- sr-users - kamailio.org

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.

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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

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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

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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

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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

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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.

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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?

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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.

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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.

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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

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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.

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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.

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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

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repository not present
by Jyoti bansal 23 Dec '21

23 Dec '21

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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.

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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

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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

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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! --

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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

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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

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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

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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

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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

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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]

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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

1 1

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

3 15

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*

2 1

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: >

1 0

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

1 0

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: >

2 1

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/

1 1

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

2 1

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

1 0

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

2 1

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

3 5

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

2 1

NIST advisory
by David Villasmil 02 Dec '21

02 Dec '21

Anyone knows about this? https://nvd.nist.gov/vuln/detail/CVE-2020-28361 -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 4

Re: [SR-Users] probles with listen advertise address trying to be resolved
by Richard Robson 01 Dec '21

01 Dec '21

no the problem appear to be with the advertise this makes all listeners listen = PRIV_SOCK advertise MY_PUB_SOCK name = "ext" but listen = PRIV_SOCK_TLS advertise MY_PUB_SOCK_TLS name = "ext_tls" is failing with a look up error on MY_PUB_SOCK_TLS if I try : listen = PRIV_SOCK_TLS advertise "MY_PUB_SOCK_TLS" name = "ext_tls" it fails on a lookup for what ever the def is for that variable e.g. 1.2.3.4:5061 the defs are of the correct syntax if it replace the defs it works e.g listen = tls:10.10.10.10:5061 advertise 1.2.3.4:5061 name = "ext_tls" the two defs math the text above the problen is its working for the "global" defs without any protocl, so its making the tls listener for the wrong port as 5060 so listen = LISTEN_EXTERNAL_SOCKET advertise ADVERTISE_NAT_ADDRESS name = "ext" is working with #!substdef "!LISTEN_EXTERNAL_SOCKET!10.10.10.10:5060!g" and #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" the _TLS versions are not changing the tls port so i have tls on port 5060 when iot should be on 5061 and the name ext_tls where all other sockets are called "ext" Regards, Richard On 30/11/2021 20:03, Karsten Horsmann wrote: > Hi Richard, > > Hmmm for some reason I split my vars into parts. > > And you prefix that with tls? Like this? > > "!ADVERTISE_NAT_ADDRESS_TLS!tls:1.2.3.4:5061!g" > > Best > Karsten > > Richard Robson <richard(a)rikrobson.co.uk> schrieb am Di., 30. Nov. > 2021, 17:48: > > Hi > > > the missing = was my typo in the email and not the config. > > the probelm is only with the TLS advertise and not the non tls > advertise > > these work as expected > > listen = LISTEN_EXTERNAL_SOCKET advertise ADVERTISE_NAT_ADDRESS > name = "ext" > > #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" > > > these do not works > > #!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4:5061!g" does not work > > listen = LISTEN_EXTERNAL_SOCKET_TLS advertise > ADVERTISE_NAT_ADDRESS_TL name = "ext-tls" > > or > > listen = LISTEN_EXTERNAL_SOCKET_TLS advertise > ADVERTISE_NAT_ADDRESS_TLS name = "ext-tls" > > > > if its not quoted then error = cannot resolve 1.2.3.4:5061 > <http://1.2.3.4:5061> of not quoted error = cannot resolve > ADVERTISE_NAT_ADDRESS_TLS > > > if i replace the def with the 1.2.3.4:5061 <http://1.2.3.4:5061> > it works > > > what is hte difference if also changed the name to > ADVERTISE_NAT_ADDRESS_SEC with the same results > > > > > On 30/11/2021 16:21, Karsten Horsmann wrote: >> Hi Richard, >> >> seems that you forgot the equal signs in the listen directive. >> Did kamailio -c -f /etc/kamailio/kamailio.cfg return something? >> http://www.kamailio.org/wiki/cookbooks/devel/core#listen >> >> I did the same with puppet. And I only manipulate my >> /etc/kamailio/kamailio-local.cfg with the configuration >> management tool. >> Small example from my templating. >> >> grep MY_PUB_IP /etc/kamailio/kamailio-local.cfg >> #!substdef "!MY_PUB_IP!1.2.3.4!g" >> .... >> egrep 'MY_HTTPS_ADDR|MY_PUB_IP|_PORT' /etc/kamailio/kamailio.cfg >> import_file "kamailio-local.cfg" >> .... >> #!substdef "!MY_HTTPS_ADDR!tls:MY_PUB_IP:MY_WSS_HTTPS!g" >> .... >> listen=udp:MY_PUB_IP:5060 >> listen=tcp:MY_PUB_IP:5060 >> listen=tls:MY_PUB_IP:5061 >> listen=MY_HTTPS_ADDR >> >> Kind regards >> Karsten >> >> Am Di., 30. Nov. 2021 um 16:57 Uhr schrieb Richard Robson >> <richard(a)rikrobson.co.uk>: >> >> I have a problem when trying to add an advertised address in >> Kamailio 5.4.2 >> >> >> we have a checf config that is populating various variable. >> some of these are for the Listeners >> >> >> >> we have this working and Im getting all protocols for the >> listeners based on the non _TLS Vars but the TLS config is >> trying to resolve ADVERTISE_NAT_ADDRESS_TLS. If I try and >> quote the variable "ADVERTISE_NAT_ADDRESS_TLS" id get an >> error saying 1.2.3.4:5061 <http://1.2.3.4:5061> is not resolvable >> >> >> therefroe I have udp,tcp and tls listeners all on the default >> ports where i need the tls on port 5061 ans not 5060 >> >> >> listen = LISTEN_EXTERNAL_SOCKET advertise >> ADVERTISE_NAT_ADDRESS name = "ext" >> >> listen = LISTEN_INTERNAL_SOCKET name = "int" >> >> listen = LISTEN_EXTERNAL_SOCKET_TLS advertise name = "ext-tls" >> >> listen = LISTEN_INTERNAL_SOCKET_TLS name = "int-tls" >> >> the Vars are defined as >> >> #!define TOPO_NAT >> >> #!substdef "!LISTEN_EXTERNAL_SOCKET!10.10.10.10:5060!g" >> >> #!substdef "!LISTEN_INTERNAL_SOCKET!10.10.10.10:5080!g" >> >> #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" >> >> #!define ENABLE_TLS >> >> #!define TLS_CONFIG_FILE "chef-managed/tls.cfg" >> >> #!substdef "!LISTEN_EXTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" >> >> #!substdef "!LISTEN_INTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" >> >> #!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4:5061!g" >> >> REgards, >> >> >> Richard >> >> >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> * sr-users(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not >> reply only to the sender! >> Edit mailing list options or unsubscribe: >> * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> -- >> Mit freundlichen Grüßen >> *Karsten Horsmann* >> >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> *sr-users(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to the sender! >> Edit mailing list options or unsubscribe: >> *https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >

2 1

Using path with DMQ to send calls to origin SBC - not sure when to modify destination URI
by Rhys Hanrahan 01 Dec '21

01 Dec '21

Hi Everyone, I am trying to add DMQ for redundancy of registrations and USRLOC, and I’m trying to send calls to the correct SBC that is the original registrar for a handset. I’ve been using the thread here where Charles gave some guidance on how to use path to store and use the original SBC: https://lists.kamailio.org/pipermail/sr-users/2018-February/100246.html and https://lists.kamailio.org/pipermail/sr-users/2013-September/079736.html I am struggling with when to decide to modify the destination URI. My testing shows this is required otherwise some handsets will ignore the invite, but right now I am doing it in a blanket form, right before SIPOUT (which is where the origin SBC handles the invite instead of LOCATION). I am sure this is being too heavy-handed though and there are some cases where I won’t want to set this, but I am not sure which? # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } … xlog("Setting du according to path. Current du is $du\n"); xlog("Current route header: $(hdr(Route))\n"); xlog("Current route: $(hdr(Route){uri.host})\n"); >>> $du = $(hdr(Route){param.value,received}); #xlog("New du destination uri is: $du\n"); # dispatch requests to foreign domains route(SIPOUT); In the linked threads, Charles mentioned that only the last-hop registrar should make this change, but what’s the best way to determine if I am on the last-hop registrar? As per the snippet above, I tried using the {uri.host} transformation to extract the origin SBC’s IP from the route header. My plan is to then compare this against “myself” but I am struggling to extract the right info from the route header. And I am not even sure if this is the right general approach? The route header looks like this: <sip:ORIGIN_SBC_IP:5060;received=sip:UAC_WAN_IP:2048;lr> Any guidance or examples would be appreciated. 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_555064778] <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.

2 5

Re-writing presentity domain for DMQ
by Rhys Hanrahan 01 Dec '21

01 Dec '21

Hi Everyone, I am trying to get DMQ working for presence/dialog notifications. Presence works fine locally on the same SBC. The DMQ part itself is also working fine and I can see that the entries in presentity are being replicated between my DMQ nodes. The problem is that when the presentity records are being replicated from another SBC, the domain for that record stays as the IP of the original SBC. Then when the local_watchers is populated, the presentity_uri doesn’t match, and no NOTIFY packets are sent as a consequence. What am I missing here? I can’t see anything in the docs about how to handle this, and I can’t find any examples in the list on how to handle this either, which suggests it isn’t as common of an issue at it seems. See example of what I see in the presentity table below. In this case handset 103 is subscribed to BLFs for 103 and 104. A call came into 103 and generated an early state dialog in the presentity table. MariaDB [kamailio]> select * from presentity; +-------+-------------+--------------+--------+-- | id | username | domain | event | +-------+-------------+--------------+--------+-- 27100 | nexusone103 | x.x.x.53 | dialog | I have modified my registrations and have re-written my SUBSCRIBE and PUBLISH packet headers so that everything uses a common FQDN for both SBCs, that does DNS RR to both SBC IPs. I am also registering from the handset using this FQDN now, and yet although my active_watchers table now expects a presentity URI of XXX(a)sbc.fqdn<mailto:XXX@sbc.fqdn>, the presentity table is still using the IP addresses of the individual node that originated the dialog. MariaDB [kamailio]> select presentity_uri,watcher_username,watcher_domain,event from active_watchers; +----------------------------------------------+------------------+----------------------+-----------------+ | presentity_uri | watcher_username | watcher_domain | event | +----------------------------------------------+------------------+----------------------+-----------------+ | sip:104@sbc.fqdn | nexusone103 | sbc.fqdn | dialog | | sip:103@sbc.fqdn | nexusone103 | sbc.fqdn | dialog | +----------------------------------------------+------------------+----------------------+-----------------+ I’m out of ideas unfortunately. Any advice would be appreciated. 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_1328564021] <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.

1 0

probles with listen advertise address trying to be resolved
by Richard Robson 30 Nov '21

30 Nov '21

I have a problem when trying to add an advertised address in Kamailio 5.4.2 we have a checf config that is populating various variable. some of these are for the Listeners we have this working and Im getting all protocols for the listeners based on the non _TLS Vars but the TLS config is trying to resolve ADVERTISE_NAT_ADDRESS_TLS. If I try and quote the variable "ADVERTISE_NAT_ADDRESS_TLS" id get an error saying 1.2.3.4:5061 is not resolvable therefroe I have udp,tcp and tls listeners all on the default ports where i need the tls on port 5061 ans not 5060 listen LISTEN_EXTERNAL_SOCKET advertise ADVERTISE_NAT_ADDRESS name = "ext" listen LISTEN_INTERNAL_SOCKET name = "int" listen LISTEN_EXTERNAL_SOCKET_TLS advertise name = "ext-tls" listen LISTEN_INTERNAL_SOCKET_TLS name = "int-tls" the Vars are defined as #!define TOPO_NAT #!substdef "!LISTEN_EXTERNAL_SOCKET!10.10.10.10:5060!g" #!substdef "!LISTEN_INTERNAL_SOCKET!10.10.10.10:5080!g" #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" #!define ENABLE_TLS #!define TLS_CONFIG_FILE "chef-managed/tls.cfg" #!substdef "!LISTEN_EXTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" #!substdef "!LISTEN_INTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" #!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4:5061!g" REgards, Richard

2 2

Kamailio and RTPEngine acting as SBC : wrong IP address returned
by Cyril Ramière 28 Nov '21

28 Nov '21

Hi there, I'm already using Kamailio as a simple proxy and it worked flawlessly. This time, I need to have a new setup like this: [UAC (sip phone)] <--- register & has to deal only with ----> | PUBLIC | [Kamailio + RtpEngine] | PRIVATE | <----> [Asterisk] Client = 37.100.100.100 (public) Asterisk = 172.17.200.20 (private) Kamailio/RtpEngine = 172.17.203.102 (private, advertising 3.99.99.99 public IP) Kamailio and RtpEngine will behave as a SBC/B2BUA, the client will deal only with kamailio using public access (internet), and asterisk will handle the calls but stays 100% private behind kamailio. Kamailio will handle registrations to reduce the load put on the asterisk boxes. The setup is running on AWS, my Kamailio has 1 Private IP address tied to 1 Public IP address (an Elastic public IP) so, no multi-homing, one network interface. I made a "simple" work in progress configuration, the registration is working as a PoC, and my calls are *almost* working but with some nonsense. My Main issue is that when my UAC (SIP phone) make a call, to let's say +1000, the call enters kamailio, then kamailio handle the invite, trigger rtpEngine, but the IP addresses in Record-Route and Via headers are wrong. >From Kamailio to UAC Sip Phone, it seems OK. But from Kamailio to Asterisk, the record-route/via headers are set with Kamailio Public IP when it should be the Kamailio private IP. So, when Asterisk sends a BYE, it tries to send it to the public IP of Kamailio, which is wrong. There are probably other quirks in the configuration but this one, I can't figure out what's wrong. I'm attaching a capture file, a log file and my configuration file (some informations are redacted). I am missing something and sadly I can't figure out what, a little help would be appreciated :) Regards

2 1

Problem implementing Outbound Proxy
by Christoph Russow 26 Nov '21

26 Nov '21

Hi all, im currently trying to add a Outbound Proxying to our system which right now does inbound loadbalancing for our application servers. Current: Inbound SIP Trunk -> Kamailio -> dispatcher -> ApplicationServer 1..n i now want to add the ability for our internal Application Servers to terminate their outbound calls through the kamailio system. currently i've setup a gateway in the (freeswitch) application server that has the kamailio system set as outbound-proxy. this leads to sip calls started with originate sofia/gateway/kamailio/<number>@<outbound carrier IP> to be sent to the kamailio system which then forwards it to the given carrier IP. Sadly our main carrier answers with "503 service unavailable - no bandwith available". (maybe due to the internal ip-addresses within some of the headers?) i've attached our current kamailio configuration and also a pcap of a call i tried to start with our main carrier. i tried this way as it seemed to be the easiest way so our application servers still can select the outbound carrier within their flow. maybe i'm on the wrong way here and need to solve this problem in a completely different way i don't know. TLDR: sip proxying does not work as expected. need a way to achieve outbound calls routed through kamailio from internal application servers any hint/help into the right direction is appreciated Best Regards Christoph Russow -- ________________________________________________________________ EMTEX GmbH Christoph Russow Software Engineer Bischof-Otto-Weg 9 D-91086 Aurachtal Geschäftsführer: Markus Enzinger Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 ________________________________________________________________

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Call limiting with Kamailio
by Olli Attila 26 Nov '21

26 Nov '21

Hello, What choices do I have to implement a feature to Kamailio which would limit calls per customer? The environment: Two kamailio nodes sitting in different sites and they are not aware of each others sessions/dialogs and thus serving customer calls independently. Configurations on the nodes are identical excluding ip addresses. Customer call can be routed through either one of these Kam nodes. The two nodes share a common external mysql database from which all dialplan and htable entries are loaded (used in call routing for example). Customers are identified based on IP address & tel numbering and as a final result we are populating an variable which holds the customer id $var(REALM_NAME) when the identification has been done. The call limiting should be "tagged/profiled" somehow with this information if possible. Any suggestions which way to go with this? Initial ideas if one of these are worth of setting a starting point: -> Counting calls with htable and replicating with DMQ to other node (can the data survive from a restart?) -> Using the common mysql database to hold the counted calls from where both kam nodes could load data? (database load increasing?) -> Some new exciting module for this job? Cheers, Olli

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cannot adopt OIDs in KAMAILIO-SIP-COMMON-MIB
by Arun Bonela 26 Nov '21

26 Nov '21

Hi we have configured snmp module on our Kamailio server following https://www.kamailio.org/dokuwiki/doku.php/utils:kamailio-and-snmp guidance. Description we could able to poll or snmpwalk most of the OIDs in KAMAILIO-REG-MIB , KAMAILIO-SIP-SERVER-MIB but having issues below from KAMAILIO-SIP-COMMON-MIB snmpd[358730]: Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatsMethod ::= { kamailioSIPMethodStatsEntry 1 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatsOutbounds ::= { kamailioSIPMethodStatsEntry 2 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatsInbounds ::= { kamailioSIPMethodStatsEntry 3 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeMethod ::= { kamailioSIPStatusCodesEntry 1 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeValue ::= { kamailioSIPStatusCodesEntry 2 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeIns ::= { kamailioSIPStatusCodesEntry 3 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeOuts ::= { kamailioSIPStatusCodesEntry 4 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeRowStatus ::= { kamailioSIPStatusCodesEntry 5 } Troubleshooting root@host# snmpwalk -v2c -c public localhost .1.3.6.1.4.1.34352.3.1.1.1.4.1.1.2.0 SNMPv2-SMI::enterprises.34352.3.1.1.1.4.1.1.2.0 = No Such Object available on this agent at this OID root@host# snmpwalk -v2c -c public localhost .1.3.6.1.4.1.34352.3.1.1.1.4.1.1.2 SNMPv2-SMI::enterprises.34352.3.1.1.1.4.1.1.2 = No Such Object available on this agent at this OID Reproduction using snmpwalk or snmget to any of the above mentioned kamailioSIPStats* OIDs will do Additional Information * Kamailio Version - output of kamailio -v version: kamailio 5.5.1 (x86_64/linux) 307af8-dirty * Operating System: No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 20.04.2 LTS Release: 20.04 Any advice, why we seeing this issue for these OIDs ? Are these decremented now ? Regards, Arun Bonela Arun Bonela UCaaS Ops – Technical Lead Tel: 0333 043 7319 Mobile: 07468 728 007 Email: Arun.Bonela(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at 5 Fleet Place, London, EC4M 7RD and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

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Re: [SR-Users] How to not write logs to journald with kamailio
by Ginhoux, Patrick 26 Nov '21

26 Nov '21

Hi, I resend this mail because of the initial destination address was not correct. The topic was a logging problem : With the upgrade of my SIP Router proxies to RHEL 7 (and Kamailio 5.0.7) , we encounter problems where the logs are blocked to be written in the syslog. This problem is similar as the one reported in this article : [SR-Users] Kamailio logs in RHEL 7 environment <https://lists.kamailio.org/pipermail/sr-users/2018-April/101065.html> . The resolution was to increase some journald/rsyslog settings (RateLimitBurst=1000000 / $imjournalRatelimitInterval 1 $imjournalRatelimitBurst 500000). But it is somehow a workaround. Im looking to know if it should be possible for kamailio to write directly the logs to the syslog only. Thanks in advance to those can help me. Regards Patrick Ginhoux De : Ginhoux, Patrick Envoyé : lundi 22 novembre 2021 16:49 À : 'sr-users' <sr-users-bounces(a)lists.kamailio.org> Objet : How to not write logs to journald with kamailio Hi, With the upgrade of my SIP Router proxies to RHEL 7 (and Kamailio 5.0.7) , we encounter problems where the logs are blocked to be written in the syslog. This problem is similar as the one reported in this article : [SR-Users] Kamailio logs in RHEL 7 environment <https://lists.kamailio.org/pipermail/sr-users/2018-April/101065.html> . The resolution was to increase some journald/rsyslog settings (RateLimitBurst=1000000 / $imjournalRatelimitInterval 1 $imjournalRatelimitBurst 500000). But it is somehow a workaround. Im looking to know if it should be possible for kamailio to write directly the logs to the syslog only. Thanks in advance to those can help me. Regards Patrick Ginhoux

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