- sr-users - kamailio.org

Issue with Authenticating outbound Call
by Victor ku 22 Feb '22

22 Feb '22

Hi, I'm new to sip and kamailio, I have recently setup a docker container based on kamailio/kamailio:5.5.0-stretch to check things out. While I was trying to authenticate an outbound call to a 3rd party sip provider from my kamailio container, I noticed there are times that the 2nd invite with the authorization header that's supposed to be sent from the failure route doesn't get sent out. There's no error recorded in the log nor does wireshark picking up any reply to sip phone or message to 3rd party sip provider. Anyone know what might have caused this? Some other things to note: Kamailio does sent out the INVITE with authorization header and authenticated successfully when my sip phone sent the sip request with FROM ip set to the 3rd party sip server's ip. When I change the sip phone to send INVITE with FROM Domain using kamailio container's ip, I'm able to see header is being built from the trace but no INVITE with authorization header is being sent. Changing the FROM uri using $fu or uac_replace_from() does help either. ---------------------------------------------------------------------------------------------------------------- > > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} uac [auth_hdr.c:408]: > build_authorization_hdr(): hdr is <Authorization: Digest > username="xxxxxxxx", realm="xxx.xxx.xxx.xxx", nonce="07f4a5779568", > uri="sip:+0123456789@xxx.xxx.xxx.xxx", qop=auth, nc=00000001, > cnonce="3782993246", response="d5ebd0ef00d44db2292c7b8508fa23bc", > algorithm=MD5 > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1026 a=16 > n=if > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1021 a=24 > n=t_relay > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} <core> [core/msg_translator.c:1797]: > check_boundaries(): no multi-part body > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1030 a=2 > n=exit > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483650 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1564]: > t_should_relay_response(): rps discarded - uas status: 100 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1924]: relay_reply(): > reply status=2 branch=0, save=0, relay=-1 icode=0 msg status=401 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:596]: > receive_msg(): reply-route executed in: 1753 usec > kamailio-kamailio-1 | 14(22) DEBUG: dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483652 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:606]: > receive_msg(): cleaning up ---------------------------------------------- package capture ---------------------------- 192.168.0.165 xxx.xxx.xxx.xxx SIP/SDP 1383 Request: INVITE > sip:+0123456789@ xxx.xxx.xxx.xxx | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 585 Status: 100 Trying | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 820 Status: 401 Unauthorized | > 192.168.0.165 xxx.xxx.xxx.xxx SIP 396 Request: ACK sip:+0123456789 > @ xxx.xxx.xxx.xxx | ------------------------------------------nothing more till sip phone cancel ----------------- # ----------------- below is failure route config --------------------- > failure_route[MANAGE_FAILURE] { > xlog("~~~~~ MANAGE_FAILURE ($rm) ($rs-$rr) (from $fu) (to $tu) (req :$ru) > (dest $du)\r\n"); if (t_is_canceled()) { exit; } xlog("Checking status code\r\n"); #!ifdef WITH_UAC if(t_check_status("401|407")) { if (isflagset(7)) { t_reply("503","Authentication failed"); exit; } $avp(auser) = "xxxxxxxxxx"; $avp(apass) = "xxxxxxxxxx; if(uac_auth()) { setflag(7); xlog("after uac_auth $rm (from $fu)(to $tu)(req $ru)(dest $du)\r\n"); if(!t_relay()) { xlog("Failed to relay ($rm) ($rs-$rr) (from $fu)(to $tu)(req $ru)\r\n"); t_reply("500","Error"); } } else { xlog("uac_auth() not available\r\n"); t_reply("503","Authentication invalid"); } exit; } #!endif } > # ----------- end failure route config --------------------

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reinvite with 2 video lines in SDP
by Filippo Graziola 22 Feb '22

22 Feb '22

Hi all, I'm facing a strange behaviour from a client (iOS application based on pjsip library) which is doing as follow when receiving a call (on phone locked): - accept the call rejecting video (m=video 0...) in SDP - reinvite to negotiate again video media Specifically the reinvite passing through kamailio (+rtpengine) has this SDP body after rtpengine offer (the client use TLS so I can't see the original message): v=0 o=- 3854443139 3854443142 IN IP4 PUBLIC_RTP_ENGINE_IP s=pjmedia b=AS:386 t=0 0 a=X-nat:0 m=audio 46734 RTP/AVP 8 101 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:96000 a=ssrc:1456418416 cname:441b90ac6db72f22 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:46735 m=video 0 RTP/AVP 96 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 m=video 46776 RTP/AVP 98 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=42e01e; packetization-mode=1 a=sendrecv a=rtcp:46777 from this I can understand that the client is doing something strange, because there is still the rejection line (I know that this question should go to the pjsip guys) but I would also like to know if it is possible to "clean" this sdp by kamailio. A few clients receiving this sdp are answering with only audio. Reading the docs, sdpops/textops modules allow modifying lines by prefix or media, but till now I have no luck to exclude only the line "m=video 0 RTP/AVP 96" and his attributes without touching the correct video request. Could someone please give me some hints on it?

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Enable switch redirect to other SIP server source
by bayu p 21 Feb '22

21 Feb '22

Dear Kamailio Expert, Could you help me switching redirect sip server to another sip server ? As described below architecure. I have some uac_redirect module to implement , but kamailio cannot perform running states. Pleas help to reditect sip server to another sip server. Thanks & Best regards,

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kamailio success story
by Valentin Christoph 21 Feb '22

21 Feb '22

Hi all, We had to connect to an SBC, which was not compliant to RFC 3261. It was urgent, because management was waiting for first call. Within minutes, I could implement a workaround in kamailio routing configuration: ## CV experimental 2022-02-17: fake Supported HF if (is_present_hf("Require") && !is_present_hf("Supported")) { append_hf("Supported: $hdr(Require)\r\n", "Require"); msg_apply_changes(); } Thank you, kamailio! All the best Christoph

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SIP server port service cannot be changed from default value (5060)
by bayu p 18 Feb '22

18 Feb '22

Hello Experts, Please help me to change port service from default 5060 port into other port due to security reason. Since I have change port in kamailio.cfg file, but services cannot fly normally that shown client unable login online (status not online). [image: image] <https://user-images.githubusercontent.com/76675592/154027407-24ab423b-302b-…> Besides, I need to run the kamailio services after install default kamailio, I have not found where procedure found in this installation process on github.com/kamailio. Could you please share if there another steps of compiling Kamailio source, install, deploy process, and running command for this specific installation process? Thanks Gb

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Kamailio v5.5.4 Released
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, Kamailio SIP Server v5.5.4 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.4. For more details about version 5.5.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/02/kamailio-v5-5-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Releasing Kamailio v5.5.4
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, I am considering to release Kamailio v5.5.4 later this week, likely on Thursday, Feb 17, 2022. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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Info: sipexer v1.0.0 - sip cli tool
by Daniel-Constantin Mierla 16 Feb '22

16 Feb '22

Hello, I want to announce the availability of sipexer v1.0.0 - a sip cli tool that can facilitate testing and monitoring of SIP signalling systems. It tries to have a modern approach, with a flexible templating system, supporting both IPv4 and IPv6 with all the common transport layers, respectively UDP, TCP, TLS and WebSocket (for WebRTC). The project can be found at: * https://github.com/miconda/sipexer It is written in Go language for better portability, binaries for Linux, MacOS and Windows are made available for download in the release page: * https://github.com/miconda/sipexer/releases/tag/v1.0.0 Among its features: * send OPTIONS request (quick SIP ping to check if server is alive) * do registration and un-registration with customized expires value and contact URI * authentication with plain or HA1 passwords * set custom SIP headers * template system for building SIP requests * fields in the templates can be set via command line parameters or a JSON file * variables for setting field values (e.g., random number, data, time, environment variables, uuid, random string, …) * simulate SIP calls at signalling layer (INVITE-wait-BYE) * respond to requests coming during SIP calls (e.g., OPTIONS keepalives) * send instant messages with SIP MESSAGE requests * color output mode for easier troubleshooting * support for many transport layers: IPv4 and IPv6, UDP, TCP, TLS and WebSocket (for WebRTC) * send SIP requests of any type (e.g., INFO, SUBSCRIBE, NOTIFY, …) One usage example that could ease the testing of Kamailio is initiating registrations or simulating calls over WebSocket without the need of having a JavaScript soft phone application running in a web browser. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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allow_source_address_group in failure route
by Marrold 16 Feb '22

16 Feb '22

Hello, I'm using Kamailio as a proxy between several voice gateways and if the call fails for whatever reason (far end is busy etc) I am forwarding to an Asterisk box to play an announcement. Within the failure route I am using allow_source_address_group() to ascertain if the failure response was from a Voice Gateway or Asterisk itself, however it appears to return the group for the first branch when the reply is from the second branch to Asterisk. To handle the failure I am adding some headers, and setting a new destination indirectly with ds_select_dst before t_relay. I am using 5.5.3 on Debian 11, using KEMI / Python. Does anyone have any ideas? Thanks Matthew

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Missing contact in 180 ringing causes wrong routing
by Xuo Guoto 15 Feb '22

15 Feb '22

Hello all, I have a kamailio ( kamailio 5.5.3 (x86_64/linux)) with asterisk (Asterisk 16.2.1~dfsg-2ubuntu1) with following architecture ┌─────────────┐ ┌─────────────┐ ┌─────────────────────┐ │ Asterisk ├───────┤ Kamailio │ 188.xxx.59.171 ───────── 103.xxx.xx.134 │ Linphone │ └─────────────┘ └─────────────┘ └─────────────────────┘ 10.10.1.13 10.10.1.9 In case the diagram is not displayed correctly its pasted at https://pastebin.com/b0UtE4Lj Registrations are handled by Kamailio and calls from linphone are routed to Asterisk. Similarly all calls originating in asterisk are routed over to Kamailio which in turn looks up AOR and routes the calls over to Linphone. Along with Linphone I also have a WebRTC endpoint at Kamailio. When calls are sent to WebRTC from asterisk, it gets correctly routed over to Asterisk, but if the client is Linphone the ACK of 200OK is not routed correctly to Linphone. I get this error in the logs: [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced tls:188.xxx.59.171:5061, to udp:188.xxx.59.171:5060) Digging into the difference between WebRTC and Linphone, I could see that in 1800 Ringing packet from WebRTC has a Contact headers while its missing in the packet from Linphone This is the 180 ringing from WebRTC: tag: rcv pid: 695615 process: 17 time: 1644860126.209155 date: Mon Feb 14 23:05:26 2022 proto: wss ipv4 srcip: 103.xxx.xx.134 srcport: 52508 dstip: 188.xxx.59.171 dstport: 443 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/WSS 188.xxx.59.171:443;branch=z9hG4bK730a.2cc7461b72f0c895e7cce5a0cedce4f7.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Content-Length: 0 While this is from Linphone tag: rcv pid: 695614 process: 16 time: 1644860097.970935 date: Mon Feb 14 23:04:57 2022 proto: tls ipv4 srcip: 103.xxx.xx.134 srcport: 25843 dstip: 188.xxx.59.171 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/TLS 188.xxx.59.171:5061;branch=z9hG4bKb5b1.d3231367383d07b1063098dcb35cc2b0.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPjd16f1379-1be7-448e-894a-c42d96f63146 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=cf7a7770-529d-4450-8bed-880382dce1dd To: <sip:user_jbjp@10.10.1.9>;tag=2ooYDia Call-ID: 8a025999-b096-4fe0-a623-9f6e6a69b8ae CSeq: 27838 INVITE User-Agent: Linphone Desktop/4.3.2 (Ubuntu 20.04.3 LTS, Qt 5.12.12) LinphoneCore/5.0.49 Supported: replaces, outbound, gruu Record-route: <sip:188.xxx.59.171:5061;transport=tls;r2=on;lr=on;did=3b9.b1b2> Record-route: <sip:10.10.1.9;r2=on;lr=on;did=3b9.b1b2> Content-Length: 0 When the 180 ringing is passed on to the Astrerisk it has the contact header modified with alias. tag: snd pid: 695615 process: 17 time: 1644860126.210189 date: Mon Feb 14 23:05:26 2022 proto: udp ipv4 srcip: 10.10.1.9 srcport: 5060 dstip: 10.10.1.13 dstport: 7000 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;alias=103.xxx.xx.134~52508~6;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Content-Length: 0 Now, I am not sure if this is the reason for routing the packets to wrong destination, but this is all I could figure out. Any clue to get this resolved will be much appreciated. My Kamailio config is pasted at https://pastebin.com/qyRS0usw with regards, X. Sent with ProtonMail Secure Email.

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get kamailio sip processes id
by Patrick Karton 14 Feb '22

14 Feb '22

hello, as we are all aware kamailio create a lot of processes for many tasks. im using kamailio in kubernetes and i would like to monitor only sip processes with top command. i only need a way to get only kamailio sip processes id (udp receiver that handle sip request) . but i dont knwo how to get it. top -c -p id1,id2,id3,id4 Thanks

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SEMS license with kamailio and rtpengine
by Patrick Karton 14 Feb '22

14 Feb '22

Hello im using kamailio and rtpengine to build a custom SBC. i was using freeswitch as my b2bua but i noticed a lot of issues when cps and simultaneous increased so i tried sems and i have better results. But i do not really understand sems license so i want to know if i can use it. i read from sems github "It is licensed under dual license terms, the GPL (v2+) and proprietary license. This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed. For a license to use SEMS under non-GPL terms, please contact FRAFOS GmbH at info(a)frafos.com" so my question is if i want to use sems as b2bua with kamailio and rtpengine and make profit of it do i need to purchase a license each time i want to sell it to a customer ? Thanks.

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Kamailio server behind NAT
by Tihomir Culjaga 13 Feb '22

13 Feb '22

Hi all I'm wondering what is best practice to configure kamailio for a setup shown below. Carriers only support interconnection over a private network so I need to terminate every separate network on kamailio machine (and on RTPEngine) . That will turn very quickly into an administration nightmare as we add more and more carriers. My plan is to terminate carrier private networks on a router and NAT the traffic to kamailio where it leads me to a kamailio behind NAT setup. Well, in theory set_advertised_address and set_advertised_port per message can do the trick... at least I hope. Also, some carriers require a SIP registration to activate SIP trunks. I do that using UAC module (remote registration) but here again I got a problem with contact address as you can set only one contact address (modparam("uac", "reg_contact_addr", "10.1.1.11:5060")). /--RTP 10000-19999-- RTPEngine1 -----RTP------+-- IPPBX1 carrier1 --- NAT ---| | | | | | carrier2 --- NAT ---+-----SIP 5060------ kamailio ----SIP---- Dispatcher | | | carrier3 --- NAT ---| | | \--RTP 20000-29999-- RTPEngine2 -----RTP------+-- IPPBX2 So bottom line, I'm wondering what would be the best practice for such a scenario ? That would spare me from hitting many pitfalls. Would really appreciate a hint. Regards, Tihomir.

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Vo5G with open5gs
by Hornbuckle, Mark A 11 Feb '22

11 Feb '22

Hello, I have been trying to find documentation on Kamailio IMS and Vo5G support for open5gs. I wanted to ask you directly if Kamailio IMS does support Vo5G with open5gs? Thank you

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resending same message from UE after receiving RP-ACK
by Eugene B 11 Feb '22

11 Feb '22

Hi, I trying to use smsops module as SMSC. When I sending message from UE I see re-sending same message after each RP-ACK(~10 attempt). What could be the problem? Thanks! pcap in attach. kamailio 5.3.2 -- *Yours faithfully,* *Eugene B.*

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kamilio open5gs 5G SA compatible?
by Sandland, Seth P 11 Feb '22

11 Feb '22

Were looking to test a kamilio on Open5gs 5g stand alone ( non lte, non NSA) can you conform or provide documents or install instructions on how this is accomplished. I can not find much of any details if Kamilio support Open5Gs 5g SA and what features it does support on a 5g SA core. Seth Sandland Wireless Communications IT Engineer Pacific Northwest National Laboratory Tel: (509) 375-6842 Email: seth.sandland(a)pnnl.gov<mailto:seth.sandland@pnnl.gov> www.pnnl.gov<http://www.pnnl.gov/> [A close up of a logo Description automatically generated]

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Publish XML and xpath
by Igor Olhovskiy 10 Feb '22

10 Feb '22

Hello, I'm trying to get state of endpoint from PUBLISH message in a most simple way using xmlops module Trying to make this $xml(body=>doc) = $rb; log("L_ERR", "$rU is now in state $xml(body=>xpath:/dialog-info/dialog/state)\n"); XML in PUBLISH itself <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:88881@dev.company"> <dialog id="v6qmEYf5yI" direction="initiator"> <state>confirmed</state> </dialog> </dialog-info> Actual text in log is ... <script>: 88881 is now in state Means it's empty. xmlops module is loaded with no parameters. Tried also string $xml(body=>xpath:/dialog-info/dialog/state/text()) -> same result What am I missing here? kamailio 5.4.6 -- Thanks in advance, Igor

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KEMI KSR.async Python3
by Ahmed Moubayed 10 Feb '22

10 Feb '22

Hi, I'm trying to use the async module from a python3 Kemi script but it conflicts with the keyword async. Any suggestions on how it can be used? Is there an alternative to KSR.async.route()? Thanks, Ahmed

1 0

CDR preserve original call direction
by Ahmed Fouad 10 Feb '22

10 Feb '22

Hi, I've enabled ACC and CDR logging and it's working fine for the logging but I noticed something wrong in the CDR table , the call source and destinations are depending on the BYE message not the initial invite dialog ACC table +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | id | method | callid | time | src_ip | src_domain | src_user | dst_ouser | dst_domain | +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | 216 | BYE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:57:51 | x.x.x.x | domain.com | DST_CLID | SRC_CLID | domain.com | | 215 | INVITE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:55:17 | x.x.x.x | domain.com | SRC_CLID | DST_CLID | dst_domain.com | The INVITE is the correct one with SRCIP x.x.x.x and domain.com but the BYE reversed the order since the the DST_CLID who sent the BYE so it's now logged as the source and the final CDR log looks like +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | cdr_id | src_username | src_domain | dst_username | dst_domain | call_start_time | call_end_time +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | 195 | DST_CLID | domain.com | SRC_CLID | domain.com | 2022-02-09 06:55:12 | 2022-02-09 06:57:51 | the CDR as shown is reversed regarding the source/dst username I tried to enable the direction ACC direction flag but didn't help and here is my current paramters modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "cdrs") Can you please advise how to configure ACC/CDR to log the CDR with the correct initial INVITE call flow regardless of the BYE direction ?

3 5

REGISTER with http_async_client ?
by Cyril Ramière 10 Feb '22

10 Feb '22

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); } } ##### END SNIP Best regards!

4 15

Kamailio doc request
by Mustafa Rahman 10 Feb '22

10 Feb '22

2 1

Failure route with t_relay()
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I am trying to make a re-transmission mechanism from the failure route, I've tried to use $dp and $du as a relay, but variables are not accepted in t_relay(), When using ip and port straight forward t_relay("144.0.0.0", "1111"), I am getting: [t_fwd.c:1752]: t_forward_nonack(): no branches for forwarding [tm.c:1753]: _w_t_relay_to(): t_forward_noack failed Is there a way to re-transmit the call to the same carrier again? Thanks in advance, and have a great day, Max Kemel

2 1

Failure route mechanism with counter
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I would like to use a failure route with a retry mechanism, which provides a MAX_NUM_OF_RETRIES (max number of times reaching the failure route), Which can be achieved by a simple global counter, from what I've seen, the Counters module doesn't provide a GET function in script, What is the easiest solution for this kind of behavior? Thanks a lot, Max Kemel

2 1

Timers firing too soon
by Alberto Izquierdo 04 Feb '22

04 Feb '22

Hi all, I'm using Kamailio timers to execute a series of calls with a certain delay. I have found out that from time to time, the timers are triggered too soon, sometimes they are triggered 2/3 seconds before they should. I have read in the documentation that it is normal to have a bit of delay in some of the timers, but it doesn't mention anything about this behavior. Is this normal? Thank you.

1 0

Debian kamailio.init vs. kamailio.service
by Juha Heinanen 04 Feb '22

04 Feb '22

By comparing pkg/kamailio/deb/bullseye dir kamailio.init and kamailio.service files, it appears to me that they are not equal alternatives. Is this correct interpretation? -- Juha

1 0

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