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37456 discussions

[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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