- sr-users - kamailio.org

TypeError exception in Kemi with python3
by Michel Pelletier 06 Jan '23

06 Jan '23

Hello, I am using Kemi with Python3. I am getting a TypeError exception when using the following: KSR.presence.handle_subscribe() 40(208) INFO: PY3 {SUBSCRIBE}: app_python3 [apy_kemi.c:1839]: sr_apy_kemi_exec_func(): alert - action KSR.presence.handle_subscribe(...) took too long [2004761 ms] (file:(null) func:(null) line:733) 40(208) ERROR: PY3 {SUBSCRIBE}: app_python3 [python_support.c:156]: python_handle_exception(): apy_exec: ksr_request_route((null)): Unhandled exception in the Python code: TypeError: expected bytes, str found The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 152, in ksr_request_route if self.ksr_route_presence(msg) == -255: File "/etc/kamailio/kamailio.py", line 733, in ksr_route_presence KSR.presence.handle_subscribe() SystemError: <built-in function handle_subscribe> returned a result with an error set Using the following appears to be a good workaround, as it does not exhibit the issue at all: KSR.presence.handle_subscribe_uri(KSR.pv.get("$fu")) Cheers, Michel Pelletier

1 0

Call drop when reinvite is proceed
by nathan.castillo＠snapcom.fr 05 Jan '23

05 Jan '23

Hello all, I'm trying to figure out one problem on my Kamailio server. When kamailio receive a reinvite from the infra, I get the error message "477 - Unforunately error on sending to next hop" and the call drop. I don't understand why this message appears as the first invite is well answered and the call is going well. Any ideas how to resolve this issue ? Thanks in advance for your help. Regards, Nathan

6 6

Too many 200:OK sip responses
by Waqar 40 05 Jan '23

05 Jan '23

Dear All, For a normal call, Kamailio receives 200:OK sip response for the method invite only once but for some calls, it receives 200:OK too many times around 6 to 7. And I also do not receive BYE or any other packets for this kind of call. How can I detect such calls in Kamailio and cancel them? I am inserting data of the calls in the database when the call occurs and deleting it when BYE is received but for these calls as I don't receive BYE so I am unable to delete the record of such calls. Is there a way to detect such calls? Like How can I count if the number of 200:OK responses is more than one or anything like that? Regards Vicky

4 4

How to have Kamailio retry backends
by Unai Rodriguez 04 Jan '23

04 Jan '23

Dear List, We’re using Kamailio to load balance MRCP requests to multiple backend groups with a configuration as follows: # kamailio.cfg ... ... route[DISPATCH] { if($ua=="mrcp_backend_1") { if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } } if($ua=="mrcp_backend_2") { if(!ds_select_dst("2", "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; } ... ... # dispatcher.list 1 sip:mrcp01.server.int:8060;transport=tcp 1 sip:mrcp02.server.int:8060;transport=tcp 2 sip:mrcp03.server.int:8060;transport=tcp 2 sip:mrcp04.server.int:8060;transport=tcp With this configuration, Kamailio load balances the initial SIP INVITE among the MRCP servers. After the INVITE, the service communicates directly to the MRCP servers via SIP (for hanging up the call), MRCPv2 (for sending speech control messages), and RTP (for sending audio). We would like to implement a configurable number of retries, so that if a particular backend times out, Kamailio would retry X times to other backend(s). In short, something equivalent to HAProxy’s retries, but for Kamailio. This probably implies having Kamailio always as part of our communication (not just load balancing the initial SIP INVITE). I haven’t been able to find much information about this, could someone provide some pointers? Thank you so much With best wishes, Unai Rodriguez

3 6

Recover call in ringing state
by Devang Dhandhalya 04 Jan '23

04 Jan '23

Hello All I have below setup : SBC -> kamailio -> Media server(remote) -> web-client Call comes from sbc to kamilio and it relays to Remote server to end user, In Ringing state if remote server goes down then in kamailio there is any way to Recover that ringing call like we can generate new invite for other Remote server which is active state, for relaying call i am using dispatcher module.. I tried with tm.cancel RPC command but it works only if the remote server is running , when I crash the remote server at that time tm.cancel is not generating cancel to remote server. Please suggest if there is any other way to generate INVITE. Regards Devang Dhandhalya -- * <https://www.ecosmob.com/> * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 3

Job offer Kamailio/VoIP developer
by Henning Westerholt 03 Jan '23

03 Jan '23

Hello and Happy New Year, we are looking for a new Kamailio & VoIP developer colleague to join and grow our international team. More detail about the position can be found at: https://gilawa.com/en/about/careers If you are interested, please send me a (direct) email. Please include the information described in the linked job description. Best regards, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Localy generate BYE message from ims_charging
by masoud96＠mail.ru 03 Jan '23

03 Jan '23

Hello! I have a problem with writing the BYE message from the ACC module to the database, because the messages are generated locally by the ims_charging module at the end of the credit and the flag will not work. You can help catch such local messages and make a record in the database.

2 2

Question about UAC and PUA modules routing behavior
by Jean-Francois Lemieux 03 Jan '23

03 Jan '23

Hello, I have some trouble to figure out how routing works when the request coming from the kamailio itself (used as a user agent per example) .I try to use presence PUA... modules. The issue I have is the NOTIFY requests aren't sent to the good destination; it doesn't seems to obey to the contact or other informations when the device did a SUBSCRIBE. I don't want to ask to troubleshoot the presence module directly I just want to know how Kamailio deals with request coming from Kamailio itself (in the case of a NOTIFY generated by PUA_... modules My mains questions are: Does the pua modules when generating request like NOTIFY will pass through request_route ? Does the pua modules when generating request like NOTIFY will be sent only when t_relay() is used or UA kamailio based request bypass all the logic of the Kamailio proxy behavior ? I'm trying to understand if I can modify the NOTIFY request to change destination before the "routing decision" is made by Kamailio Kamailio 5.6.0 used as : -proxy to remote pbx for call dialog -registrar/location server -media relaying and nat management -...and I try to use it as a presence server instead to use the pbx behind kamailio

3 2

Steering egress on replies
by Alex Balashov 03 Jan '23

03 Jan '23

Hi, Happy New Year! From what I understand, replies are always sent back out of the listener on which they were received, though `reply_to_via` (off by default) can override this and cause the exit listener to be recalculated based on a real-time assessment of the best way to reach the last Via hop. Two questions arise: 1) If turning `mhomed` off, can `$fs` be used to override the exit interface of locally generated replies, too? If so, where should it be done? In a request_route prior to a send_reply() / sl_send_reply() / t_reply() call? Or should it be done inside a local reply route or onsend_route? 2) Does this differ for locally generated replies vs. relayed replies, either statefully or statelessly? Thanks! -- Alex -- 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 1

Registration Proxy
by nutxase 02 Jan '23

02 Jan '23

Hi Guys, can someone point me in the right direction as to how i can use kamailio as a registration proxy similar to how opensips does mid_registrar? i want to proxy all sip registrations through kamailio to asterisk or freeswitch Sent with [Proton Mail](https://proton.me/) secure email.

3 2

Happy New Year 2023!
by Daniel-Constantin Mierla 01 Jan '23

01 Jan '23

A Happy, Peaceful And Prosperous Year 2023 to everyone using Kamailio and helping to improve the project! Looking forward to meeting many of you around this world during 2023 and collaborating in a friendly environment! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST)
by Lewis Hutchinson 30 Dec '22

30 Dec '22

Hi Brandon Apologies if you have provided this but do you have a PCAP of the INVITE / 400 Bad Request rather than the Pastebin of the INVITE The last time I had this was because of a missing /r/n on the SDP granted it was not with teams but still difficult to track down. Also, what is the MTU Size, have you checked to see if the is causing packet fragmentation and manifesting in a Bad Request. Are there any clues in the Teams - Call Analytics. Regards Lewis -----Original Message----- From: sr-users-request(a)lists.kamailio.org <sr-users-request(a)lists.kamailio.org> Sent: 29 December 2022 20:17 To: sr-users(a)lists.kamailio.org Subject: sr-users Digest, Vol 211, Issue 67 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe 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. Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST) (Brandon Armstead) 2. Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST) (Brandon Armstead) ---------------------------------------------------------------------- Message: 1 Date: Thu, 29 Dec 2022 12:15:16 -0800 From: Brandon Armstead <brandon(a)cryy.com> Subject: [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST) To: Henning Westerholt <hw(a)gilawa.com> Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Message-ID: <CABAX3EoF+hfruLiV2Nh_g117_q8diaKxVEyp5hMFO=wyKbQidQ(a)mail.gmail.com> Content-Type: multipart/alternative; boundary="0000000000005692d905f0fd2538" Henning, There was no reply (reason) available in my case. - Brandon On Thu, Dec 29, 2022 at 11:48 AM Henning Westerholt <hw(a)gilawa.com> wrote: > Hello, > > > > as mentioned before, have a look to the 400 reply reason phrase. This > was done quite good from their side, better than many other vendors. > > Usually, Teams is not too bad if you are using a defined > infrastructure that do not introduce too many variables. We are > usually using a B2BUA in our customer projects for that purpose. > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt - > https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fskal > atan.de%2Fblog%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk% > 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b% > 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat > a=QU%2BzOgPdbmRYh51a9PvptIvVjnTZrGxT8z1G7b2RRGQ%3D&reserved=0 > > Kamailio services - > https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgila > wa.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c > 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7C0% > 7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQI > joiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=PuDqkZ > QCj2IlxvRsLQGoKFv19K3uVE5RBxvcPKPODwQ%3D&reserved=0 > > > > *From:* Brandon Armstead <brandon(a)cryy.com> > *Sent:* Thursday, December 29, 2022 7:52 PM > *To:* Kamailio (SER) - Users Mailing List > <sr-users(a)lists.kamailio.org> > *Subject:* [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400 > BAD > REQUEST) > > > > Alex & Kaufman, > > > > Appreciate the feedback. The pastebin is one example of about > 1000+ iterations literally of changes. So I'm finally reaching out > :). I haven't tested the socket attribute in record-route change, I > will try this now. As for the angle brackets I've tried with and > without, etc (already) to no effect. The SDP unknown media type seems > to be a Poly thing, so I > *want* to rule this out as its not in my other SDP's and still receive > 400 bad request. > > One more iteration here we go... please feel free to let me know if > you have any other thoughts on the matter :). > > > > Thanks! > > > > - Brandon > > > > On Thu, Dec 29, 2022 at 8:08 AM Alex Balashov > <abalashov(a)evaristesys.com> > wrote: > > Yeah, the grammar says that rr-params are just generic-params, in > which case it's not. I didn't see that -- nicely spotted! > > I think that may be the basis of the 400 Bad Request. I'd be shocked > if it weren't. > > The broken clock of "Microsoft SIP" can still be right twice a day. > > > On Dec 29, 2022, at 10:31 AM, Kaufman <bkaufman(a)bcmone.com> wrote: > > > > In your top Record-Route you have: > > > > socket=; > > > > Not sure if that is legal. > > > > Kaufman > > > > -----Original Message----- > > From: Alex Balashov <abalashov(a)evaristesys.com> > > Sent: Thursday, December 29, 2022 9:02 AM > > To: Kamailio (SER) - Users Mailing List > > <sr-users(a)lists.kamailio.org> > > Subject: [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400 > > BAD > REQUEST) > > > > Sorry to hear you're having to interoperate with Teams. It's a > > unique > form of sadism I wouldn't wish upon anyone. > > > > A few theories: > > > > 1) Microsoft doesn't like the "bare" Contact header-value here: > > > > Contact: > sip:+MY_FROM_PHONE_NUMBER_HERE@MY_FQDN_WAS_HERE:5061;transport=tls > > > > Unlike the careted one right above: > > > > P-Asserted-Identity: > > <sip:+MY_FROM_PHONE_NUMBER_HERE@MY_FQDN_WAS_HERE> > > > > A bare URI absent other header-params is of course completely legal, > > but > I'm really trying to get inside the imaginary world of antisocial > "Microsoft SIP" here. > > > > 2) Could it be that antisocial "Microsoft SIP" sends 400 as a way of > objecting to something in the SDP, e.g. where a non-demented SIP stack > would send "488 Not Acceptable Here" or "415 Unsupported Media Type"? > > > > I know you've said you tried multiple clients to rule that out, but > > I > wonder if the thing it's objecting to hasn't been ruled out that way. > > > > 3) I saw this media line in the SDP: > > > > m=application 41356 <unknown media type> > > > > What's that? > > > > -- Alex > > > >> On Dec 29, 2022, at 9:51 AM, Brandon Armstead <brandon(a)cryy.com> wrote: > >> > >> Outbound calls from my SBC into Teams (Polycom -> SBC -> Teams) > >> always > result in a 400 BAD REQUEST. > >> Example invite below: > >> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fn > >> am11.safelinks.protection.outlook.com%2F%3Furl%3Dhttps%253A%252F%25 > >> 2Fpast&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c > >> 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7 > >> C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDA > >> iLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sda > >> ta=5v20CvNnmdiHwfWMBX6tgjMlE1i2aKpFgTnuVFfpyl0%3D&reserved=0 > >> ebin.com%2FF1G1Ce59&data=05%7C01%7Cbkaufman%40bcmone.com%7C55672dbf > >> 739 > >> a4ff447f008dae9b0c9b2%7Cafc1818e7b6848568913201b9396c4fc%7C1%7C0%7C > >> 638 > >> 079242759152531%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjo > >> iV2 > >> luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=Cbff6B4g > >> Nr8 > >> A9zs89smJ129y8IyNM9%2B3zVkGhlzpa54%3D&reserved=0 > >> I've taken care to make sure numbers are all E.164 format in > From/To/Contact. I've also taken care to make sure that FQDN is used > in Contact and Record-Route header. > >> I've tried many different variations and have followed the SIP > information here: > >> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fn > >> am11.safelinks.protection.outlook.com%2F%3Furl%3Dhttps%253A%252F%25 > >> 2Flear&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c > >> 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7 > >> C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDA > >> iLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sda > >> ta=ELk0UD1Pleq%2FG9j1%2BT18tiuFe0yjflimR9ztnW%2F7oPM%3D&reserved=0 > >> n.microsoft.com%2Fen-us%2Fmicrosoftteams%2Fdirect-routing-protocols > >> -si > >> p&data=05%7C01%7Cbkaufman%40bcmone.com%7C55672dbf739a4ff447f008dae9 > >> b0c > >> 9b2%7Cafc1818e7b6848568913201b9396c4fc%7C1%7C0%7C638079242759152531 > >> %7C > >> Unknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6I > >> k1h > >> aWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=GO26A9FOOoS42yMCWPPGo5PkKo > >> 75c > >> kjSJSkjAeYRsU0%3D&reserved=0 I've also tried several different > >> clients (Bria, Polycom CCX 600, Grandstream, etc) to see if maybe > >> it was > something in the SDP or otherwise causing an issue. > >> SIP Transport is TLS, RTP is SRTP > >> I might also add that OPTION pings are active and Direct Routing > Endpoint is active, so this is successful. I also am able to receive > calls FROM teams to my IP phone(s) without issue. It is only when I > try and call INTO teams (INVITE -> Microsoft Teams) that I always > receive a 400 BAD REQUEST to my INVITE. > >> Any help is appreciated, thank you! > >> > >> - Brandon > >> > >> __________________________________________________________ > >> Kamailio - Users Mailing List - Non Commercial Discussions To > >> unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > >> Important: keep the mailing list in the recipients, do not reply > >> only > to the sender! > >> Edit mailing list options or unsubscribe: > > > > -- > > Alex Balashov | Principal | Evariste Systems LLC > > > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > > Web: > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.e > varistesys.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk% > 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b% > 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat > a=hbww6wEEztE85YqxcN92bpMR%2BkiI9%2BE3jF1i6fQpjEk%3D&reserved=0, > > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c > srpswitch.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7 > Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7 > C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMD > AiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata > =4E7a5ZxSvttBiMxl1VTtolmTvS4T5VGZHA00CsbPR2c%3D&reserved=0 > > > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions To > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply > > only to > the sender! > > Edit mailing list options or unsubscribe: > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions To > > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply > > only to > the sender! > > Edit mailing list options or unsubscribe: > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.e > varistesys.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk% > 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b% > 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat > a=hbww6wEEztE85YqxcN92bpMR%2BkiI9%2BE3jF1i6fQpjEk%3D&reserved=0, > https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c > srpswitch.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7 > Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7 > C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMD > AiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata > =4E7a5ZxSvttBiMxl1VTtolmTvS4T5VGZHA00CsbPR2c%3D&reserved=0 > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions To > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only > to the sender! > Edit mailing list options or unsubscribe: > >

1 0

Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST)
by Brandon Armstead 29 Dec '22

29 Dec '22

Outbound calls from my SBC into Teams (Polycom -> SBC -> Teams) always result in a 400 BAD REQUEST. Example invite below: https://pastebin.com/F1G1Ce59 I've taken care to make sure numbers are all E.164 format in From/To/Contact. I've also taken care to make sure that FQDN is used in Contact and Record-Route header. I've tried many different variations and have followed the SIP information here: https://learn.microsoft.com/en-us/microsoftteams /direct-routing-protocols-sip I've also tried several different clients (Bria, Polycom CCX 600, Grandstream, etc) to see if maybe it was something in the SDP or otherwise causing an issue. SIP Transport is TLS, RTP is SRTP I might also add that OPTION pings are active and Direct Routing Endpoint is active, so this is successful. I also am able to receive calls FROM teams to my IP phone(s) without issue. It is only when I try and call INTO teams (INVITE -> Microsoft Teams) that I always receive a 400 BAD REQUEST to my INVITE. Any help is appreciated, thank you! - Brandon

5 9

Remove all Via Header or create custome Via Header
by Benoit Panizzon 27 Dec '22

27 Dec '22

Hi out there! While experimenting with the listen and adverize 'hostname' option I came across a voice switch, which I suppose fails if the Via header is not an IPv4 address. There is a record_route_advertised_address(address) in the RR module, to set a customer Record-Route Header. Is there something similar for the Via header so I could try if having an IP address instead of a hostname in the top-most Via would solve the issue? Or is there a way to remove all via header? Remove_hf("Via") in the branch route trigger garbles the From: HF. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 4

Proxy edge
by Giovanni Iamonte 26 Dec '22

26 Dec '22

Dear List We have configured kamailio to work as a proxy edge and register. As a proxy edge cfg file we have used the example *edge_websocket.cfg.** * We are developing two different client the first one based on *PJSIP* and the other one based *JsSIP*. Client PJSIP (no websockets) everything work perfectly, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, INVITE Client JsSIP (with websockets) The REGISTER, SUBSCRIBE works, the NOTIFY does not work. In more in details we have: Browser Proxy Edge Register Client JsSIP Websocket | | | | | | | ----------Register-----------> | -----------Register----------> | | <----------200 OK ----------- | <----------200 OK ----------- | | | | | ----------Subscribe----------> | -----------Subscribe---------> | | <----------200 OK ----------- | <----------200 OK ----------- | | | | | X<--- | <----------NOTIFY ----------- | | | | As you can see the NOTIFY start from Register and reach the proxy, beloe the Register log file and the sip messages. Could somebody put me in the right direction. Thanks Bye *Log File* Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/tcp_read.c:1533]: tcp_read_req(): content-length=0 Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 2 (reply(status)) flags 1 Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:689]: parse_msg(): SIP Reply (status): Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:690]: parse_msg(): version: <SIP/2.0> Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:692]: parse_msg(): status: <478> Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/msg_parser.c:694]: parse_msg(): reason: <Unresolvable destination (478/SL)> Dec 20 10:51:58 kr-server /usr/sbin/kamailio[87875]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 *Sip Messages* *REGISTER* sip:sip-register.quintetto.it SIP/2.0 Via: SIP/2.0/WSS 13569868356.invalid;branch=z9hG4bK7825787 Max-Forwards: 69 To: <sip:quintetto00@sip-register.quintetto.it> From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=2tnuk9btmp Call-ID: unrnseei8924hf67khoi3g CSeq: 2 REGISTER Authorization: Digest algorithm=MD5, username="quintetto00", realm="sip-register.quintetto.it", nonce="Y6GUq2Ohk3//HTza7ZYMfd85XMdavgnc", uri="sip:sip-register.quintetto.it", response="c9cc56d2d5e36b0a68873c1d5258d062" Contact: <sip:quintetto00@13569868356.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:64e71c57-fd55-4868-be2b-a4a3fa546ac8>";expires=3600 Expires: 3600 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY,SUBSCRIBE,PUBLISH Supported: path,gruu,outbound User-Agent: JsSIP 3.10.0 Content-Length: 0 SIP/2.0 *200 OK* Via: SIP/2.0/WSS 13569868356.invalid;rport=37566;received=192.168.100.192;branch=z9hG4bK7825787 To: <sip:quintetto00@sip-register.quintetto.it>;tag=2b789d0fa67ad6c2da65f0cce35dd60b.71a84df9 From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=2tnuk9btmp Call-ID: unrnseei8924hf67khoi3g CSeq: 2 REGISTER Contact: <sip:quintetto00@13569868356.invalid;transport=ws>;expires=3600;+sip.instance="<urn:uuid:64e71c57-fd55-4868-be2b-a4a3fa546ac8>";reg-id=1 Server: kamailio (5.6.2 (x86_64/linux)) Content-Length: 0 *SUBSCRIBE* sip:quintetto03@sip-register.quintetto.it SIP/2.0 Via: SIP/2.0/WSS 13569868356.invalid;branch=z9hG4bK3648599 Max-Forwards: 69 To: <sip:quintetto03@sip-register.quintetto.it> From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=o261bh2uh7 Call-ID: 4beh90hur8hdhptnbjt8 CSeq: 5813 SUBSCRIBE Proxy-Authorization: Digest algorithm=MD5, username="quintetto00", realm="sip-register.quintetto.it", nonce="Y6GU+mOhk86y3fnrMzoKgiVAdkaTvBId", uri="sip:quintetto03@sip-register.quintetto.it", response="04ca6d83661ac338810baa95ffaa568b" Event: presence Expires: 3600 Accept: application/pidf+xml, application/xpidf+xml Contact: <sip:quintetto00@13569868356.invalid;transport=ws> Content-Type: application/pidf+xml Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY,SUBSCRIBE,PUBLISH Supported: outbound User-Agent: JsSIP 3.10.0 Content-Length: 35 SIP/2.0 *200 OK* Record-Route: <sip:192.168.100.24:15006;transport=tcp;r2=on;lr> Record-Route: <sip:192.168.100.24:16005;transport=ws;r2=on;lr> Via: SIP/2.0/WSS 13569868356.invalid;rport=37566;received=192.168.100.192;branch=z9hG4bK3648599 To: <sip:quintetto03@sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec From: "quintetto00" <sip:quintetto00@sip-register.quintetto.it>;tag=o261bh2uh7 Call-ID: 4beh90hur8hdhptnbjt8 CSeq: 5813 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.100.23:15006;transport=tcp> Server: kamailio (5.6.2 (x86_64/linux)) Content-Length: 0 *NOTIFY*.sip:quintetto00@13569868356.invalid;transport=ws.SIP/2.0 Via:.SIP/2.0/TCP.1 92.168.100.23:15 006;branch=z9hG4bK28ad.a5d3bdf6000000000000000000000000.0 To: <sip:quintetto00@sip-register.quintetto.it>;tag=o261bh2uh7 From: <sip:quintetto03@sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec CSeq : 2 NOTIFY Call-ID:.4beh90hur8h dhptnbjt8 Route : <sip:192.168.1 00.24:15006;transport=tcp;r2=on;lr>,.<sip:192.16 8.100.24:16005;transport=ws;r2=on;lr> Content-Length:.0 User-Agent: kamailio.(5.6.2.(x86_64/li nux)) Max-Forwards:.70 Event: presence Contact:.<sip:192.168.100.23:15006;transport=tcp> Subscription-State: active;expires= 3600 SIP/2.0.478.Unresolvable.destination.(478/SL) Via: SIP/2.0/TC P.192.168.100.23 :15006;branch=z9hG4bK28ad.a5d3bdf6000000000000000000000000.0;rport=38019 To:.<sip:quintetto00@s ip-register.quintetto.it>;tag=o261bh2uh7 From: <sip:quintetto03 @sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec CSeq: 2 NOTIFY Call-ID: 4beh90hur8hdhptnbjt8 Server :.kamailio.(5.6.2.(x86_64/linux)) Content-Length: 0 SIP/2.0.478.Unresolvable.destination.(478/TM) Via: SIP/2.0/TC P.192.168.100.23:15006;branch=z9hG4bK28ad.a5d3bdf6000000000000000000000000.0;rport=38019 To: <sip:quintetto00@s ip-register.quintetto.it>;tag=o261bh2uh7 From: <sip:quintetto03 @sip-register.quintetto.it>;tag=c906ce7d8b5c7cc84a3505ebe369640b-e45527ec CSeq: 2 NOTIFY Call-ID: 4beh90hur8hdhptnbjt8 Server : kamailio.(5.6.2.(x86_64/linux)) Content-Length: 0 -- ------------------------------------------------------------------------ Ing. Giovanni Iamonte Developments and technologies area Quintetto Srl Via Monte Navale, 1 10015 - Ivrea (TO)  mobile: +39 393 9196310  tel: +39 0165 1845290  e-mail: giovanni.iamonte(a)quintetto.it  web: www.quintetto.it

3 4

ACK received instead of Status Code
by Waqar 40 26 Dec '22

26 Dec '22

Dear All, Attached in this email, is the screenshot of Kamailio logs for a call that did not originate. When Kamailio sends the call to the media destination which is an asterisk server, it should start sending status codes 100, 183, and 180 but instead, it sends ACK and the call did not originate. This happens only for some calls and I want to know why this is happening. Also, I want to identify these cases and do some further logic like trying some other media destination. How can I do that? [image: image.png] Regards, Vicky

1 0

Re: Call drops after 1 minute
by James Browne 23 Dec '22

23 Dec '22

Hi Christian I read your trace. There's no ACK after the SIP-200 response. It's cumbersome to read the trace because the client and server are at the same IP address, but I did it anyway. There's no ACK after the SIP-200 response, so it looks like the client (213.52.37.107) either doesn't sent it, or sends it directly to the server (213.52.37.107, the same IP address) without properly using the RR header field in the SIP-200 response. If you can set up a similar test and have a different host set up for the client and server (instead of 213.52.37.107 for both), and if you then trace the traffic again at all hosts, then you may find the rogue ACK request or be able to prove that the client never sent it (which might prove a broken client). When you use the client and server at the same IP address, it's not always straightforward to trace any traffic that might go directly from client to server, because it won't leave any network interface. Also I find strange that your call ends after "about a minute or so". I expect that they should end after about 31.5 seconds. It's almost certainly a SIP problem, and not any RTP/rtpproxy problem, so I recommend that you focus on the SIP. (I wrote this message before noticing that the thread is two weeks old.) James On Wed, 7 Dec 2022 at 08:41, Henning Westerholt <hw(a)gilawa.com> wrote: > > Hello, > > > > it looks that the 200 OK is not properly transmitted, as they seem to be retransmits. > > Maybe you should read a bit about SIP if you have not done it yet. It will help you debugging this problem (and probably others in the future). 😉 > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com > > > > From: Christian B Wiik <cbw(a)itf-as.no> > Sent: Wednesday, December 7, 2022 9:13 AM > To: Henning Westerholt <hw(a)gilawa.com> > Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] Call drops after 1 minute > > > > Link to entire trace: > > > > https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf… > > > > -- > > Regards > > Christian > > > > > > ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt <hw(a)gilawa.com>: > > Hi Christian, > > > > this ACK is the reply to the 407 and not the relevant one for the dialog. > > > > Please have a look to the full SIP dialog. > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com > > > > From: Christian B Wiik <cbw(a)itf-as.no> > Sent: Wednesday, December 7, 2022 8:14 AM > To: Henning Westerholt <hw(a)gilawa.com> > Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] Call drops after 1 minute > > > > Thanks Henning. > > > > These are the first 3 packets filtering on my user. I see the ACK but I'm not able to spot the error. > > > > U 213.52.37.107:50336 -> 10.1.2.10:5060 #1 > INVITE sip:kmm@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9 > 706413f868bdd222cadbed8..Max-Forwards: 70..From: <sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d1 > 4fb4c6..To: <sip:kmm@sip2.itf-as.com>..Contact: <sip:cbwlap@213.52.37.107:35270;ob>..Call-ID: b3dd380f0c1d4e > 0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: <sip:sip2.itf-as.com;lr>..Allow: PRACK, INVITE, ACK, BYE, CAN > CEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS..Supported: replaces, 100rel, timer, norefersu > b..Session-Expires: 1800..Min-SE: 90..User-Agent: MicroSIP/3.21.3..Content-Type: application/sdp..Content-Le > ngth: 345....v=0..o=- 3879388988 3879388988 IN IP4 213.52.37.107..s=pjmedia..b=AS:84..t=0 0..a=X-nat:0..m= > audio 35276 RTP/AVP 8 0 101..c=IN IP4 213.52.37.107..b=TIAS:64000..a=rtcp:35277 IN IP4 213.52.37.107..a=send > recv..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ssrc > :1053777612 cname:28d400de4b7d5918.. > # > U 10.1.2.10:5060 -> 213.52.37.107:50336 #2 > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 213.52.37.107:35270;rport=50336;branch=z9hG4bKPj > 398365dc9706413f868bdd222cadbed8;received=213.52.37.107..From: <sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e > 4531a7a39f45d14fb4c6..To: <sip:kmm@sip2.itf-as.com>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: > b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860 INVITE..Proxy-Authenticate: Digest realm="sip2.itf-as.com", no > nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2 (x86_64/linux))..Content-Length: 0.... > # > U 213.52.37.107:50336 -> 10.1.2.10:5060 #3 > ACK sip:kmm@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706 > 413f868bdd222cadbed8..Max-Forwards: 70..From: <sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d14fb > 4c6..To: <sip:kmm@sip2.itf-as.com>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: b3dd380f0c1d4e0eb > dd7fc223710d938..CSeq: 23860 ACK..Route: <sip:sip2.itf-as.com;lr>..Content-Length: 0.... > > > > -- > > Regards > > Christian > > > > > > ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt <hw(a)gilawa.com>: > > Hello, > > > > as you’ve guessed, this can be a common problem related to the routing of the ACK message. > > > > Have a look e.g. with ngrep or sngrep to the SIP signalisation on the server side and check if everything is correct in the SIP messages. > > > > > > From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Christian B Wiik > Sent: Wednesday, December 7, 2022 7:43 AM > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Call drops after 1 minute > > > > Greetings! > > > > I have a CentOS setup in AWS where all my calls are dropped after about a minute or so. I realize this typically is a NAT problem, but I can't see where my error is. > > Sound is fine both ways. > > > > Kamailio is set with WITH_NAT and I use rtpproxy like this: > > OPTIONS="-l 10.1.2.10 -s udp:127.0.0.1:7722 -d INFO:LOG_LOCAL5 -m 35010 -M 35110 -A 54.171.168.48" > > (10.1.2.10 is the local IP for CentOS) > > > > Tested with MicroSIP and Linphone and tried numerous configurations. It seems the receiving client is not able to verify the call has been set up, and disconnects. MicroSIP has the status "Connecting..." until it disconnects. > > > > All tips appreciated. Will post configuration and logs if needed. > > Kamailio version 5.6.2 from rpm and rtpproxy 2.1.0 compiled from source. > > > > __________________________________________________________ > 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

1 0

Kamailio Lagging or Unreachable in Asterisk Peer
by Waqar 40 23 Dec '22

23 Dec '22

Dear All, I have three Kamailio in my production environment and I have written some code in Kamailio.cfg using SqlOps to insert data into the database on another remote server for every call. I deployed the code in one of the Kamailio and it inserted data perfectly fine into the database but as soon as I deployed it in the other two Kamailios. All three Kamailio's become unreachable or lagging in asterisk. I don't know if it has anything to do with SqlOps or queries being stuck somewhere as I am receiving logs consistently for every call in Kamailios. How can I identify if the queries are being stuck somewhere and that is why Kamailio is unreachable or lagging? Calls are not dependent on that database, this I know because even if I stop the mysql service calls will still be dialed. If it is due to the SqlOps, is there any other way to save information on every call without Kamailio being stuck? Regards VIcky

2 1

Dockerized VoIP Stack
by Terrance Devor 22 Dec '22

22 Dec '22

Hello Everyone, Does anyone have experience in deploying Kamailio, Asterisk and RTPProxy to a dockerized environment with K8s? In short, we are looking to deploy the following applications to our Kubernetes environment to support a SIP based PBX for our company: - Kamailio (Sip Proxy with DID, Outbound, edge routing and LB capabilities) - Asterisk Realtime (PBX, IVR) - RTPProxy (Media Relay) Has anyone been successful, or failed, in doing this? I would love to hear from you. Terrance

8 21

Re: sr-users Digest, Vol 211, Issue 47
by Luke Escudé 22 Dec '22

22 Dec '22

It's not easy and takes a long time to perfect, but our entire VoIP infrastructure is Dockerized, and composed of Asterisk, MySQL, Redis, NodeJS and Kamailio+RTPEngine (not rtpproxy, since rtpproxy isn't maintained anymore). No docker-level port forwarding required anywhere, since the entire system is designed to be volatile (see below). We have over 500 Asterisk docker containers running, supporting many thousands of users across the world at this point, and it works quite well, so it can be done. The concept behind the Asterisk containers is designing everything to be completely volatile: Outages happen, even in datacenters, so assume a container will reboot at some point on some random node at some random datacenter location. You don't know Asterisk's private IP, you don't even know where it might pop back up in the cluster. If you can ensure service continuity in that scenario, then you've built a solid setup. Our uptime has been 100% for over 4 years now (this is 3rd-party monitored and validated, in addition to our internal monitoring systems) so I assure you it is very possible. I also HIGHLY recommend not using a cloud provider for any of the core infrastructure. We tried this, with AWS and GCP, and the voip quality was... meh. The cloud outages were frequent, although customers generally don't notice them when you're using lightweight containers to run everything. So we are fully operating on our own hardware across multiple colo locations, and only using cloud for file storage (eFaxes, voicemails, greetings/recordings, MMS content, phone LCD background images for provisioning, etc). I am unable to help with your project, but I can at least provide the above pointers/experiences in case they get you in the right direction. An alternative to building your own infrastructure might be reselling (white label) an existing service provider as well, in case the decision to build your own infrastructure is still up in the air. Luke Escudé 972.600.1150 support(a)primevox.net Schedule a meeting!<https://outlook.office365.com/owa/calendar/PrimeVOXCommunications@primevox.…> View the PrimeVOX R&D Roadmap here<https://trello.com/b/pzpoyn8m/primevox-engineering> View the PrimeVOX Status Page here<https://status.primevox.net/> ________________________________ From: sr-users-request(a)lists.kamailio.org <sr-users-request(a)lists.kamailio.org> Sent: Thursday, December 22, 2022 9:44 AM To: sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org> Subject: sr-users Digest, Vol 211, Issue 47 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe 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. Re: Dockerized VoIP Stack (Terrance Devor) 2. Re: Dockerized VoIP Stack (Terrance Devor) 3. Re: Dockerized VoIP Stack (Terrance Devor) ---------------------------------------------------------------------- Message: 1 Date: Thu, 22 Dec 2022 09:31:33 -0500 From: Terrance Devor <ter.devor(a)gmail.com> Subject: [SR-Users] Re: Dockerized VoIP Stack To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Message-ID: <CA+w+2f8CtdwkiDtz+Md-pZz5xA4-59gv768b_vq4bkdjf2V3hw(a)mail.gmail.com> Content-Type: multipart/alternative; boundary="000000000000398ece05f06b87af" I did see the CyCore project and tried to connect with Laurel Lauson and Sean McCord on LinkedIn On Wed, Dec 21, 2022 at 9:18 PM Fred Posner <fred(a)palner.com> wrote: > On Dec 21, 2022, at 8:41 PM, Terrance Devor <ter.devor(a)gmail.com> wrote: > > > Does anyone have experience in deploying Kamailio, Asterisk and RTPProxy > to a dockerized environment with K8s? > > > Check out some past Kamailio World and Astricon presentations, including > some from Sean McCord. He even has a GitHub demo: > > * https://github.com/CyCoreSystems/asterisk-k8s-demo > > -- Fred > The Palner Group, Inc. > +1 (212) 937-7844 > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: >

2 1

Which KEMI-language has the lowest runtime overhead?
by Дилян Палаузов 22 Dec '22

22 Dec '22

Hello, KEMI can be programmed in Java, JavaScript, Lua(sr), Squirrel, Mono, Perl, Python(2,3,3s), Ruby. Which language binding has the lowest runtime overhead in CPU-terms? How about lowest overhead in terms of RAM? Kind regards Дилян

4 3

IMS integration with AS
by Oleg Belousov 22 Dec '22

22 Dec '22

Hi. Can you please advise if existing ims implementation supports integration with ims AS (Application Server), in particular for the third party registration: - upon UE registration is completed, S-CSCF constructs Register message (which keeps all details of subscriber register message) and submits it to AS server as regular SIP-message (TS 24.229 5.4.1.7 Notification of Application Servers about registration status). S-CSCF obtains AS address from SAA (Server Assignment Answer) diameter message, received from HSS upon successful registration. -- obelousov.tel

2 2

How to make a branch trigger persistent (still being active after failure route is triggered)?
by Benoît Panizzon 20 Dec '22

20 Dec '22

Hi List Some time ago, during my learning curve with kamailio, I learned that a branch route is where you make modifications to callerId's. So I extensively re-engineered the config to move as much as possible of callerid modification into branch routes. But now I discovered, when a call is dispatched via dispatch group with multiple destinations and one destination fails, therefore the failure route is triggered, the branch route is not being called on the second attempt, thus callerid being in the wrong format. Example: dispatcher.lst As an example, consider having dispatcher group 2001 for national traffic over two dedicated links and Dispatcher group 2002 for international traffic. 2001 sip:192.168.1.1:5060 0 0 name="Carrier1" 2001 sip:192.168.2.1:5060 0 0 name="Carrier1" 2002 sip:192.168.1.3:5060 0 0 name="Carrier2" request_route { # We determine where we want to send the call # and choose dispatch group 2001, the 'national' one. # But there are are other dispatch groups and branch triggers # set this way and routed to DISPATCHCALL $avp(dispgroup) = 2001; t_on_branch("BR_TO_NATIONAL"); route(DISPATCHCALL); } route[DISPATCHCALL] { t_set_fr(120000,1500); # Fast Failover t_set_max_lifetime(300000,0); t_set_retr(200,800); if (ds_select_dst("$avp(dispgroup)", "6")) { if (ds_is_from_list(-1,0,"$du")) { t_on_failure("DISPATCH_FAILURE"); route(RELAY); } } } branch_route[BR_TO_NATIONAL] { # Here I make sure, all CallerId and Header are correct for the # national Carrier } failure_route[DISPATCH_FAILURE] { if (t_check_status("(5[0-9][0-9])") or (t_branch_timeout() and !t_branch_replied())) { # Try next DS. if (ds_next_dst()) { xlog("L_INFO", " --> retargeting request\n"); t_on_failure("DISPATCH_FAILURE"); route(RELAY); #### RIGHT HERE! exit; } } # We fail xlog("L_ERROR", "Call Failed\n"); } My issue is not, that if the call is routed after the first path failed (marked with #### RIGHT HERE!) then the appropriate branch route is not triggered again and the headers are wrong. But I can not set t_on_branch within route[DISPATCHCALL] or route[DISPATCH_FAILURE] because this is called with a different pre-set branch trigger, depending on the target dispatch group. Is there a way to make a branch trigger more persistent? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 7

Topoh module not decoding the BYE received from the callee
by Michel Pelletier 20 Dec '22

20 Dec '22

Hello, I am using the topoh module and it works fine except for the BYE requests coming from the callee (the person receiving the call). When a BYE is received from the destination of the call, then Kamailio relays it to the dummy IP 127.0.0.8. This is exemplified in the topoh:msg-outgoing and topoh:msg-sending events where the $sndto(ip) is set to 127.0.0.8. I am using all default modparams with topoh. The only modparam I have is for setting the event_callback. I am using Kamailio 5.6.2 with Kemi using python3. Michel Pelletier

2 3

dialog_vars accumulating in database?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi List I noticed that the database is accumulating entries on dialog_vars. I don't know when and why this is happening. Are there any special measures that have to be taken like making sure to unset variables after using them to prevent this? Or is this something which is supposed to happen when kamailio is restarted? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 5

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users