- sr-users - kamailio.org

wrong char / bad PVE format
by Nathan Stratton 09 Jan '23

09 Jan '23

Trying to rename rtpengine files with: exec_msg("mv /var/spool/rtpengine/rtpengine-recording/$ci* /var/spool/rtpengine/rtpengine-recording/$ci.wav"); But get this error: Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/pvapi.c:924]: pv_parse_spec2(): error searching pvar "ci.wav" Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [v/118] in [$ci.wav] at [6 (0)] Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/sr_module.c:1157]: fix_param(): bad PVE format: "mv /var/spool/rtpengine/rtpengine-recording/$ci* /var/spool/rtpengine/rtpengine-recording/$ci.wav" Jan 06 13:59:19 proxy1 /usr/sbin/kamailio[2929]: ERROR: <core> [core/mod_fix.c:564]: fixup_spve_null(): Cannot convert function parameter 1 to space Any ideas? ><> Nathan Stratton Founder, CTO Vocinity, Inc. <https://www.vocinity.com/gabbi>

3 2

RABBITMQ and AWS
by Fred Posner 06 Jan '23

06 Jan '23

Has anyone had any success connecting to RabbitMQ via a TLS 1.2 connection (such as the AWS MQ product)? Fred Posner https://pgpx.io

3 3

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

How to unset_dlg_profile in request_route or react more quickly to CANCEL?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi List Testing failure situations, I discovered unset_dlg_profile can not be used in request_route: I count the channels per customer in a dlg_profile to know when they are busy. Residential POTS customer have 1 channel. Now this situation (Trying to mimik POTS behavior) Kamailio <=> CPE of 'John-Doe' => INVITE set_dlg_profile of John-Doe is 0: Not busy set_dlg_profile of John-Doe +1 <= 100 trying <= 180 ringing == LINK to CPE DISRUPTED == X<= 200 OK (not reaching kamailio) CPE is sending 200 a couple of times and fails. Caller is still hearing ringing tone as it never got 200 OK. The caller does not want to wait longer and hangs up =>X CANCEL (NOT reaching CPE) This is the moment, on which I would like to unset_dlg_profile of John-Doe to mark his channel available again. But the CPE is never going to acknowledge this CANCEL with 487. Unfortunately: unset_dlg_profile(): dialog delete profile cannot be used in request route Only when kamailio has sent CANCEL a couple of times and the failure_route for this CANCEL is triggered, I can remove the call from the dlg_profile of that customer within failure_route. If I am unlucky, in the meantime other calls get 'busy' instead of being re-routed to the configured backup number of that customer. Agreed, after a couple of seconds, the call goes to the failure route and the dialogue is ending thus calls work as expected. But wouldn't it be nice to be able to unset the dlg_profile upon receiving the CANCEL instead on when the CANCEL transaction is successful? 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 2

DNS Host-Names in Via and Record-Route for more redundancy?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi Folks We use two 'core' instances and two 'registrar' instances. The pairs sync their information (Dialogues + Locations) via DMQ. But I noticed that they record the Via and Record-Routes by IP addresses. So I guess if one fails, signaling would not automatically switch to the remaining host as the IP of the failed one is in the header, right? So, is there a way to put a DNS resolvable host-name in the Via and Record-Route header, so that DNS round-robin would make sure requests reach the available server if one of the pairs fails? 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 6

Multiple websocket connections with the same port
by Benjamín Visón 19 Dec '22

19 Dec '22

Hello! I'm examining the WebSocket module and I am wondering how it handles multiple connections from the same port. For context, imagine the following scenario: UA1 ==> Load Balancer:41914 ==> Kamailio (WS) UA2 ==> Load Balancer:41914 ==> Kamailio (WS) Now, imagine Kamailio perceives UA1 from port 41914. What happens if that UA1 suddenly drops and another UA2 comes in from the same port? (before the connection closed event is fired for UA1) I understand by looking at the source code that `wsconn_add` is called and that this new connection is added to the list. But what happens with the previous UA entry? does it gets overwritten or does it stays there? The reason for my question is: I am perceiving the above scenario like this: Let's say UA1 is *sip:d3315cf8@9a431b1f90e5.invalid;transport=ws SIP/2.0* and UA2 is *sip:asd08a08@555343439666.invalid;transport=ws. *If UA2 comes in and takes over the port that UA1 was using, and I send an INVITE to UA1 UA2 ends up receiving the invite (obviously because it now has that port allocated). So what I am thinking is that both *d3315cf8(a)9a431b1f90e5.invalid *and *asd08a08(a)555343439666.invalid *are mapped to the same port for a short period of time. Thoughts? Saludos, [image: Twitter] <https://twitter.com/benjaminvison>[image: Instagram] <https://instagram.com/bvisonl/> Benjamín Visón / IT Engineer / Software Developer bvisonl(a)gmail.com / (829)-664-5163 https://www.linkedin.com/in/bvisonl/

3 2

auth_challenge() and async
by Alex Balashov 19 Dec '22

19 Dec '22

Hi, Is there any good solution for the scenario of an auth_challenge() with two separate transaction suspensions? To clarify (simplified): request_route { ... t_newtran(); if(!is_present_hf("Authorization") && !is_present_hf("Proxy-Authorization")) { auth_challenge("$fd", "1"); exit; } # TM suspend/continue t_newtran(); async credentials query("RESUME_AUTH"); } route[RESUME_AUTH] { # Credentials received into PVs. if(!pv_auth_check("...")) { auth_challenge("$fd", "1"); exit; } # Create transaction shell if not exists already from auth query. if(!t_lookup_request()) t_newtran(); # TM suspend/continue async route query("RESUME_ROUTING"); } route[RESUME_ROUTING] { # Unmarshal etc. t_relay(); # etc. } What happens here is that the first auth_challenge() results in a retransmission of its 407 challenge without absorbing the negative ACK. This is despite the ostensibly stateful behaviour of auth_challenge() without "force_stateless_reply" enabled. In this case, it eclipses the second, unrelated 407 challenge from the subsequent routing query (407 challenge with +1 CSeq): -- 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/

3 17

Dealing with failed SRV peers
by Sebastian Damm 17 Dec '22

17 Dec '22

Hi, we have some Kamailios working as outboundproxy. So they get requests from internal systems and send them to different providers. From time to time, one provider returns a server as primary resource which is currently unavailable. I guess if the internal systems connected directly to the target, they would remember the failed server and remember to always use the server with second priority at least until DNS refresh time. In our setup, since every request is "new" for Kamailio, it doesn't remember, which host is reachable or not. Example: target: example.com _sip._udp.example.com SRV resolves to: 10 192.0.2.42 5060 20 198.51.100.42 5060 30 203.0.113.42 5060 192.0.2.42 is unavailable. Still, Kamailio uses it for every new request and failover to 198.51.100.42 occurs only after timeouts hit. Is there a best practice for solving this? I have played around with the dst_blocklist settings, but that caused even more trouble because Kamailio started blocking requests to peers that have only one server in the record having a short hickup. Thanks in advance for every input, as this is causing trouble every time we run into such a situation. Regards, Sebastian

4 7

Redundancy
by Jawaid Bazyar 16 Dec '22

16 Dec '22

Hi, I am exploring different redundancy / load-balancing models for a Kamailio cluster. When I say cluster, I mean, a number (N) of Kamailio nodes acting as stateful proxies. Each node is configured the same as the others, and all have access to the same lookup data to make routing decisions. I would appreciate any advice or experience any of you can share on these different models. Overall model: Direct to proxies Redirect servers first, which redirect to proxies Selecting the first node to talk to. Each model could use either type of selection. DNS-based (SRV or NAPTR, client makes call to dns name) Anycast with ECMP (equal-cost multi-path routing) Cluster with a mobile IP and service-down detection (this would just provide 1:1 protection) Have clients make calls through the proxy using a DNS record containing an SRV record for each node (or, alternatively, done with NAPTR). Would rely on the client to switch nodes in the event of a node failure mid-call. (Is that even possible?) Anycast would only work with UDP signaling. Use Anycast to find the first proxy, then use stateful responses to direct client back to same node for subsequent messages in a dialog. So for anyone who has tried any of these methods, I would love to hear the pros and cons.. Thanks in advance! Jawaid

2 4

Must openssl_mutex_shared.so be preloaded?
by Дилян Палаузов 16 Dec '22

16 Dec '22

Hello, The documentation on using openssl_mutex_shared.so is contradicting. https://github.com/kamailio/kamailio/blob/master/src/modules/tls/README#L293 says: > For OpenSSL (libssl) v1.1.x, it is required to preload 'openssl_mutex_shared' library shipped by Kamailio. For more details see 'src/modules/tls/openssl_mutex_shared/README.md'. https://github.com/kamailio/kamailio/blob/master/src/modules/tls/utils/open… says: > IMPORTANT: the workaround of using this preloaded shared library is no longer needed starting with Kamailio v5.3.0-pre1 (git master branch after September 14, 2019). The code of this shared library has been included in the core of Kamailio and the same behaviour is now achieved by default. It is unclear, if openssl_mutex_shared.so must be preloaded. In case the library is not needed anymore, it shall not be compiled and installed. Kind regards Дилян

1 1

Logs suddenly inundated with qm_malloc errors
by George Diamantopoulos 16 Dec '22

16 Dec '22

Hello all, I'm testing out release 5.6.2 on one production kamailio instance, and today our monitoring system triggered an alarm for lack of disk space on /. Indeed, kamailio.log started growing at an alarming rate at some point. At about the same time disk space started being depleted, the following line was frequently logged in the logs (no instances of it before that time): qm_malloc(): qm_malloc(0x7f5ade9c0010, 536) called from rtpengine: bencode.c: __bencode_piece_new(79), module: rtpengine; Free fragment not found! They don't only concern rtpengine, core has produced similar "free fragment not found" errors as well. There's additional multiple "had header field", "bad header" and "invalid header name" errors appearing after that too, but I'm guessing that's only a symptom of the qm_malloc issue? kamailio is started with -m 2048 -M 16 and there's another instance with almost identical traffic and configuration, but running on 5.6.1, which has never exhibited such failures in several months uptime. So I'm wondering, is there anything in 5.6.2 that could explain this? Thanks! Best regards, George

3 4

NAT issues with Kamailio + Asterisk with 2 NICs
by Sergio Charrua 15 Dec '22

15 Dec '22

Hi all! Got a random issue with NAT which I really have no clue how to solve. Setup is as follows: - Client A sends SIP INVITE through Public IP 1.2.3.4, where a Cisco Firewall forwards NATed data to a Kamailio instance (acting as a SIPProxy with dispatcher) listening with internal address 192.168.0.1 on port 5060. RTPEngine is in the same server and address too. - Kamailio will forward the invite to Asterisk NIC 1 (192.168.0.2) which will do some provisioning and accounting stuff and redirect the call (if authorized) to Client B's public IP address (all clients are outside the network, reachable via WWW) using NIC 2 (192.168.1.2) and public address 4.3.2.1. The issue is that from time to time, calls will not have audio. Here is what I have found: - when Client A sends INVITE, Asterisk NIC 1 will reply SIP 183 with SDP IP Address set to the internal IP of NIC 1 (192.168.0.2) - when Asterisk forwards call to Client B public IP address the SDP IP address is 192.168.1.2 (NIC 2) And I think the issue is the SDP address is wrong: when Asterisk forwards call to Client B, the SDP address should be the public IP 4.3.2.1 but it sends 192.168.1.2. And when Asterisk replies to Kamailio the SDP is 192.168.0.1. Correct me if I am wrong but i think the solution should be: - when Kamailio receives a reply from Asterisk to be forwarded to Client A, it should modify the SDP address to public IP 1.2.3.4 - when Asterisk forwards call to Client B, the sip.conf should have the parameter externip set to the public IP 4.3.2.1 Questions: - are my analysis & solution correct? - is there an example of the above setup where I could get some ideas? - how do I set Kamailio to modify the SDP address for me? Any examples? Thanks in advance. *Sérgio Charrua*

2 1

kamailio.org server maintenance - Dec 14, 2022 (tomorrow)
by Daniel-Constantin Mierla 15 Dec '22

15 Dec '22

Hello, not that tomorrow, Dec 14, 2022 starting after 09:00UTC (10:00Berlin), is planned some maintenance work on kamailio.org server. It will be unavailable from time to time, hopefully not for long time. Affected services: - web site - mailing lists - wiki site (alternative use the github repo: https://github.com/kamailio/kamailio-wiki) - html version of modules' documentation (text variant README is every module folder of git clones or on github.org repository) Short notifications will be sent when it starts and ends. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

Failover - how does it work?
by Jawaid Bazyar 15 Dec '22

15 Dec '22

I have the following basic script: request_route{ record_route(); $var(enumres) = enum_query(); switch( $var(enumres) ) { case -1: xlog("enum_query lookup failure - To URI $tU"); sl_reply("503", "Temporary lookup failure"); return(0); case 1: break; # uri replaced } xlog("INVITE ENUM query - To URI $tU"); t_relay(); I have the following ENUM records for a test setup: ;; ANSWER SECTION: x.x.x.1.5.1.8.3.0.3.1.e164.arpa. 60 IN NAPTR 100 10 "U" "E2U+sip" "!^.*$!sip:+13038151xxx@10.20.20.198!" . x.x.x.1.5.1.8.3.0.3.1.e164.arpa. 60 IN NAPTR 110 10 "U" "E2U+sip" "!^.*$!sip:+13038151xxx@10.20.20.75!" . What happens is the following: tcpdump: listening on ens3, link-type EN10MB (Ethernet), capture size 262144 bytes 17:57:19.961041 IP (tos 0x10, ttl 64, id 59648, offset 0, flags [none], proto UDP (17), length 1481) 10.20.20.40.5060 > 10.20.20.198.5060: SIP, length: 1453 INVITE sip:+13038151xxx@10.20.20.198 SIP/2.0 … 17:57:19.961058 IP (tos 0x10, ttl 64, id 24143, offset 0, flags [none], proto UDP (17), length 1480) 10.20.20.40.5060 > 10.20.20.75.5060: SIP, length: 1452 INVITE sip:+13038151xxx@10.20.20.75 SIP/2.0 … These basically go out at the same time. Eventually I get a ICMP Port Unreachable on the .198 one because that’s a dummy I just set up for testing. My question is, with this kind of setup, is there a way to tell Kamailio to respect the NAPTR order value (note, 100, 110 above) and only try the 2nd route if the first fails? Thanks, Jawaid

2 5

testing mailing list
by Daniel-Constantin Mierla 14 Dec '22

14 Dec '22

after mailing list service upgrade - ignore!! -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

None
by None 14 Dec '22

14 Dec '22

ACK messages SER keeps sending the 487 Request cancelled message as if the ACK didn't match. Any suggestion of what it can be? I'm using as the endpoint an ATA186, v2.15, and ser version 0.8.11. Best Regards, Alejandro _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ------=_NextPart_000_0009_01C322CD.41C9F360 Content-Type: text/plain; name="remote.ata.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="remote.ata.txt" interface: eth0 (200.75.73.0/255.255.255.0) filter: ip and ( port 5060 ) # U 200.75.73.22:5060 -> 200.75.91.137:5060 INVITE sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..M ax-Forwards: 10..Record-Route: <sip:2001@200.75.73.22;branch=0>..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171e2f6.0..Via: SIP/2.0/U DP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=9 28411572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 329755117 1@192.168.0.15..CSeq: 1 INVITE..Contact: <sip:1010@200.75.73.22:5061;u ser=phone;transport=udp>..User-Agent: Cisco ATA 186 v2.15 ata18x (030 122a)..Expires: 300..Content-Length: 245..Content-Type: application/sd p....v=0..o=1010 8066 8066 IN IP4 192.168.0.15..s=ATA186 Call..c=IN IP 4 200.75.73.22..t=0 0..m=audio 20002 RTP/AVP 4 8 0 101..a=rtpmap:4 G72 3/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c 171e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobr eip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=p hone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE ..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258. c171e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsob reip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user= phone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq: 1 INVIT E..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 CANCEL sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..M ax-Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171 e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip .biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phon e>..Call-ID: 3297551171@192.168.0.15..CSeq: 1 CANCEL..User-Agent: Cisc o ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171e 2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip. biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone >;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq: 1 CANCEL..Se rver: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP 200.75.73.22;branch=z9 hG4bK6258.c171e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1 010(a)vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip .biz;user=phone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSe q: 1 INVITE..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Le ngth: 0.... # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171e2f6.0..From: <sip:10 10(a)vozsobreip.biz;user=phone>;tag=928411572..Call-ID: 3297551171(a)192.1 68.0.15..To: <sip:2001@vozsobreip.biz;user=phone>;tag=3742565922..CSeq : 1 ACK..User-Agent: Sip EXpress router(0.8.10 (i386/linux))..Content- Length: 0.... # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . exit ------=_NextPart_000_0009_01C322CD.41C9F360 Content-Type: text/plain; name="local.ata.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="local.ata.txt" interface: eth1 (192.168.0.0/255.255.255.0) filter: ip and ( port 5060 ) # U 192.168.0.15:5061 -> 200.75.73.22:5060 INVITE sip:2001@vozsobreip.biz;user=phone SIP/2.0..Via: SIP/2.0/UDP 19 2.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411 572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 3297551171@192 .168.0.15..CSeq: 1 INVITE..Contact: <sip:1010@200.75.73.22:5061;user=p hone;transport=udp>..User-Agent: Cisco ATA 186 v2.15 ata18x (030122a) ..Expires: 300..Content-Length: 245..Content-Type: application/sdp.... v=0..o=1010 8066 8066 IN IP4 192.168.0.15..s=ATA186 Call..c=IN IP4 200 .75.73.22..t=0 0..m=audio 20002 RTP/AVP 4 8 0 101..a=rtpmap:4 G723/800 0/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 tele phone-event/8000..a=fmtp:101 0-15.. # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 1 92.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=92841 1572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 3297551171@19 2.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress router (0.8.10 (i386/l inux))..Content-Length: 0..Warning: 392 200.75.73.22:5060 "Noisy feedb ack tells: pid=32747 req_src_ip=192.168.0.15 in_uri=sip:2001@vozsobrei p.biz;user=phone out_uri=sip:2001@200.75.91.137:5060;user=phone;transp ort=udp via_cnt==1".... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:10 10(a)vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip. biz;user=phone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq : 1 INVITE..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Len gth: 0.... # U 192.168.0.15:5061 -> 200.75.73.22:5060 CANCEL sip:2001@vozsobreip.biz;user=phone SIP/2.0..Via: SIP/2.0/UDP 19 2.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411 572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 3297551171@192 .168.0.15..CSeq: 1 CANCEL..User-Agent: Cisco ATA 186 v2.15 ata18x (03 0122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 200 cancelling..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip :1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobre ip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 CANCEL..Server: Sip EXpress router ( 0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73.22:506 0 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_uri=sip: 2001(a)vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5060;use r=phone;transport=udp via_cnt==1".... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... exit ------=_NextPart_000_0009_01C322CD.41C9F360--

1 0

None
by None 14 Dec '22

14 Dec '22

ACK messages SER keeps sending the 487 Request cancelled message as if the ACK didn't match. Any suggestion of what it can be? I'm using as the endpoint an ATA186, v2.15, and ser version 0.8.11. Best Regards, Alejandro

1 0

None
by None 14 Dec '22

14 Dec '22

/bin/sh -c 'cd /usr/local/mysql ; ./bin/safe_mysqld --user=mysql &' Gary Goodloff KALG Communications, LTD T: 402-898-4152 C: 402-208-5183 F: 413-294-3404 -----Original Message----- From: serusers-admin(a)iptel.org [mailto:serusers-admin@lists.iptel.org] On Behalf Of Todd Sent: Wednesday, May 21, 2003 5:05 PM To: Sip Questions List Subject: [Serusers] Mysql Connection Errors Ok..I have setup SER according to the guide on the iptel site. Here is a copy of what's in my ser.cfg _________________________________________ # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//usr/lib/ser/modules/mysql.so" loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//usr/lib/ser/modules/auth.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present rewriteFromRoute(); # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~"computergeeks.hopto.org") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("computergeeks.hopto.org", "subscriber")) { www_challenge("computergeeks.hopto.org", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } _________________________________________________ The modules are all where they say they are but when I try to start SER I get this in my messages log file May 21 18:58:45 computergeeks ser: ser startup succeeded May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: connect_db(): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: db_init(): Error while trying to connect database May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: mod_init(): Error while connecting database May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: init_modules(): Error while initializing module usrloc Can someone please tell me what I'm doing wrong? I've searched all over the net for information on this...I'm sure it's something simple..but I'm overlooking it. If I disable the Mysql integration is works perfect, so I'm thinking it has something to do with the mysql.so module?!? Thanks Todd ------=_NextPart_000_00B1_01C31FBD.C81364A0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN"> <html xmlns:o=3D"urn:schemas-microsoft-com:office:office" = xmlns:w=3D"urn:schemas-microsoft-com:office:word" = xmlns=3D"http://www.w3.org/TR/REC-html40"> <head> <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; = charset=3Dus-ascii"> <meta name=3DProgId content=3DWord.Document> <meta name=3DGenerator content=3D"Microsoft Word 10"> <meta name=3DOriginator content=3D"Microsoft Word 10"> <link rel=3DFile-List href=3D"cid:filelist.xml@01C31FBD.C730CCD0">  <style>  </style>  </head> <body lang=3DEN-US link=3Dblue vlink=3Dpurple = style=3D'tab-interval:.5in'> <div class=3DSection1> Those SQL = Errors:<o:p></o:p> <o:p> </o:p> May 21 18:58:45 computergeeks ser: ser startup succeeded May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: connect_db(): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: db_init(): Error while trying to connect = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: mod_init(): Error while connecting = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: init_modules(): Error while initializing = module usrloc<o:p></o:p> <o:p> </o:p><= /p> This means that the mysql daemon is either not running or is not running on the default = port.<o:p></o:p> <o:p> </o:p><= /p> On freebsd – this is how I start my sql daemon:<o:p></o:p> From /etc/rc.local<o:p></o:p> /bin/sh = -c 'cd /usr/local/mysql ; ./bin/safe_mysqld = --user=3Dmysql &'<o:p></o:p> <o:p> </o:p> <div> Gary Goodloff KALG Communications, LTD T: 402-898-4152 C: 402-208-5183 F: 413-294-3404 <o:p></o:p> </div> -----Original = Message----- From: = serusers-admin(a)lists.iptel.org [mailto:serusers-admin@lists.iptel.org] On = Behalf Of Todd Sent: Wednesday, May 21, = 2003 5:05 PM To: Sip Questions = List Subject: [Serusers] Mysql Connection Errors <o:p> </o:p> Ok..I have setup SER according to the guide on the iptel site. Here is a copy of what's in my ser.cfg _________________________________________ # # $Id: ser.cfg,v 1.12 = 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start = config script # # ----------- global = configuration parameters ------------------------ debug=3D3       &= nbsp;  # debug level (cmd line: -dddddddddd) fork=3Dyes = log_stderror=3Dno # (cmd = line: -E) check_via=3Dno # (cmd. = line: -v) dns=3Dno       &n= bsp;   # (cmd. line: -r) rev_dns=3Dno      # (cmd. line: -R) port=3D5060 = children=3D4 = fifo=3D"/tmp/ser_fifo" # ------------------ = module loading ---------------------------------- # Uncomment this if you = want to use SQL database loadmodule "//usr/lib/ser/modules/mysql.so" loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" # Uncomment this if you = want digest authentication # mysql.so must be loaded = ! loadmodule "//usr/lib/ser/modules/auth.so" # ----------------- = setting module-specific parameters --------------- # -- usrloc params = -- #modparam("usrloc", "db_mode",   0) # Uncomment this if you = want to use SQL database # for persistent storage = and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params = -- # Uncomment if you are = using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), = # uncomment also the = following parameter) # modparam("auth", "password_column", "password") # = -------------------------  request routing logic ------------------- # main routing = logic route{ # initial sanity checks -- = messages with # max_forwars=3D=3D0, or = excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) = { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if = pre-loaded route headers present rewriteFromRoute(); # if the request is for = other domain use UsrLoc # (in case, it does not = work, use the following command # with proper names and = addresses in it) if (uri=3D~"computergeeks.hopto.org") { if = (method=3D=3D"REGISTER") { # Uncomment this if you = want to use digest authentication if (!www_authorize("computergeeks.hopto.org", = "subscriber")) { www_challenge("computergeeks.hopto.org", "0"); break; }; save("location"); break; }; # native SIP destinations = are handled using our USRLOC DB if = (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri = now if (!t_relay()) = { sl_reply_error(); }; } _________________________________________________ The modules are all where they say they are but when I try to start SER = I get this in my messages log file May 21 18:58:45 = computergeeks ser: ser startup succeeded May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: connect_db(): Can't connect to local MySQL server = through socket '/var/lib/mysql/mysql.sock' (2) May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: db_init(): Error while trying to connect = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: mod_init(): Error while connecting = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: init_modules(): Error while initializing module = usrloc Can someone please tell me what I'm doing wrong?  I've searched all = over the net for information on this...I'm sure it's something simple..but = I'm overlooking it. If I disable the Mysql integration is works perfect, so I'm thinking it = has something to do with the mysql.so module?!? Thanks Todd <o:p></o:p> </div> </body> </html> ------=_NextPart_000_00B1_01C31FBD.C81364A0--

1 0

None
by None 14 Dec '22

14 Dec '22

stream, for such cases, is it possible? Any pointers would be great. -- Michael Vasilenko

1 0

None
by None 14 Dec '22

14 Dec '22

fgets failed: Illegal seek. It happens when 2 end points try to set up a session and one of them cancels it before any end-to-end connection is established. Has anyone found this problem before? (See attached file: cancelsegdump.tar.gz) Jaime --0__=PnXC9kr0ykdwrnb6uT2PR1PNFM05c9HrUcAPjHVL7TjiDE1fTDc3v998 Content-type: application/octet-stream; name="cancelsegdump.tar.gz" Content-Disposition: attachment; filename="cancelsegdump.tar.gz" Content-transfer-encoding: base64 H4sIAAseXj4AA+0da3PaSDKfqcp/mMuHjX0HRKM32Lg2cew912adLezsfkilVIM0Aq2FRPSI7b3a /e3XM5KwhAQIG5LYqykbBOrumenp6ackTOKZ1A3p2Iqns250Ez3bfhOwIKiy/ExIWvEdCyLGyjNB U0VNxqqkSAAvYVl5hoQdjKXU4jAiAULPAt9fOfd15x9pQ8IeliRR3keuTyzHG6Opb8UuRd1XyUH4 anobfnaT127ot2pghC781YSdBY4XJa81MaIp/NWEjcPA9c30rSZOEMBfXVg6dsIoIMHdUU1MEkcT /lJ31vQm8mfz95pY9Iaa/KU2fMT+a0Lb5oz9F6Ftx7OMGQnI1KA3Mz+I+sj2Y89Ch9bIAEx6hBwv o5UsDPq4fMU+1SIdOVMaGCBDNPhC3F30AIOPg69EmYlFnm5eVko0F6iBDDpmZAR+HBU4HQR5ipmM r6VGvhSpZFJeIFa5CdaRdn3/Kp7tiHjoGiGFrwM6c2/zfcCJ9OhjWV+towpspS4p0IumeUKZaqox PD4ygwaBH9xrfGyB2aaMyAiAhX7uFIIdTAOPZKQMk7juXm5ExiwKDoUbWRAUiyjC0d2p/QN098Gx UQ5rSqOJb919Hgzujl8MT346u7g8Gb7IkUL/u8+QsNjT7PyQ2ugF7CYSOb73YslArcCf5YhFtzN6 uHRWfyHm7+QH91f+NN84OWK4Yc7mzKkLt1SecV6ei6w+f3+5h+7HPN3UN2TejqnvP1giBFnTtIVu ZUF+Ufji3I/QKdM8X0NGtrVaWCuK+n3XZTWd7awAWdpBw9nHwNmcsvn99fD87PynPoomFE0JWGZu qRf0kxMiOp1Ft605uOeDcQ+umB2neXpvT958gNOO50Rswn3EA6kWf0Ud/r1DXOdPoL0Ka+4XtCBe jSh4BeFa7MRhAr/YdmzfMKdAxqPXyPSnUwJeyB5zkoBYuD+HpFaZyi9vFQRLZMYu9AskdF0eybKs WKomSpaiYtNSsCjII6rrKu7hVbPgYR50agEzW/zDBiyIprmzl78sYHa73c05EBkxMQ3LHa9mwXoy ExJOVtMAW3BlzAdMky/uZsDtBGK5iOh+I1i1ksev3707e7vQWR8pljgSbAntudQb6PslBGEgpCDr pULs2VhUZL0nyqps2aIiyRbGPcvSpNFIF/X1FCRZ7umarPQI0BJFVZAUSkmvJ5o6ljFRV4lGEnFt zra41g6oQSWYboGEYfpeRMzowaRYLu/BRGw3XifTNagQy3owjXDiX6/hTVWQ9TcLrENqcI2dj7ES 3ftxWYZrXfDG6DLR2zJJ0/XDbQ/zc0yD2y3TDMj1TujaAaU7Iex4IQ22vVoWmOBo28sVzyxyb6IA wcVyb7+P3hIQegIOD+waj5rctvgz6lELhbFpgvNgx+7tKo0aBK0hNf3AQkMWy21gpudpmm+Sovmb adJMpcSWzzy4BybtKnpIRMqIA86i7dNPpGt39Md0h4Pnns7OWB97u6UfAOth5+yOPZnspAZtd8Kz sw649OyMeqIEt9BBSTOxDHrjMzQ+Q+Mz7NYulyP3pGbZSt83cCZY6bLFXjbAsc1ZC/6XY1RXo4hl RNOHl3bmzkc0NUcPJ5eWnIzI3xqtbRBK0ybbGdOCnD2IkHHtRBNj5FtbIemEzJspFpTvS4tZTXCL vdBwLOpF2yCZlFC3R8/2g2sSWIbne8S8qklRYNRuWGq0XF5G+Wr0UuDKCi9ipecUJYxHU9jdXNGz 8CYERWpG4A7QoJ26CW2YjBPQsP25zfLJjtU2Q/oZ2YE/RVkZBm04goS9y5AyzYcKCnMZdMYylG7C OlRzJeIU/Oz89H0fxdaMOxN9dPHeGB7/9ubDKUtIp9rOvUWqokhKipKqxzucKLhlXc5R+whLWNCE InxIo6gAdsC+GiSg7S80cOzbQRXiqo5EVcRyvY4S0FUdlRDRhIQsDU9tG6Rjk+lLGOs1p89AHz6q lesIe5OvYkYe802rspx3n0fxXPuydFnykXXHPvEM+Bw4HQ87gUD0Zq/CNMeWJgLavFIxkCWYuciR tGxcHGoW+CxPkKSkoQvOJ1HLAS8YXsOcOK6F9sT9JFW/CJdsSyOMbRD6Purwsz/intjFqt5Vu1iR 6hHPkr11YIOgDFdKPGNLUmVRo6Rnij3J1LAqE1EaCRLVqKXJ9XpKgos6kOAE1QNkjsxSPycDFPYX KiMr2C0vsrsO8RK7L06GJTnkIgMSBjsDxTNEoqLMVRZnqjoLgjLc5utVp6eFYHAFJF+vOoDF9bo4 +xUNKdisMCpcnJNcKtJH6DC7LuQofzoOHDgH7TB0Zn0CHKR/0K4PjsOYdk2/G18VwIHhIS/mHEJ/ r8SukD87IwHEshNKLICC8blkHA7yBTowW8i3UQKBAkrMCdMMvMQ4UJZT+s0hidfQriC6ABtNmG0K eT3VdoIwQl8ckgN/bYO3nCJNw4U63gxsIAl43dVHQezNy7ChGTizKFyArpqvqKh1Z9wrrzPz2iYW k2LqwqIdXoIf/lGSPx2whRp8ZGv0B3Gm9MeqlfpUUQiGeTDvFH08XI179LyVd9gWxsGdm8PjC/r5 CAaFj6qFaWEOvN8+ywlH4DAaLvXG0WQg5N1C7gkWeLToNPIlTBw65oqd+2mCOOV6QiKHlPcAGcLl BObK5MHLMP97um4JO3nxGsWw4bJyEKN4DKsIigEdJ18BvX523E92EdbEroi7oPc0vY9VTZeODj4P hK4gHKRu4wCLgvC8tdFAQHLMK1bgAvxwL99HGxU/CfnaakBN6nzhsg5r4VLCDXi8meq4+PDm4nh4 9uZkhe6YkuCqWrQaBfKtFYiUVyBLF2q9/li+xhuqj0qBeoT6I4nZGOBLxpuXiF37lgwLotnSdQJP Y5eHzTb/Pre5vLjNq1eq3j5fssrNRufMefo7fUZnS1zFZqN/641esOdLF2r9Pl++xv/QbR5cB04C OKTskC11upU6H4ZniKX70cs56wqevqQIgvCyguVskzu2R6/7CAQMOdZARm00dv0RcbNvBPjmEvke zJ5l7U06EG7stG2kVioh1DyX54n8INURSdINhZQE5oTn3tj1pQNVkQW5Dbx6ffxzYZnSefGUHbXS zQE7pTM8PZZEFSNedyDJdUBTEnGiVXxZHAesSR51cS1TvOmVAV3f3PaR5bPO355fpOp5Ie/zJs1L jYh5xS/YRhJMx0ISY/LCFX3r2XZPfZ0OGvCMyDci4riGbxv8BsKP8qc+Em5kURwRXa7gcCWSkENS 8he68kQZu5ixwMNr62VhorXtB7sKHe0xxRSH+/3aOh8lGMzKYKF4BrRdyNBAi/Gs+PZMxRNTnPzm GcTv3kzkVpTENjocMcUwOUIDdPhnb/KTPPrZ1Gy7q2Msi5rK1uEg5QeubXK+ByOb7KyQujbqJB+Y SDowZD+MBgMmTFgaDLCEfvgBoY+FpPIndu/bwlcFi5KUGHltKys9zouMbRhRGLLOkpsuYI2diuRr ZPBBzbW3kulurskvE/W5gb7GWlUfSfktU5eJDkZcByOXjKiLVFkRRNzTVJQIAqrQGGUyycz4Z2qh vctBqjs0Qd//12ZzVWCuIEw5ClX4vpdx2w/G3VQdoJiEA6ENbyYoMLB2vM4N705oON4XsLmDgtbs HGUNXRomqxwBMui25BgUS7nn5MKFtOuEQwyJ1XXZOz9dNJWrtKy2TDU3+bTG/27yaU0+rcmnPf1t 3uTTmnxak0/7B2z0xxEWPsJt/pXyaXopn6b88/Jpnu91gHQW69VJuJWybQsB4mI2jYWXHIddH5uG Wp4fTbINPT/ZKNhGwTYK9uko2F5JweqNgm0UbKNgGwXbKNhtKFgslDRsr9Gw372GXbz/hRcPUpVA +1o7cgd3VQTk0ZtosOc57n4Zc95xyLSnMSGe5QIpdoLPFN7ZbV+s470oN8E2mu80BLoH5UmvrqIo gpQfRrlarvBquXxXLS+PuUbJJX9L0VJZnUNslsWZo2XPRfvyeQrjCfkTr5J3dv8Tm1b1+Pmj1tgw qdVFo9gGxupCz1YVvY9S84UUYCtMajKl05TvIwnnz6/mosC5iCuuOchuneG3dLjUjkoWu6nBfzNT 1dTgv+saPMa5wjQYz6ddhS/OFj+wDg+Kf16Jh+PNa/GcwNOpxp+d/3Z2uSrwWXHb1PYin1q+MRZr 6+MKoajWx+IW7i5bjbqhPi6vx3pl/PoYK7Isn+j6Gz35vY9TqXeqvVV6x1oHS/9+rHp56+HmY4qo yjfS1QymxFIwBUrzu4umdE36ni6vnV8/muy/jZhyz+JrPbe9bFuYMYVoKEIgXXm/3ZZyfjku+e2m kPfbK5TvPCKEfeCElRHoN7r6OO39jNvnLrpIn/vAHjzEoFnk9/DlqHet8kg1N79WeaRa/5xrlWvH Dj1V6gpUtrUeVprQ4WuFDlLemRYfGjo8At9Eq7fyIQVv01pY+tz25nMEFDIeCLhn2kQxOzKR9A7G ltbpgbXqwNxUZSSbGJujulMR62YONGnnnuoD/dRKaal4QsiSOBMcgjTKxJKq97CERWEbYeZI0euF mYWdIRXCTEZjszATVOrKC77x6iCTo3/NIBMvM1Zf0wzpy8zQEPps7NCTskPy4m5r7FBjhxo7tLgz 5C3aITh+sCXS72uJ8njIdy2UBLxST+Pks49SfvMljx+CMM5gz5Y0QGrYB5YXnDk0o+AHzrivi5xK H4mYdcgCfgkrjKWD1jzk1QWUGpLnLdif82D41Ye3v6Jy7fogncpazfG8dQrD66OVdZkDtmM7mt7r abqiAM6ln2As3wcHdTf589YxxNqds7d9pMuioEuiJohgzVQ40gtVeYCEvQO2IM1zPG99gI47r8cQ lvfR745n+dchGl4ev8JdGONxkoDtvOMJWPAQnrcKD25aV0q8f0pik1Ji7QTCjjyfVen149fnxyfv vnV6/YnmysvMfQR+wGPJlT/py5XunVxXysl1uUmu13t2RYb3Hiw2e+7uZf9u4Ix/3P1iTLxlTlgt fJPx2wVfoOJKqLzbVaWSFqmx38GasV8NKAz9ET1Hw1RrGz0hh6RVGAAqst+KZLzlxymfc5LzHRZI Nq2n1HAx5s/hPRkO3w+TR+8a6aN3bdCMoLGArexBuWdAfQxKIaT06hHiJQ+jDp0xbAQESi71ffIs +IVOfdiZaWyxN7sasyzP/DQwPTvFfT+qm0IBgCn5GffZBwgLsorF/NOw2QPmLXZCBylgx//xYfBM qAaShsGGs58fgWBEFmVml+/wpuSGw6O9Ocb+AEmanH+WNPuRPqZkYLXZv29eMT/VDsh4CmsdFobJ m9BF5wilEps6s6Kso2T8klhCwFUI2goEsQqBZAhyTy1hyBUYsrm8C1w1CVWp7iLjEGMzch3wPfiP NaI8a7hpSeY6Zx2o/m7izYoVkGJtSLUmJJa0Skicg+ysaq137CdDvNTktFChFXJgiymxfl8RQMG+ dh1wXUHRpQ4s/xZVebPJmVb6S0ppxmzhl1LOch/ufs4tBW6xB2cvQXzWtKY1rWlNa1rTmta0pjWt aU1rWtOa1rSmNa1pTWta05rWtG/d/g/YEn7NAKAAAA== --0__=PnXC9kr0ykdwrnb6uT2PR1PNFM05c9HrUcAPjHVL7TjiDE1fTDc3v998-- ------<80256CDA.004DFAC0.00@ruddick> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. ******************************************************************************* ------<80256CDA.004DFAC0.00@ruddick>

1 0

None
by None 14 Dec '22

14 Dec '22

Please give me an example of their interaction. Could I use B2BUA without SER(and without any SIP proxy)? Thanks Igor Vasiliev >

1 0

Re: [SR-Users] sr-users Digest, Vol 211, Issue 8
by Pooja Yadav 13 Dec '22

13 Dec '22

Hi Daniel and Everyone, Thanks for the inputs. We understand that Kamailio is a SIP signalling server irrespective of any core. But we have seen a thread in Open5Gs community where they established the call with Kamailio IMS. Below thread is for your reference. https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ If you have any inputs on this it is most welcome. We just want to check the feasibility of establishing voice/video call with 5G core. Your inputs will be appreciated. Thanks and regards Pooja Yadav > On 09-Dec-2022, at 4:33 PM, sr-users-request(a)lists.kamailio.org wrote: >

2 1

Delays sending over WSS
by Alex Balashov 12 Dec '22

12 Dec '22

Hi, I’ve run into stubbornly persistent problems with packets not reaching WSS clients until some seconds (5-6s or more!) after they were supposedly sent. The symptom is that Kamailio logs the event as sent at 20:00:00, including in the onsend_route, but in actual reality it takes quite a bit longer to send at the OS network stack level. This happens occasionally and quite universally, with no discernible pattern tied to particular endpoints, networks, etc. For a long time, I’ve just assumed this was a receive delay, since Kamailio logged an expectedly prompt relay time. I figured it was an application or browser execution delay and had nothing to do with networking. Based on what we know about the capriciousness of browsers and their internal task scheduling, this seemed rather plausible. However, I’ve since been able to ascertain that this is not the right understanding of the problem, because a parallel and unrelated WS keepalive, going from a different backend to the same application/browser tab/event loop, is consistently replied to in a timely manner. Anyway, I’m curious what else can be done to debug and/or fix this. I did turn up the size of my write buffers — tcp_wq_conn_max and the global one — quite a lot, some time ago and for unrelated reasons. I wonder if this might actually make the problem worse, since it leads to more bytes queued to send to an endpoint on the other side that could, conceivably, not be reading them fast enough. But I also wonder if this is tied up in some low-level TLS or WS parameters unrelated to more general TCP configuration. Thanks much! — Alex — Sent from mobile, apologies for brevity and errors.

2 2

Kamailio SIP client registration issue
by Shiva Shankar Chittallu 09 Dec '22

09 Dec '22

Hi Team, I have installed kamailio sip server 5.5.5 version in ubuntu 20.04 in AWS instance with public IP. When im trying to register an device im getting below error in the syslog. Thanks & Regards Shiva 9100663346

2 1

Does Kamailio supports Voice and Video call support for 5G Core
by Pooja Yadav 09 Dec '22

09 Dec '22

Hi Kamailio community, We just want to check does Kamailio IMS supports the functionality of Voice and Video call support (Vo5G) with 5G core? If yes, please share the document reference. Else, please guide us when it will be available. Any plans in upcoming release to support HTTP2 based SBI interface between P- CSCF and PCF. Your inputs will be highly appreciated. Thanks & Regards, Pooja Yadav

4 4

Next online devel meeting - Thu, Dec 8, 2022, 15:00UTC
by Daniel-Constantin Mierla 08 Dec '22

08 Dec '22

Hello, to summarize what happened at the devel meeting in Dusseldorf a few weeks ago and plan the next steps for the project, we are planning an online development meeting on Thu, Dec 8, 2022, 15:00UTC. As usual, one of the targets 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 v5.7 (or 6.0) 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://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings… Feel free to add there topics that you want to be discussed. The proposed date is Thursday, December 08, 2022, at 15:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev

1 3

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