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37385 discussions

kamailio & rtpengine question
by Gerry Kernan 15 Oct '22

15 Oct '22

Hi Looking for advise if anyone could point out where im going wrong. My setup is Kamailio Ver 5.6.0 running in azure , rtpengine Version: git-master-dc395d5, behind kamailio I have asterisk ver 18.12.1 Kamailio conf I have interface set as below listen=udp:10.3.1.6:5060 advertise WANIP:5060 and rtpengine.conf set as below interface = eth0/10.3.1.5 but when I can make a call I don't get any audio, rtpengine.log extract below shows that rtp is going to the private IP rather that the Public IP of the phone that initiated the call, do I need to set rtpengine_manage() to fix this? Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Confirmed peer address as 10.3.1.5:12890 Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Kernelizing media stream: 10.3.1.5:12890 -> 10.3.1.6:30278 | 10.3.1.6:30298 -> 192.168.0.123:40042 Oct 7 09:39:00 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30279]: [core] Confirmed peer address as 10.3.1.5:12891 Kind regards gerry Gerry Kernan | Technical Director Mobile: +353(0)86-170 9790 [Graphical user interface, website Description automatically generated]<https://www.infinityit.ie/> [A picture containing text, clipart Description automatically generated,Icon Description automatically generated]

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tsilo usage more generally
by Alex Balashov 14 Oct '22

14 Oct '22

Since on the topic of tsilo: 1. I watched Federico’s presentation on it from 2015, and his examples insinuate that it is necessary to store the index and label of a suspended transaction in some external vehicle, e.g. htable, for use with ts_append_to(). Why would one do this when it is possible to call ts_append() based on R-URI alone? 2. The appropriate pattern of interaction between tsilo and http_async_client — which I imagine to be a rather common use-case — is rather unclear. If I use http_async_query() to call out to a PN service, with $http_req(suspend) = 1, what should happen to the current instance of SIP request processing? That is to say: request_route { . . . t_check_trans(); if(is_method(“REGISTER”)) { . . . save(“location”, “0x06”); if(ts_append(“location”, “$tu”)) { xlog(“success”); } return; } else if(is_method(“INVITE”)) { # PN stuff. ts_store(“$var(uri)”); http_async_query(“http://url”, “PUSH_REPLY”); return; # <— ??? } } route[HTTP_PUSH_REPLY] { . . . if(!t_relay()) sl_reply_error(); } I should add that this proxy front-ends a registrar, but is not in itself a registrar. It contains a shadow registrar - hence the save() calls - in order to be able to take advantage of the ’tsilo’ module, given its dependency on registrar/usrloc. This leads to the difference that the gateway receives requests with RURIs that have already been resolved to a contact binding elsewhere, so there are no lookup() calls. How should this play into the http_async_client workflow? Appreciate any help! — 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

tsilo logging
by Alex Balashov 13 Oct '22

13 Oct '22

Hi, When there are no stored transactions for an AOR, it logs this error: Oct 13 07:48:18 gw1 /usr/sbin/kamailio[14758]: ERROR: tsilo [ts_append.c:64]: ts_append(): failed to retrieve record for sip:abalashov@sip.evaristesys.com But not having transactions ts_store()’d is a rather normal occurrence, right? It is the expected case the vast majority of the time that someone registers. Does that mean one is simply destined to have this scary-looking error, to the syslog ERROR facility, for a normal runtime condition? Is there not some principle against using attention-grabbing error messages this way? Would a patch to fix be welcome? — 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/

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About kamailio + freeswitch issue on Kamailio v5.6
by SparkleZou 12 Oct '22

12 Oct '22

Hi Sir/Mada, I'm trying to integrate kamailio + freeswitch, according to the manual http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc on Kamailio v5.6 modified the kamailio.cfg file, add WITH_FREESWITCH parts, then kamailio could NOT start up. Could you please help check, what should be modified to fit the version 5.6？ Thanks！ BR， Sparkle Zou

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Re: [SR-Users] play_media sounds interrupted
by Stepan Kislyakov 12 Oct '22

12 Oct '22

> The duration of the media playback is returned from the play_media() > command (see `media_duration` modparam). You could use some sort of > timer to invoke media unblocking after media playback ends. Sadly my > Kamailio-Fu isn't strong enough to be able to tell you how to run this > sort of timer. :) > > Cheers Thank you for the hint. I was thinking about timers, but rtimer cannot be triggered in an event route. I used a "timer" module, it could trigger another route from the event route but when a new route is triggered unblock_media() is not aware of the call-id and if i assign call-id to a variable in the event route, that variable is unavailable at new route. Hope that make sense. here is my peace of code: loadmodule "timer.so" modparam("timer", "declare_timer", "PB_TIMER=START,5000"); route[START]{ xlog("current call id at event route is $ci, but call-id avp is $avp(call_id)"); unblock_media("all"); timer_enable("PB_TIMER",0); #we need to disable timer or it will be triggered over and over again } event_route[dialog:start]{ xlog("L_INFO", "dialog started, received 200OK"); #block_media(); play_media("all file=/usr/local/src/myfile.wav"); $avp(call_id)=$ci; timer_enable("PB_TIMER",1); }

1 0

play_media sounds interrupted
by Stepan Kislyakov 11 Oct '22

11 Oct '22

Hi, I'm using latest kamailio 5.6.2 and rtpengine. I need to play announcement to both parties when call is answered by callee. Now I run play_media in event route but when audio start announcement sounds like we have a 50% packet loss. After it is completed there is no problem with audio both sides. I figured out that if i do block_media() my announcement play perfectly but i dont understand how to do unblock_audio() after announcement ends. Appriciate any help! here is my code: event_route[dialog:start]{ xlog("L_INFO", "dialog started, received 200OK"); block_media(); play_media("all file=/usr/local/src/myfile.mp3"); #unblock_media(); } -- Best Regards Stepan Kislovich

2 1

IPSec problem with P-CSCF
by Ali Akbar Hemmati 11 Oct '22

11 Oct '22

Hi everyone, I am using Kamailio as an IMS server. The UE first sends its SIP REGISTER over TCP and gets a 401 in response. Then the UE starts sending its packets over ESP. These ESP packets reach P-CSCF but are not delivered to its process by the kernel. We can see ESP packets in P-CSCF using wireshark. The ports used are correct (both client and server) based on their prior negotiations. We have also checked xfrm policies using the command `ip xfrm policy list` and seems right regarding prior negotiations. Our Linux kernel version is 5.4 and the OS is Ubuntu 20.04. Has anyone any clue of the root cause of this problem? Any ideas are appreciated. Thanks, Aliakbar

2 6

kamcli folder
by Volker Denneberg 11 Oct '22

11 Oct '22

hello, I'm not very familiar with PHP, so this may be a stupid question. I'm following https://github.com/kamailio/kamcli to install kamcli. Now i am at $ pip3 install -r requirements/requirements.txt which i should "run inside kamcli folder". So far, I don't have a kamcli folder, so i assume i have to create it. Any recommendations where ? Thanks in advance, Volker

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detect tcp connection close using tcpops module
by sthustfo 11 Oct '22

11 Oct '22

I am running latest kamailio 5.6.2 and attempting to detect tcp/tls connection being disconnected or closed using tcpops module. As suggested in tcpops documentation at https://www.kamailio.org/docs/modules/devel/modules/tcpops.html, registered event routes as below. event_route[tcp:closed] { xlog("L_INFO", "$proto connection closed ($conid)\n"); } event_route[tcp:timeout] { xlog("L_INFO", "$proto connection timed out ($conid)\n"); } event_route[tcp:reset] { xlog("L_INFO", "$proto connection reset by peer ($conid)\n"); } Now it is observed that the event route is being called but the $conid is not set. The logs show the below log statements and extra data is (nil). 24(4263) DEBUG: <core> [core/tcp_read.c:284]: tcp_read_data(): EOF on connection 0x7ffff2ff8b90 (state: 0, flags: 4018) - FD 7, bytes 0, rd-flags 10000 ([10.8.137.33]:51069 -> [10.8.137.33]:5060)24(4263) DEBUG: <core> [core/tcp_read.c:1500]: tcp_read_req(): EOF 24(4263) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x555555ab6960, 7, -1, 0x10) fd_no=2 called 24(4263) DEBUG: <core> [core/tcp_read.c:1870]: handle_io(): removing from list 0x7ffff2ff8b90 id 1 fd 7, state 2, flags 4018, main fd 51, refcnt 2 ([10.8.137.33]:51069 -> [10.8.137.33]:5060) 24(4263) DEBUG: <core> [core/tcp_read.c:1653]: release_tcpconn(): releasing con 0x7ffff2ff8b90, state -1, fd=7, id=1 ([10.8.137.33]:51069 -> [10.8.137.33]:5060) 24(4263) DEBUG: <core> [core/tcp_read.c:1657]: release_tcpconn(): extra_data (nil) 32(4271) DEBUG: <core> [core/tcp_main.c:3651]: handle_tcp_child(): reader response= 7ffff2ff8b90, -1 from 0 32(4271) DEBUG: <core> [core/tcp_main.c:3576]: tcp_emit_closed_event(): TCP closed event creation triggered (reason: 0) 32(4271) DEBUG: tcpops [tcpops.c:295]: tcpops_handle_tcp_closed(): received TCP closed event 32(4271) DEBUG: tcpops [tcpops.c:249]: tcpops_tcp_closed_run_route(): event reason id: 0 rt: 1 32(4271) DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 32(4271) DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): SIP Request: 32(4271) DEBUG: <core> [core/parser/msg_parser.c:681]: parse_msg(): method: <OPTIONS> 32(4271) DEBUG: <core> [core/parser/msg_parser.c:683]: parse_msg(): uri: <sip:you@kamailio.org> 32(4271) DEBUG: <core> [core/parser/msg_parser.c:685]: parse_msg(): version: <SIP/2.0> 32(4271) DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 32(4271) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=2 32(4271) DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 32(4271) DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via 32(4271) exec: *** cfgtrace:dbg_cfg_trace(): unknown_route=[tcp:closed] c=[/usr/local/etc/kamailio/kamailio.cfg] l=976 a=26 n=xlog32(4271) INFO: <script> tcp connection closed (<null>) Can anyone provide any pointers on what I am missing? Thanks in advance.

2 3

suppress t1 timer retransmissions?
by Brooks Bridges 10 Oct '22

10 Oct '22

So I have a very odd situation that requires me to suppress retransmissions when using t_reply() to send a response back to a UAC that is horribly broken but I'm stuck with for the time being. In the previous iteration of this platform it was fine since I was using a stateless reply, so even though it wasn't "right" it still worked out. I need to replicate this behavior in the event route for http_async_client(), which prevents me from being able to use t_release(). Anyone have any clever ideas? Brooks Bridges 1200 S. Rogers Circle, Unit 6 Boca Raton, FL 33487 Office: 916-235-2097 Main: 888-444-1111 email: bbridges(a)call48.com | web: www.call48.com Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

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