement that I used. Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:1478]: dukopen_KSR(): module 'KSR' has been initialized Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:473]: jsdt_sr_init_child(): loading js script file: /usr/local/etc/kamailio/kamailio-basic-kemi-jsdt.js Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: app_jsdt [app_jsdt_api.c:447]: jsdt_kemi_load_script(): failed running: TypeError: undefined not callable (property 'ifdef' of [object Object]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [core/sr_module.c:865]: init_mod_child(): error while initializing module app_jsdt (/usr/local/lib64/kamailio/modules/app_jsdt.so) (idx: 0 rank: -127 desc: [main]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [main.c:1671]: main_loop(): error in init_child(PROC_INT) -- exiting In syslog I can see that kex.so module is loaded successfully and 'KSR' has been initialized. I don't know what I'm doing wrong. A few sample JS Kemi code that I could find did not use the kx.ifdef function. Was this a new addition to the interpreter? Perhaps my Kamailio version (kamailio 5.5.0 x86_64/linux) does not support it? Thank you all in advance. Kaan.

1 0

Help with rewriting headers for NAT manually
by Chad 17 Jan '22

17 Jan '22

We are looking for some help (possibly a paid consultant) to help us with our Kamailio setup. To keep this as short as possible: we use Kamailio as a NAT proxy to bridge our external IP and our private IP asterisk servers (via dispatcher). However both the external IP and the internal IP that the Kamailio server uses are virtual IPs created by keepalived. Because of that neither mhomed nor fix_nated_contact work, and we use force_send_socket to direct the traffic. We run linux Debian 10 for the OS. Also we do not use a DB at all, everything is done with local config files. The problem is that when traffic goes out the Contact header has a private IP in it, like: Contact: <sip:##########@10.10.10.###]:5060> There are 2 possible solutions to this: 1. Make changes to linux, keepalived and/or Kamailio so that Kamailio recognize the virtual IPs so that mhomed and fix_nated_contact work as usual. 2. Create a manual header rewrite system. If solution #2: What we need to do is create a way to rewrite the contact header to the external IP on the way out, and on the way back rewrite it back to the internal server that the call is already connected to. Not sure if we will need to store those paths on the server or if we can do some kind of cheat with another persistant header like P-Preferred-Identity or P-Asserted-Identity (i.e. store the internal IP in the name field or something). If anyone out there know of a way to do this or wants to give it a try please reach out to me. Thank you all for your time. -- ^C Chad

5 27

Alternate to FoHSS for Kamailio IMS?
by Simon,Benjamin T 17 Jan '22

17 Jan '22

Hello, We are attempting to setup an instance of Kamailio to serve as an IMS for a test setup of 4G/5G NSA with Open5GS. The instructions on the Kamailio wiki and provided by Open5GS currently specify to use FoHSS as the HSS solution for Kamailio IMS. As this HSS solution has not seen significant development in a few years we are reticent to deploy it on our implementation. Is there an alternative open source HSS which can be deployed to serve as the HSS for Kamailio instead of FoHSS, and would there need to be any adjustments to Kamailio to reflect the deployment of this HSS instead of FoHSS? Thank you, Benjamin Simon Research Assistant Florida Institute for Cybersecurity Research, University of Florida Email: bsimon4(a)ufl.edu Disclaimer: Florida has a very broad public records law. As a result, any written communication created or received by the University of Florida will be made available to the public and media, upon request, unless otherwise exempt. Under Florida law, e-mail addresses are public records. If you do not want your e-mail address released in response to a public records request, do not send electronic mail to this address. Instead, contact by phone or in writing.

2 2

RR balance kamailio
by Gokan Atmaca 16 Jan '22

16 Jan '22

Hello I installed kamailio on Ubuntu. The version information is as follows. (0) I added two dispatcher sip servers. (1) When I enter the parameter as below in the configuration file , I get an error as follows. (2) I am getting RR module not available error. Packages installed are "kamailio-berkeley-modules, kamailio-tls-modules, kamailio-presence-modules" How can I solve the problem? (0) version: version: kamailio 5.5.3 (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: unknown compiled with gcc 9.3.0 (1) Dispatcher list: { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } DEST: { URI: sip:Y.Y.Y.Y FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } } } } } (2) if ( method=="INVITE" ) { ds_select_dst("1","4"); sl_send_reply("100","Trying"); forward(); exit(); } Err_log: Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

2 3

Mid-dialog failover without virtual IP
by Muhammad Danish Moosa 14 Jan '22

14 Jan '22

Hi, I have a pair of kamailio stateful proxies and am trying to achieve Active-Passive failover model with DMQ dialog replication. Primary Proxy is working in the middle of 2 SBCs and all nodes are within the same network so NATing is NOT involved ;secondary proxy is accessible through VPN. It's pretty simple configuration, proxy is just adding and removing headers in the middle. So if I stop the kamailio service on primary IP , SBCs blacklist that and start sending new invites to secondary IP. That's all good. But if I reproduce this failover during a connected session , SBCs do not send mid-dialog requests/responses to failover IP. I can see they still try to send messages to primary SBC which obviously end up with timeout. I understand the most natural solution is HA with Virual IP but it's not possible because secondary IP is accessible through VPN. I tried to use DNS SRV record with priorities but that also does not work with mid-dialog messages, I also have tried to change record-route/via header to fqdn through advertise_address. Its not changing record-route but via only but I am still not getting expected results. Any guidance of possible solution(s) here will be appreciated. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

3 2

Massive uploading of aliases through a dbtext
by Denys Pozniak 14 Jan '22

14 Jan '22

Hello! I need help uploading a list of aliases through a dbtext to use them via myself. It is likely that there is some configuration problem: *loadmodule "db_text.so"* *modparam("db_text", "db_mode", 1)modparam("db_text", "emptystring", 1)* *loadmodule "alias_db.so"modparam("alias_db", "db_url", "text:///etc/kamailio/dbtext")* *[root@kamailio]$ cat dbtext/dbaliasesid(int,auto) alias_username(string) alias_domain(string) username(string) domain(string)1::sip.test.local::* *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { }}* If done through a script (alias=sip.test.local), then everything works fine: *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { alias: { proto: * address: sip.test.local port: * } }}* -- BR, Denys Pozniak

2 4

Retrieve last via IP from SIP MEssage
by Adarsh Chauhan 14 Jan '22

14 Jan '22

Hi All, I have tried couple of methods to retrieve last via IP of SIP message but failed to retrieve it, trying to retrieve the source - @hf_value.via[-1].host, it simply prints the exact text back @hf_value.via[-1].host - $sel((a)hf_value.via[-1]) it doest give the IP but the entire via value SIP/2.0/TCP 172.31.30.144;branch=z9hG4bKf8bd.753341d2000000000000000000000000.0.host @via[-1].host it also simply prints the exact text back @via[-1].host Regards, Adarsh Chauha

2 1

EVAPI workload allocation across processes
by Alex Balashov 13 Jan '22

13 Jan '22

Hi, Given several EVAPI worker processes: modparam("evapi", "workers", 4) # kamcmd -s /run/kamailio/kamailio_ctl ps | fgrep -i evapi 539257 EvAPI Dispatcher 539258 EvAPI Worker 539259 EvAPI Worker 539260 EvAPI Worker 539261 EvAPI Worker How should one understand the way that the EVAPI workload is allocated among them? My best guess is that the EvAPI dispatcher process is the sole accept()or of client connections and the place where all relaying to client connections is done, regardless of their number, and that the ‘workers’ represent an async pool of processes in which TM transactions can be resumed or other work can be done when an outside caller sends a message to the dispatcher. However, the documentation[1] is a little vague on the meaning of the ‘workers’ modparam: "Number of worker processes to be started to handle incoming messages from remote applications.” Or maybe it’s not. Just wanted to confirm that my understanding is correct. Thanks! — Alex [1] https://kamailio.org/docs/modules/5.5.x/modules/evapi.html#evapi.p.workers -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Reverse Kamailio Dispatcher order
by Adarsh Chauhan 12 Jan '22

12 Jan '22

Hi All, I was looking for a way where i could reverse the order of priority set on destinations in dispatcher file. for eg: if i have the following file: 1 sip:10.252.168.41:5060;transport=tcp 0 5 # (US) 1 sip:10.252.168.42:5060;transport=tcp 0 4 # (US) 1 sip:10.252.168.43:5060;transport=tcp 0 3 # (EU) 1 sip:10.252.168.44:5060;transport=tcp 0 2 # (EU) I wanted to use top to bottom when in US region and bottom to top when in EU region. One option is to make another set, however that will force Kamailio to send ping (OPTIONS) twice when probing. Regards, Adarsh Chauhan

4 4

Suggestion on receiving AAR
by Jyoti Bansal 10 Jan '22

10 Jan '22

Hello all, We are trying to connect kamailio ims to a pcrf, connection is successful. Whenever we register through bhoghe client , we receive AAR on pcrf from kamailio ims(pcscf), so just need a suggestion can we disable that till we receive invite from the soft phone? Can anyone help on this? Thanks, Jyoti bansal -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

Call Preservation Mode in Kamailio
by varun pratapsingh 10 Jan '22

10 Jan '22

Happy New Year 2022 Everyone. Wish you a lot of innovation this year! I was just curious to know if any of us had handled the mid-call failures using the Kamailio. I am writing a SIP Client to handle the failures by using the DNS SRV based failover and Fallback approach where I am using two Kamailio instances as Outbound Proxy resolved by DNS SRV lookup. Everything is fine except for on-going calls. How to handle the mid-call failovers? Is there any way in Kamailio like Call Preservation mode or something? Any suggestions or ideas to handle midcall failover so that if a current Outbound Proxy goes down then all the ongoing calls can be seamlessly routed through another Kamailio (secondary). Thanks and Regards Varun

3 2

Kamailio call flows with Asterisk
by Nauman Sulaiman (SESSIONTALK) 05 Jan '22

05 Jan '22

Hi, We are using Asterisk as a PBX with users directly registered to Asterisk and Asterisk registering to SIP trunks. We are now looking to put Kamailio in front of Asterisk to handle SIP registrations from the SIP clients. In a ‘typical’ architecture should we keep the SIP trunk registrations on Asterisk or is Kamailio used for this? We want to keep Asterisk as the RTP proxy so we don’t want a call setup by Kamailio with RTP then going direct between user agents. Regards Nauman

3 2

Calling t_relay([host, port]) with variables
by Nick J 05 Jan '22

05 Jan '22

Hi all, I've been banging my head on the table with this one and hoping someone has some insight here. I'm trying to use a variable as the host & port input in t_relay() - t_relay(host, port) This works fine if statically set: t_relay("10.0.1.1", "5060"); But if I change the host or the port to a variable like this: $avp(destip) = "10.0.1.1"; $avp(dstport) = "5060"; t_relay($avp(destip), $avp(dstport)); Then Kamailio fails to start, calling out the t_relay line: kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio-routelogic.cfg, line 62, column 35: bad command I get the same behavior with pseudovariables and vars as well: $var(destip) = "10.0.1.1"; $var(dstport) = "5060"; t_relay($var(destip), $var(dstport)) Any tips or pointers would be much appreciated. Thanks! Nick

4 3

Features of Kamailio as SBC
by Nattha Vudhiprasittipol 04 Jan '22

04 Jan '22

Hello I would like to know all the feature lists of Kamailio as SBC. Could you please send me the document? I cannot find it on the website. Best regards Nattha Vudhiprasittipol Product Development Engineer Creaturelab Network Solution Co.,Ltd 357 Charansanitwong Rd., Bang-Bamru, Bangphlat, Bangkok 10700, THAILAND Email: nattha.v(a)creaturelab.co.th Website: www.creaturelab.co.th

7 19

Recommendation for compatible IMS soft phones
by Ashay Matekar 04 Jan '22

04 Jan '22

Hi Team, We were trying to test a call flow with Kamailio IMS that requires preconditio. Do you know any free or paid IMS softphones that support preconditio and are compatible with Kamailio IMS. -- *Thanks and Regards,* *Ashay Matekar* Associate Manager Engineering | +91 932446823 Great Software Laboratory | www.gslab.com LinkedIn <https://www.linkedin.com/company/gs-lab> | Twitter <https://twitter.com/_gslab> | Email Disclaimer <https://www.gslab.com/privacy-policy/#email-disclaimer> -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

Catching Register 200 OK, then using it to validate other packets.
by Richard Edmands 04 Jan '22

04 Jan '22

Hey Kamailio Users, I've written a TLS-UDP gateway. It works well, but I want to reduce the internet noise it pushes through. I have an idea on what I'd like to try. Wondering if it would be possible to catch the Register 200 OK, store it and then when receiving packets it future to then validate them against the data? I've managed to capture a 200 OK, but the save("location") function says it will return it's own 200 OK which isn't the plan since we just want to capture the location. Anyone done something similar and got some working example code? Is this a bad idea for some reason?

2 2

Re: [SR-Users] Kamailio docker image run in Kubernates
by Henning Westerholt 27 Dec '21

27 Dec '21

Hello, (please keep the list in CC) Can you pull the v2 image on the command line with the CLI commands from your repository? Do you have Maybe its some naming or access rights issue. You need to try out different things in isolating the cause, it’s difficult to guess. Have a look e.g. here for more debug options for this error https://stackoverflow.com/questions/34848422/how-can-i-debug-imagepullbacko… Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Maryam Baghdadi <maryam.baghdadi(a)gmail.com> Sent: Monday, December 27, 2021 10:15 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio docker image run in Kubernates I dedicate a service and demonset.apps in yaml file. When I assign a label to a node of k8 and then select node according to node label in yaml file for first image for example image:v1 every thing is OK and pulling done successfully . But for second time when I rebuild docker image after changing kamailio configuration for example image:v2 generated , pulling error happens… On Tue, Dec 28, 2021, 12:34 AM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, as mentioned in the issue tracker, the error seems to be related to a failure of k8s pulling the second docker image. Have you tried to debug the second services locally, for example with docker-compose or some local k8s distribution? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Maryam Baghdadi Sent: Monday, December 27, 2021 9:44 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] Kamailio docker image run in Kubernates Hello everyone, I have a problem over running local docker images of Kamailio in Kubernate cluster.. Consider I build a local docker image from Kamailio and create a pod in one of my nodes . My process created successfully and Kamailio listen over port 5060 now I need to rebuild another docker image with new Kamailio configuration and run it as another pod with new name and listening over port 5061 but I have error "ImagePullBackOff" . I test this scenario many times but problem exists.. It is devastating for me and I need emergency help.

1 0

Kamailio docker image run in Kubernates
by Maryam Baghdadi 27 Dec '21

27 Dec '21

Hello everyone, I have a problem over running local docker images of Kamailio in Kubernate cluster.. Consider I build a local docker image from Kamailio and create a pod in one of my nodes . My process created successfully and Kamailio listen over port 5060 now I need to rebuild another docker image with new Kamailio configuration and run it as another pod with new name and listening over port 5061 but I have error "ImagePullBackOff" . I test this scenario many times but problem exists.. It is devastating for me and I need emergency help.

2 1

Kamailio Stop processing SIP traffic
by pwerspire 27 Dec '21

27 Dec '21

Hi, I have been doing some load testing with Kamailio and having some issues. I have been trying with 100 CPS and at the beginning, everything is working well but after some time (it can be 10 minutes or for example 40 minutes, is just random but normally happens more in the first 10 minutes) Kamailio stop replying to all SIP messages but still processing HTTP requests, some commands, etc. If I use "kamcmd dlg.list" no output happens and after that, all kamcmd commands just keep loading with no output. Kamailio is sharing Dialogs and htables using DMQ with other Kamailio that is the failover, if I shut down the failover one (so no DMQ replication), the test works well with the 100 CPS. Also tried with modparam("dmq", "worker_usleep", 1000) but the behavior is the same, it will stop processing traffic. This is some of configuration used: # ----------------- setting module-specific parameters --------------- modparam("dmq", "server_address", "sip:INTERNAL_INSTANCE_IP:5060") modparam("dmq", "notification_address", "DMQ_NOTIFICATION_ADDRESS") modparam("dmq", "multi_notify", 1) modparam("dmq", "ping_interval", 10) modparam("dmq", "num_workers",4) modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) # ----- dialog params ----- modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", DBURL_RW) modparam("dialog", "db_mode", 0) modparam("dialog", "enable_dmq", 1) modparam("dialog", "db_update_period", 10) modparam("dialog", "h_id_start", H_ID_START) modparam("dialog", "h_id_step", H_ID_STEP) Memory: Shared: 4096 Private: 512 kamailio -v version: kamailio 5.5.3 (x86_64/linux) 473cef 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: 473cef compiled on 09:34:57 Dec 20 2021 with gcc 10.2.1 In attach is the trap collected after the issue happens. Any more logs or configurations that can help identify or solve the issue? Thanks for the help, Regards, Tiago

6 11

How to change language in siremis web interface
by mohammad baghdadi 27 Dec '21

27 Dec '21

Hello I’m new in siremis as web interface of kamailio and I immediately need your help,I installed kamailio and siremis and login in siremis web interface,I can work with it but now I have a problem. I want to change menu language there is an option in(setting->translate->manage languages) add my desired language and click save and i see this error: Error message: #2, count(): Parameter must be an array or an object that implements Countable Script name and line number of error: [/var/www/siremis-5.3.x/siremis/modules/translation/lib/LangPackCreator.php I needed to change the language and thanks for your the help to me.

2 1

MongoDB and SIREMIS
by Tihomir Culjaga 27 Dec '21

27 Dec '21

hi, I'm running kamailio 5.4 using MongoDB as a backend database. Can I run SIREMIS on that database as well? Regards, Tihomir.

2 1

Merry Christmas and Happy Holidays 2021-2022!
by Daniel-Constantin Mierla 25 Dec '21

25 Dec '21

Another challenging year is getting close to its end, pressuring even more everyone! Let's hope things will go back to a more usual life (as it used to be) in the near future! Kamailio project continued to develop, bringing out a new major release, preparing for the next one in a few months, relying on an amazing community of developers and users. We also trust it continued to help people and companies world wide to connect with the dear ones and build a better way of living. The season holidays are ahead, time to relax, spend time with family and friends! I take again this chance to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2021 another great year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

repository not present
by Jyoti bansal 23 Dec '21

23 Dec '21

2 1

Active/active keepalived - Kamailio includes non-local virtual IP in "myself" which breaks DMQ call routing
by Rhys Hanrahan 22 Dec '21

22 Dec '21

Hi Everyone, I have Kamailio acting as a registrar, using DMQ to replicate usrloc between nodes. I’m relaying INVITEs to the home registrar, but skipping this if (uri == myself) is true. In my lab I just setup my SBCs in an active/active mode, similar to the link below. So I have 2x VIPs and both SBCs listen on both, but the first VIP is only master on SBC01 and the second VIP is only master on SBC02. https://docs.nginx.com/nginx/admin-guide/high-availability/ha-keepalived-no… My issue is that now “myself” will always return true even if the virtual IP is actually not active on this node at the moment, so the invite is never relayed properly if it’s on another node. I guess “myself” is just blindly looking at the list of Ips that are configured with either listen= or alias= and doesn’t care if the IP actually exists on the server at the time, or not? I am thinking I may have to ditch relying on “myself” entirely and come up with some other way of checking for local requests, but I am wondering if there’s a good way of approaching this? I was thinking of perhaps triggering an add/remove of listen directives (or something) when a failover of a VIP happens, though this seems kind of heavy handed. Hoping there is a better way. Has anyone else tried doing active/active Kamailio instances like this? Thanks! Rhys Hanrahan | Chief Information Officer e: rhys(a)nexusone.com.au<mailto:rhys@nexusone.com.au> [www.nexusone.com.au]<http://www.nexusone.com.au/> [signature_132622389] <http://www.fusiontech.com.au/> NEXUS ONE | FUSION TECHNOLOGY SOLUTIONS p: 1800 NEXUS1 (1800 639 871) or 1800 565 845 | a: Suite 12.03 Level 12, 227 Elizabeth Street, Sydney NSW 2000 www.nexusone.com.au<http://www.nexusone.com.au/> | www.fusiontech.com.au<http://www.fusiontech.com.au/> The information in this email and any accompanying attachments may contain; a. Confidential information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; b. Legally privileged information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; and or c. Copyright material Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties. If you have received this email in error, please notify the sender immediately and delete this message. Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd does not accept any responsibility for loss or damage arising from the use or distribution of this email. Please consider the environment before printing this email.

3 6

Issue Bug in SEMS (transparent SBC profile)
by Mojtaba 21 Dec '21

21 Dec '21

Hello, Why in SEMS, some changes (in transparent SBC profile) in sip headers do not apply. For example i want to change CSeq or Contact header like this: # transparent SBC profile # defaults: transparent RURI=$r From=$f To=$t Call-ID=$ci_leg2 CSeq=$H(CSeq) Contact=$H(Contact) When I run SEMS, CSeq and Contact do not change, while other headers change. [#7fc688d7b1c0/9379] [onLoad, SBC.cpp:189] INFO: loading SBC call profiles from '/usr/local/etc/sems/etc/' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:418] INFO: SBC: loaded SBC profile 'mo' - MD5: d6a2c29084043edde1357231f6866737 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:423] INFO: SBC: RURI = '$r' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:424] INFO: SBC: RURI-host = '' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:425] INFO: SBC: From = '$f' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:426] INFO: SBC: To = '$t' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:431] INFO: SBC: Call-ID = '$ci_leg2222' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:434] INFO: SBC: force outbound proxy: yes This issue caused the Asterisk could not pass the Authentication process in "match_req_to_dialog" function because the CSeq is the same as the dialogs. So the Asterisk thought this message was a LOOP message. Has anybody encountered this issue in SEMS? How did you solve this? I know there are some ways for doing this with Kamailio, But i focused on issues in SEMS With Best Regards *-- Mojtaba Esfandiari.S* *-- PhD student and Research Affiliate, * *-- Technical Manager at IP-PBX Laboratory, Ferdowsi University of Mashhad, Iran.* -- Address: IP-PBX Lab., Engineering Faculty, Ferdowsi University Main Campus, Mashhad, Iran. -- Tel: +98-51-38763635 -- Mobile: +98-915-117-6713

4 14

uac module - use htable instead of database
by Markus Bönke 20 Dec '21

20 Dec '21

Hello all, we want to use the "remote registrations feature" of the uac module to let kamailio register with other sip providers. As far as I see the module works only with a database backend. As our kamailio script should be independent from a database, a solution could be to load the configuration on startup (and refresh it periodically) from a database view into a htable and then trigger the remote registrations based on the expiration of the htable columns. Could something like this be implemented by using the uac module or by an other workaround ? Or does it require some enhancements in the uac module ? Thanks and regards Markus

2 2

Using kamcmd tm.t_uac_start to send SIP REGISTER messages in Kamailio 5.5
by Steve Sether 20 Dec '21

20 Dec '21

I'm trying to use the kamcmd tml.tc_uac_start command to send raw SIP messages in Kamailio 5.5, but I always get a error: 400 - Invalid headers. I've also tried sending OPTIONS messages with the same result. I've looked for examples of sending SIP messages via kamcmd, but can't find any. kamcmd help tm.t_uac_start specifies the following: kamcmd> help tm.t_uac_start starts a tm uac using a list of string parameters: method, ruri, dst_uri, send_sock, headers (CRLF separated) and body (optional) Essentially my problem is I don't really know how to properly encode these headers. How do I send special characters? Do I need to use double quotes, single quotes, etc around the whole header? Which characters are special? How do I encode the CRLF? I tried \r\n, but I still get this error. It says only From: and To: are required. It's a bit hard trying to make progress when all I ever get is one error message. The kamailio logs are a little more helpful, and largely point me down the road that it doesn't understand the CRLF encoding, and says my From: header includes the To: line, so it's not separating them. For example, I'm trying things like this: kamcmd -v tm.t_uac_start OPTIONS sip:200@exampleserver.com:5060 . . "From: <sip:example@192.168.0.1>;tag=d27ca05e-1789-4774-a698-dcfe91dec9b4\r\nTo: <sip:example@192.168.0.2>"\r\nExpires: 1200\r\nContent-Length: 0" Nothing seems to work, so whatever the syntax I'm using must be wrong. Just providing an example of sending SIP via kamcmd tm.t_uac_start would likely help a huge amount. Thanks! --

2 1

tcp_set_connection_lifetime(1) does not close connection
by Sebastian Damm 19 Dec '21

19 Dec '21

Hi, for having a maintenance mode, I need to implement some sort of draining functionality. All our clients register via TLS. So I thought, I just answer the next REGISTER of a client with a 503 and kill the tcp connection afterwards, so they need to open up a new connection to the loadbalancer in front and end up on a different server. This is my code: 510 if msg.is_method("REGISTER"): 1 if KSR.pv.get("$shv(MAINTENANCE_MODE)") == 1: 2 KSR.info("In maintenance mode, rejecting registration") 3 KSR.hdr.append_to_reply("Retry-After: 2\r\n") 4 KSR.sl.sl_send_reply(503, "Draining") 5 if KSR.tcpops.tcp_set_connection_lifetime(1) > 0: 6 KSR.info("Set Connection lifetime to 1") 7 return False (Side note: msg is some internal class, this is obviously KEMI with python) So now when I set the MAINTENANCE_MODE var to 1, Kamailio rejects the REGISTER requests, logs both lines and sends out a 503. Debug log tells me this: Dec 10 15:32:10 ifens5 /usr/sbin/kamailio[3851]: DEBUG: tcpops [tcpops.c:229]: tcpops_set_connection_lifetime(): new connection lifetime for conid=1: 908866403 Dec 10 15:32:21 ifens5 /usr/sbin/kamailio[3855]: DEBUG: <core> [core/tcp_main.c:4554]: tcpconn_main_timeout(): entering timer for 0x7f759cb26218 (ticks=908866499, timeout=908866499 (0 s), wr_timeout=0 (211631299 s)), write queue: 0 bytes In tcpdump I can see that the connection does not get closed immediately. Is there some timer in the tcp stack which cleans up expired connections from time to time? Can I manipulate how often this timer runs? I couldn't find anything, and clearly even though I tell Kamailio to close the connection after 1 second, it doesn't do it. Thanks for any input. Regards, Sebastian

3 3

Kamailio SIP Application server
by Weiqi Yao 17 Dec '21

17 Dec '21

Hello community, We were testing using Kmailio work as an IMS, and then turn out the phone wants to do MMTel for mobile real-time multimedia communication. We are not familiar with this topic, but anyone know Kamailio have such module or any other MMTel service can work with Kamailio? BR, Weiqi

2 2

RPC command dlg.profile_get_size with per_value response
by David VILLAUME 17 Dec '21

17 Dec '21

Hello, I would like to suggest you an implementation of the rpc command dlg.profile_get_size (on a valued profile) that would return a list of values and the number of current dialogs for any of them . Din’t know where to suggest it so I sent it there, please accept my apologies if it’s not the good place. Regards, David

2 2

SRTP/DTLS + RTP Header encryption
by Rahul mm 15 Dec '21

15 Dec '21

Hi guys, Is it possible to achieve RTP header encryption along with payload in SRTP/DTLS ? So far what I've been able to see is RTP payload encryption. I'm using kamailo 5.5.0 and rtpengine 9.2.1.6 in webRTC scenario. Many thanks. BR, Rahul

2 3

Next online devel meeting - Wed, Dec 15, 2021, 15:00UTC
by Daniel-Constantin Mierla 15 Dec '21

15 Dec '21

Hello, 2021 is almost over, time ran fast somehow this year, with a major release in the middle, which delayed the development discussion for the next one, so it's time for it! As usual, the aim is to sketch the roadmap to next major release so the developers and community members can syncronize and plan what should be done till Kamailio v5.6 will be out as well as discuss about current state of the project, look for new ideas to improve collaboration within the community, a.s.o. The meeting will be done on a matrix chat room. More details are in the wiki page of the event, available at: * https://www.kamailio.org/wiki/devel/irc-meetings/2021a Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, December 15, 2021, at 15:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev

1 2

ACK is sent to wrong address when using serial forking
by Minh Phan 15 Dec '21

15 Dec '21

Dear List, As shown in the image below, Proxy 1 tries to do serial forking to Proxy 3 and then Proxy 4. Upon reception of the 480 answer ( blue frame) from Proxy 2, the Proxy 1 generated an ACK (red frame) and sent it to Proxy 1. The problem I have is that the ACK generated by Proxy 1 has a wrong RURI. I expect the ACK to be sent to the URI of the CONTACT header of the 480 message (blue frame). However, its RURI is the same as the one in the first INVITE of the dialog. Note that there is NO ROUTE header in the ACK message. As there is serial forking here, the ACK is generated by the Proxy 1 itself. Any hint would be appreciated, Best Regards, Minh Phan [image: image.png]

3 2

RTP Proxy with Local IP and Public IP
by Vijo Jose 13 Dec '21

13 Dec '21

Hai Everyone. I have configured the RTP Proxy with Kamailio SIP Proxy Server. Where I have my Kamailio is configured with private IP 10.0.0.1 and Public UP 103.XX.XX.XX and the SIP user is connecting with RTP proxy with local IP 10.0.0.1 to register. I also have a SIP Asterisk server on the public Internet with IP 90.XX.XX.XX that makes the call to sipuser@103.XX.XX.XX:5060 which is coming on the sip proxy server and the sip proxy server forwards those packets to the SIP user who is registered with local IP 10.0.0.5. Once the call is answered the RTP is started to send from the SIP server to the SIP Proxy server over the public IP from 90.XX.XX.XX to 103.XX.XX.XX which is working as expected. But from the sip user 10.0.0.5 the RTP is flowing to Proxy server public IP 103.XX.XX.XX instead of RTP Proxy server Localip 10.0.0.1. How can we configure the same? Presently I have started my RTP proxy server with command /usr/local/bin/rtpproxy -F -A 103.XX.XXXX -l 10.0.0.1 -m 10000 -M 20000 -s udp:127.0.0.1:7722 -d DBUG:LOG_LOCAL5 Thanks in advance.

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Re: [SR-Users] Manipulation of request-line
by Mihovil Kolaric 12 Dec '21

12 Dec '21

Hi, unfortunately, Alex is right: manipulations ot the request-URI do not seem to re-write the whole line, the malformed line-ending (SIP/2.0<blank><CR><LF>) remains in the forwarded message.

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remove rport in INVITE Via HDR and make kamailio behave asymmetric
by Nuno Miguel Reis 11 Dec '21

11 Dec '21

Hi guys. I've a scenario here where I'm receiving inbound traffic from a Mitel PBX on a LAN environment with a bunch of router hops between kamailio and the Mitel. Sometimes, not always, I get traffic from the MItel PBX where the source port is changed in transit (some kind of NAT, still don't know where exactly) from 5060 to a dynamic port different from 5060. Every INVITE comes with rport in via header as: .. Via: SIP/2.0/UDP 10.21.252.35:5060 ;branch=z9hG4bK-524287-1---d6bac01809527627*;rport* .. when this happens kamailio sends all the replies to the $si:$sp symmetrically (force_rport behavior) and I would like to force the opposite instead, making it asymmetric and sending the traffic to $si:5060 everytime. I'm having a hard time forcing the *unforce*_rport() behavior. Does anyone have a good suggestion for this? Looking forward to hear from you. Cheers, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

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Unquoted values from #!defenv
by Ben Kaufman 09 Dec '21

09 Dec '21

Hello all, Looking for some clarification regarding quoting and !#defenv . I understand that the example below fails because the environmental variable HOME is not quoted, thus when used in the xlog() call it fails. How is it possible to evaluate and/or use the value of the environmental variable if defined as a preprocessor variable? Looking at the documentation the reason for this directive is "It is a simplified alternative of using #!substdef with $env(NAME) in the replacement part." But #!substdef would allow setting the value inside of a quoted string, so does it really meet that requirement? I'm aware that I could use $env(HOME) rather than a preprocessor directive of HOME. I'm just trying to understand the usage of the #!defenv feature. #!KAMAILIO ## Tested in 5.5.2 loadmodule "xlog" loadmodule "pv" loadmodule "evrexec" modparam("evrexec", "exec", "name=evrexec:timer;wait=1000;workers=1;") #!defenv HOME request_route{ forward(); } event_route[evrexec:timer] { xlog("L_N", "HOME: " + HOME + "\n"); } Ben Kaufman

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wireshark decode kamailio / rtpengine ng messages
by Karsten Horsmann 08 Dec '21

08 Dec '21

Hi List, maybe someone wrote a wireshark helper (didnt know the right term) to view kamailio to rtpengine communication in human form? Some hints would be great. -- Mit freundlichen Grüßen *Karsten Horsmann*

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Re: [SR-Users] wireshark decode kamailio / rtpengine ng messages
by Alex Balashov 08 Dec '21

08 Dec '21

That sounds far too niche for a helper, though, then again, Wireshark has some phenomenally exotic helpers. — Sent from mobile, with due apologies for brevity and errors. > On Dec 8, 2021, at 8:14 AM, Karsten Horsmann <khorsmann(a)gmail.com> wrote: >

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Kamailio v5.3.9 Released
by Daniel-Constantin Mierla 08 Dec '21

08 Dec '21

Hello, Kamailio SIP Server v5.3.9 stable release is out. This is a maintenance release of the old stable branch, 5.3, that includes fixes since the release of v5.3.8. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.9 or even better to v5.4.x or 5.5.x series. For more details about version 5.3.9 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/12/kamailio-v5-3-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.3 is an old stable branch, v5.3.9 being the last planned release out of it, to mark the end of official maintenance. The maintained stable branches are now 5.4 and 5.5, at this time with v5.5.3 being the latest stable release. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Re: [SR-Users] Manipulation of request-line
by Alex Balashov 08 Dec '21

08 Dec '21

That will manipulate the request URI, but not the request line. — Sent from mobile, with due apologies for brevity and errors. > On Dec 7, 2021, at 4:16 PM, Karsten Horsmann <khorsmann(a)gmail.com> wrote: >

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Releasing Kamailio 5.3.9 - last in 5.3.x series
by Daniel-Constantin Mierla 08 Dec '21

08 Dec '21

Hello, it is time to mark the end of official packaging for 5.3.x releases, therefore during the next days I am going to cut the v5.3.9 release from branch 5.3, likely on Wednesday, Dec 8, 2021. In other words, if no major regression is detected shortly after this release, the v5.3.9 is going to be the last one from the 5.3 branch. It is strongly recommended to move to a series that is under active maintenance, respectively 5.5.x or 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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HTTP Requests for API routing - Question on which module to use with KSR
by Emilio Panighetti 08 Dec '21

08 Dec '21

Looking into using Kamailio to route calls via an HTTP API. Basic scenario is this: INVITE comes in (no registration). And Kamailio sends an HTTP request to a server with data extracted from the INVITE. Based on the response, call KSR.setdsturi() and other actions like normalizing phone numbers, etc. (Not sure if changing Contact: headers would be a better option). For now I want to be able to make a simple call routing as a proof of concept. Planning on using Lua as a KEMI interpreter. Looking at different modules: http_client http_client_async xhttp Which module would be more appropriate? http_client_async provides a simple async method and callback, so the SIP threads can continue processing requests while the API server is preparing the response. But the call has to be parked (How is this done?) and this is also an overhead. http_client seems straightforward, but the SIP threads are tied up waiting for HTTP responses. Also looking at the KEMI documentation, the KSR function calls for HTTP_CLIENT doesn’t seem to match the config function documentation. For example, In KEMI manual: KSR.http_client.curl_connect_post() doesn’t have a corresponding in kamailio.cfg. So which would be the preferred approach? Also examples would be appreciated. Thanks in advance!

1 0

Manipulation of request-line
by Mihovil Kolaric 07 Dec '21

07 Dec '21

Hi, I have an issue with a SIP-client, that occasionally puts a blank between the version SIP/2.0 and the CRLF at the and of the request-line. (i.e. sends "ACK <uri> SIP/2.0 \r\n" instead of "ACK <uri> SIP/2.0\r\n"). Kamailio (as well as most SIP UAs) does not care about that error and successfully processes/forwards the request (also when using sanity_check()). But I also deal with a client that fails to parse that request. (Furthermore, wireshark's dissector fails to recognize such malformed messages as SIP, which makes debugging even harder). I tried to fix this issue using textops, but all functions I tried operate only on headers or message body, but do not seem to affect the request line. Is there any possibility to modify the request-line, or to replace it with a new one? Best Regards, Mihovil

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Announcing public beta release of TeamsPhone - powered by Kamailio
by Sergiu Pojoga 06 Dec '21

06 Dec '21

Hello Kamailio-nians, Just to bug everyone quickly: today I'm happy to announce the public beta release of *TeamsPhone <https://teamsphone.net>*, a MS Teams <=> PBX integration solution for ITSPs, IP-PBX service providers and integrators, *powered by Kamailio*. Unlike other similar services offered as SaaS pay-per-user model, our product is oriented for ITSPs to be self-hosted and fully integrated into one's telecom network, providing an add-on service to supplement any Hosted PBX offering with seamless calling using Microsoft Teams *as-as-softphone*, or as we like to call it - TeamsPhone. Some insights about the platform: - 100% automation of MS Teams Direct Routing provisioning - multi-tenant application - API-driven - Microservices oriented architecture - RabbitMQ-powered Message Broker distribution - SIP server (you guessed it - Kamailio, of course!) Seeing how it works is worth a thousand words. A cloud *demo* instance is available at https://my.teamsphone.net for anyone to try it out, use it or abuse it. If you have questions about the product - feel free to contact me in private. Respectfully, Sergiu Pojoga Founder, teamsphone.net sergiu(a)teamsphone.net

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detecting compact headers?
by Brooks Bridges 06 Dec '21

06 Dec '21

We have an internal component of our callflow that does not support compact headers (vendor says it's on the roadmap but no eta, etc) so I'm trying to find a clean way to detect if a message has compact headers and reject it. I haven't had much luck in finding a way to identify specifically if any of the headers of the message are compact however. Can anyone point me in a direction please? Thanks! Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

3 3

ENV Values in cfg
by H Yavari 06 Dec '21

06 Dec '21

Hi to all, Is it possible to set modules parameters with env values? I know that in routes we can use $env(NAME), but what about for other parameters? for example setting DB address in K8s cluster ENV values. Regards,Hossein

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DMQ Usrloc/Dialog - Experiences
by Carsten Bock 03 Dec '21

03 Dec '21

Hi, I wanted to quickly ask the group about experiences with DMQ and Usrloc/Dialog. I don't expect any issues when choosing such an approach, especially since we would have re-transmits for a message (fr_timer = 30 seconds) if we did not receive any answers or scenarios alike. So I assume, having short "outages" (less than 30 seconds) for the communication is just fine. So far I used DMQ in a non-IMS setup for synchronizing usrloc with ~15k Subscribers and it worked really well. One concern brought up (not by me) was, that synchronizing usrloc with DMQ might not be reliable at scale. I would also love to understand how to treat situations when the communication is lost for a longer period of time (e.g. if the communication between nodes is lost for 5 minutes?). Can anyone share some real-life experiences? Thanks, Carsten -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

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topos xavu_field_contact_host kamailio 5.5
by Jonathan Hunter 03 Dec '21

03 Dec '21

Hi All, I was wondering if anyone can help, and advise me if I am missing something when using topos in an environment where I have more than one interface for kamailio. Essentially I would like to set the contact_host to 2 different values depending on the flow of the SIP requests, as we have a private LAN interconnect for example 10.10.10.10, and a public interconnect using 2.2.2.2 (for example). I notice there is xavu for A-side contact and B-side contact, but what about just the contact host? As I have set for example; modparam("topos", "xavu_cfg", "_tps_") modparam("topos", "xavu_field_contact_host", "contact_host") ... $xavu(_tps_=>contact_host) = "2.2.2.2"; And this works fine but obviously sets the contact_host to 8.8.8.8 across both interfaces, I ideally want it to change between 10.10.10.10 and 2.2.2.2. How can I set more than one contact_host $xavu that topos can use? As my understanding is you can only set $xavu once, so is there another way to manipulate the contact_host on requests coming in and out on different interfaces? I assume I am missing something? :) Thanks in advance Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows

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