- sr-users - kamailio.org

sslmode in db_postgres
by Centelles Martin, Joel 20 Jan '23

20 Jan '23

Hello everybody, I'm working on a kamailio instance as a basic registrar service and I thought to use db_postgres to connect a postgres service only allowing SSl connections. I see that that module doesn't provide any parameters to select sslmode and I was wondering if this module actually supports TLS connections. Does anybody know if this is supported and, if so, how to enable it? or should that support be added into db_postgres module code? Thank you very much in advance. BR, Joel Centelles RTC Lead Engineer | Care Communications R&D joel_centellesmartin(a)baxter.com<mailto:joel_centellesmartin@baxter.com> | baxter.com<https://baxter.com/> [cid:9ee6bcbd-9c30-45ec-bb43-8f439f19f724]<https://baxter.com>

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[core/tcp_main.c:617]: _wbufq_add(): (1244714 bytes): write queue full or timeout
by Benoît Panizzon 20 Jan '23

20 Jan '23

Hi I'm trying to get dlg.list on a busy system via jsonrpc and xhttp. I already increased: tcp_conn_wq_max=2048 tcp_wq_max=20480 modparam("ctl", "binrpc_max_body_size", 2048) modparam("ctl", "binrpc_struct_max_body_size", 2048) But it looks like the tcp write queue overflows up at shy over 1MB reply no matter what I change. Any ideas what else to toggle? -- 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 ______________________________________________________

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What are extended flags?
by Дилян Палаузов 20 Jan '23

20 Jan '23

Hello, The corex module manages extended flags with functions the setxflag(), isxflagset(), and resetxflag(). tmx.t_flush_xflags() does “flush the extended flags from the current SIP message into the already created transaction. It makes sense only in routing block if the transaction was created via t_newtran() and the extended flags have been altered since. It is not needed to execute this function when using t_relay() (or similar tm relay functions, xflags are synchronized automatically in that case).” The documentation of the tm module does not say anything about extended flags - in particular it does not say that t_relay() synchronizes them. Moreover, while the corex module utilizes the extended flags, the documentation of tm and tmx do not state, that they depend on the corex module. So, what are extended flags? Greetings Дилян

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SIP Trunk with multiple phone numbers for account
by Carlos Escalona 19 Jan '23

19 Jan '23

Hello all! I have a question. In my system we use a series of VoIP providers to receive external calls to our telephone numbers, for the same provider we have several registered numbers and we need our providers to send that information in some way, so far it does not seem that there is a common way to do it (report the number that was dialed by the person who made the call). From what I understand, SIP does not have a place for that information because there are no things like phone numbers in the specification, my question is, is there a correct place to send such information, or is it entirely up to each provider? (Now, the most common is to have in the RURI and To header, the id of the trunk registered in the uacreg table) I have seen a couple of options that seem to me the most “correct”: send the information in the To header or send in a custom header. I really appreciate any help you can provide. Atenciosamente, [image: photo] <https://www.techer.com.br/> *Carlos Escalona* IT - Development 41 3073-0091 | R 1011 www.techer.com.br -- O conteúdo desta mensagem é confidencial. Se você não se encontra na lista de destinatários ou tenha recebido por engano, não a copie, imprima, envie, ou utilize, de qualquer forma, seu conteúdo. Neste caso, destrua a mensagem e, por favor, notifique o remetente.

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Setting socket_workers does not work if using interface name
by Kaufman 19 Jan '23

19 Jan '23

At a quick glance it looks like using socket_workers when the listen parameter uses the interface name does not work. For example: children = 4 socket_workers=2 listen=udp:10.10.10.10:5060 Gives two listeners on udp:10.10.10.10:5060 children = 4 socket_workers=2 listen=udp:eth0:5060 Gives four listeners on the socket. Note this is just a quick cursory observation. Is it by design or a design limitation? Kaufman

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RTC/Kamailio dinner at Fosdem 2023
by Daniel-Constantin Mierla 17 Jan '23

17 Jan '23

Hello, Fosdem 2023 returns to Brussels, Belgium, for a physical event and I'm writing to see if there are many people from the community going to the event and want to organize again the usual RTC/Kamailio dinner on Saturday evening (Feb 4, 2023). Each person should pay for himself, likely to be some price increase comparing with the last one. If there is enough interest, Torrey will help to book a place to accommodate us (first choice could be the same restaurant like in the past, if available). Reply if you want to join the dinner and say how many other people are joining you. Note that RTC Devroom got at this edition a half a day room allocation, on the afternoon of Feb 5, 2023 (Sunday). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

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Re: rtp lost package in 300 concurrent calls and above
by Henning Westerholt 16 Jan '23

16 Jan '23

Hello, please keep the list in CC. Ok, the machine seems to not really using all its cores. Regarding playing a file, this can be also done with rtpengine: https://kamailio.org/docs/modules/5.5.x/modules/rtpengine.html#rtpengine.f.… Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com> Sent: Monday, January 16, 2023 3:35 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] rtp lost package in 300 concurrent calls and above thanks for your reply my kamailio server has 8GB of ram and 4core*2.6 on esxi(dedicated)Ubuntu 22.04.1 LTS. is it low? htop on server shows load average 1.04 at max and I didn't see more than 10% load on any core of the server. I've selected rtp media server because of playback of a voice file over incoming calls in the future usage On Mon, Jan 16, 2023 at 5:45 PM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, do you observe (too) high load on the system with the standard system management tools? If you want to just proxy calls, my recommendation would be to use rtpengine. Here you should be able to achieve higher concurrent calls rates, if you have a few CPUs on a decent machine. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com<mailto:keshavarzianpoorm@gmail.com>> Sent: Monday, January 16, 2023 2:50 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] rtp lost package in 300 concurrent calls and above Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

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rtp lost package in 300 concurrent calls and above
by Mohammad Reza Keshavarzianpoor 16 Jan '23

16 Jan '23

Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

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Authenticated ACK: $au not set?
by Benoit Panizzon 16 Jan '23

16 Jan '23

Hi Gang I have this code snipplet: if (has_credentials("$fd")) { xlog("L_INFO", "$cfg(route): got $rm with credentials. Validate them!\n"); if ($aU == $null) { xlog("L_INFO", "$cfg(route): no auth user, send challenge\n"); } auth_challenge("$fd", "0"); exit; } } Please don't ask if this makes sense, it's a constructed snipplet to show the situation. If $rm is INVITE, this works fine, no challenge is being sent, $au is populated. If $rm is ACK (some CPE seem to send the Credentials with each message) then $au is $null, auth_challenge is called and 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 ______________________________________________________

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Parallel call forking with call authorization
by Mathias Schneuwly 13 Jan '23

13 Jan '23

Hi As a first notice; I already wrote to the mailing list, but without registration. I read afterwards that I should register first because of spam reasons. If this is a duplicate, then I'm sorry. I'm new to mailing lists... I've an issue with parallel call forking together with call authentication. I hope somebody can give me some hints about what the problem could be and how I could solve this problem. I'm using Kamailio 5.5.4 with the python KEMI binding. Kamailio is in front of an Asterisk, which requires call authorization. RTPEngine is also inplace which does SRTP to RTP bridging. Simple calls from and to Asterisk through Kamailio and RTPEngine work without any issue. So the basic setup seems to work fine. The problem starts when I do parallel call forking where both calls are sent to Asterisk. If I use "KSR.tm.t_on_failure("trunk_authentication")" for authentication, the first branch is authenticating and the device rings, but the second branch does not authenticate. Well, that's not 100% true, if the SIP timeout appears on the first branch, I can see in the log, that the second branch enters the trunk_authentication method, but then, the call is already terminated. Call forking to two devices (not behind Asterisk and without call authentication) works also as expected. I made some trials with "KSR.tm.t_on_branch_failure("trunk_authentication")" but in this case, the behavior is strange. Then the authentication for both branches work and both devices are ringing. If a device hooks up, the second device does not stop ringing. And after a few seconds, the first device cancels the call, because it did not get the ACKs. I can also see that the CANCEL is sent from Kamailio to Asterisk, but Asterisk does not react to it. For trunk authentication, I'm using the implementation from the well known examples plus a few own stuff. Btw, NAT is not enabled...: def trunk_authentication(self, msg): scode = KSR.pv.get("$T_reply_code") self.log('notice', 'trunk_authentication {}'.format(scode)) if WITH_NAT and self.ksr_route_natmanage(msg) == -255: return 1 # Call manage_rtp_engine to destroy RTPEngine if it has been used. call_context = self.get_call_context(msg=msg) callee_object = call_context.callee.get_callee_object() if callee_object.connection_type: call_context.manage_rtp_engine() if KSR.tm.t_is_canceled() > 0: return 1 if scode in [401, 407]: KSR.pv.sets('$avp(auser)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_username)')) KSR.pv.sets('$avp(apass)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_password)')) if KSR.uac.uac_auth(): # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if KSR.is_method_in('IBSU'): if KSR.tm.t_is_set("branch_route") < 0: KSR.tm.t_on_branch("ksr_branch_manage") self.log('notice', 'Authentication Asterisk trunk') if KSR.tm.t_relay() < 0: KSR.sl.sl_reply_error() return 1 Any ideas? Best regards Mathias

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src/modules.lst - difference between skipping a module from compile list and excluding a module from compile list
by Дилян Палаузов 13 Jan '23

13 Jan '23

Hello, the generated src/modules.lst file contains: # the list of modules to skip from compile list skip_modules= … # the list of modules to exclude from compile list exclude_modules= … For me excluding a module from the compile list and skipping the module when compiling is the same. Please describe what is the difference between skipping and excluding, either in modules.lst, at https://www.kamailio.org/wikidocs/devel/makefile-system/ or at https://kamailio.org/docs/tutorials/devel/kamailio-install-guide-git/ . Thanks Дилян

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Newbie question
by Joao Paulo Vaz 12 Jan '23

12 Jan '23

Hello masters, I am a newbie who recently got to know Kamailio. I have a scenario and I would like to know if this tool would suit: a central asterisk server with all user accounts and three remote units with Kamailio acting as a proxy. Is there any module or configuration that if the central asterisk server fails to respond, would the remote proxies act in survival mode, allowing local communication by users until the central asterisk server returns a response? Att, Joao Paulo

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unsubscribe.pls
by Stanly Levyy 11 Jan '23

11 Jan '23

*** THIS E-MAIL MESSAGE AND ANY DOCUMENTS ATTACHED IS/ARE INTENDED ONLY FOR THE USE OF THE INDIVIDUAL(S) OR ENTITY TO WHICH IT IS ADDRESSED; AND MAY CONTAIN INFORMATION THAT IS LEGALLY PRIVILEGED, CONFIDENTIAL AND/OR EXEMPT FROM DISCLOSURE UNDER APPLICABLE LAW, including the Electronic Communications Privacy Act, 18 U.S.C. Secs 2510-2521 *** On Wednesday, January 11, 2023, 01:36:44 AM PST, <sr-users-request(a)lists.kamailio.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via 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: Performing async_http calls on stateless Loadbalancers (Julius Flohr) 2. Re: Releasing v5.6.3 (Daniel-Constantin Mierla) 3. Performing async_http calls on stateless Load balancers (Lewis Hutchinson) ---------------------------------------------------------------------- Message: 1 Date: Tue, 10 Jan 2023 18:50:44 +0100 From: Julius Flohr <flohr(a)sipgate.de> Subject: [SR-Users] Re: Performing async_http calls on stateless Loadbalancers To: Henning Westerholt <hw(a)gilawa.com> Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Alles wird besser <alleswirdbesser(a)sipgate.de> Message-ID: <34C7487C-0900-4C49-BBB4-49A0291D83E6(a)sipgate.de> Content-Type: multipart/alternative; boundary=Apple-Mail-0AF7C672-1C85-4A98-9DDC-EED95FFB4B7E A message part incompatible with plain text digests has been removed ... Name: not available Type: text/html Size: 7117 bytes Desc: not available ------------------------------ Message: 2 Date: Wed, 11 Jan 2023 08:30:17 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: [SR-Users] Re: Releasing v5.6.3 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org> Message-ID: <454ae3b5-a810-f393-4196-e71c4482e9d3(a)gmail.com> Content-Type: text/plain; charset=UTF-8 Hello, the usual note that preparing the release of Kamailio v5.6.3 starts soon, therefore any commits to branch 5.6 should be announced first on sr-dev mailing list and wait for the ack before pushing to repo. Once the announcement is out, commits can go in the branch directly. Cheers, Daniel On 06.01.23 16:44, Daniel-Constantin Mierla wrote: > Hello, > > I am considering to release Kamailio v5.6.3 (out of branch 5.6) next > week (likely on Wednesday, Jan 11, 2023). If anyone is aware of > issues not yet on the bug tracker, report them there asap in order to > have a better chance to be fixed. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com ------------------------------ Message: 3 Date: Wed, 11 Jan 2023 09:20:28 +0000 From: Lewis Hutchinson <lewis.hutchinson(a)missionlabs.co.uk> Subject: [SR-Users] Performing async_http calls on stateless Load balancers To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Cc: "flohr(a)sipgate.de" <flohr(a)sipgate.de>, "alleswirdbesser(a)sipgate.de" <alleswirdbesser(a)sipgate.de> Message-ID: <CWXP265MB2517CB6E3E8188F3E3BB5B2AC5FC9(a)CWXP265MB2517.GBRP 265.PROD.OUTLOOK.COM> Content-Type: text/plain; charset="us-ascii" Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list -- sr-users(a)lists.kamailio.org To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org ------------------------------ End of sr-users Digest, Vol 212, Issue 33 *****************************************

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Performing async_http calls on stateless Load balancers
by Lewis Hutchinson 11 Jan '23

11 Jan '23

Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis

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Releasing v5.6.3
by Daniel-Constantin Mierla 11 Jan '23

11 Jan '23

Hello, I am considering to release Kamailio v5.6.3 (out of branch 5.6) next week (likely on Wednesday, Jan 11, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

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Help with rtpengine_delete()
by Jose Figueroa 10 Jan '23

10 Jan '23

Hello everyone! I've been struggling with a kamailio scenario with the rtpengine_delete. My scenario is the following: I've setup a Kamailio multihomed, since some of my vendors has provided us a direct link using a different local subnet in eth2 and I have my public IP in eth1. The kamailio i've been working on is using app_lua module. SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth2) 10.203.5.8 <------> Vendor 10.203.5.20 100.XXX.XXX.22 Basically, I receive a call from public IP eth1 and according to the logic from the vendor I forward statefully the call to 10.203.5.20 in second branch. I've been reading a lot of issues and documentation such as:[#840 rptengine], [#875 rtpengine] and [#2222 kamailio], but I don't really get working the rtpengine_delete() correctly, since I go to netstat -plan|grep rtpengine and I see some sessions still that are not freed. My second case is the following when I receive a 503 from another GW: SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth1) 200.XXX.XXX.14 <-----------> Vendor GW 233.XXX.XXX.19 100.XXX.XXX.22 In that case I still see the rtpengine ports, mostly with my public IP 200.XXX.XXX.14 but some of them with the LAN 10.203.5.8 IP. Also, something curious I see on my 2nd use case is the following: https://pastebin.com/raw/B4vCgExX So, it would be great if someone can point me to the right direction, how and when I should use rtpengine_delete in order to finish the rtpengine sessions correctly and the ports can be freed up. Any help will be great! [840] https://github.com/sipwise/rtpengine/issues/840 [875] https://github.com/sipwise/rtpengine/issues/875 [2222] https://github.com/kamailio/kamailio/issues/2222 -- Regards, Jose Figueroa

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Why do htable Keys have two colons?
by Noah Mehl 10 Jan '23

10 Jan '23

Hey everyone, Whenever I read documentation for htable usage, and other examples of other scripts in Kamailio, htable keys are typically named: something::something Examples are: $au::auth_count, $ci::srcip, join::$rU Why are there two colons (as opposed to a | )? Is this just a standard semantic thing in Kamailio? Is there something about that double character that’s unusual and easy to parse with SIP values? Or is it just what people do, and that’s a good choice to continue for all other Kamailio script writers to have understanding? Any insight would be appreciated. Thanks! ~Noah

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Kamailio Crypto module and 3rd party openssl
by Arsenijs Kabihno 10 Jan '23

10 Jan '23

Hello, I am trying to make use of crypto module. The scenario is the following: 1. Encrypt plaintext passwords on kamailio side using crypto_aes_encrypt function of crypto module 2. Decrypt them outside of kamailio using 3rd party tools, such as PHP openssl_decrypt Crypto module parametrs are the following: modparam("crypto", "salt", "") modparam("crypto", "key_derivation", 0) If encryption and decription are handled by kamailio, the initial password matches the decrypted string (initial password -> encrypted string -> decrypted string) For example: sql_query("ds_dburl", "select secret from kamailio_sip_buddies where username='$au'", "secret"); $avp(secret_plain) = $dbr(secret=>[0,0]); //value: 'kamailio' crypto_aes_encrypt("$avp(secret_plain)", "YUZySHVtdlVTYjI0TGJNd3JTeVV0MlRaRWFraFNBRlM=", "$var(encrypted)"); //value: 'qfOqTMoJMgGAherGCqLRRQc4zTlqsEj3MEtcORurFf0=' crypto_aes_decrypt("$avp(secret_plain)", "YUZySHVtdlVTYjI0TGJNd3JTeVV0MlRaRWFraFNBRlM=", "$var(decrypted)"); //value: 'kamailio' But in case I try to decrypt the value of $var(encrypted) using PHP openssl_decrypt or online tool like https://www.devglan.com/online-tools/aes-encryption-decryption, I am getting the result 'kamailio' prepended by 16 random bytes, like: 6“¤›¿Ð+IÊ\ñžïZkamailio or 6��+I\��Zkamailio I believe this 16 bytes are actually random initialization vector in some encoding. Possibly kamailio is trimming them from result here: https://github.com/kamailio/kamailio/blob/master/src/modules/crypto/crypto_… Moreover, if secret was encrypted using crypto_aes_encrypt and random IV, you can use any IV while decrypting and get 16 different bytes+result Please advice if someone have used crypto module in similar way and what are the possible ways to acheive results consistency

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pkg memory leak when acc module cdr_enabled
by Juha Heinanen 10 Jan '23

10 Jan '23

In latest stable K release, we noticed pkg memory leak (pgk memory usage increases by each processed call). It turned out that the leak goes away if acc module cdr_enable is not enabled. Could be a bug in dialog or acc module. Any debug instructions if the bug is not obvious? -- Juha

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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>

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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

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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

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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

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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

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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

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