sr-users March 2023

sr-users@lists.kamailio.org

47 participants
67 discussions

Sync NONCE between registrars?
by Benoit Panizzon 16 Mar '23

16 Mar '23

Hi We have two registrars. Today I observed a CPE doing this: CPE REGISTER (no auth) => Registrar 1 => 407 Challenge NONCE A CPE REGISTER (auth to NONCE A => Registrar 2 => 407 Challenge NONCE B CPE REGISTER (auth to NONCE B => Registrar 1 => 407 Challenge NONCE A and so on. Time on both registrars is in sync (I tought NONCE was based on time) Is there a way to sync the NONCE so that both would accept authentication based on the same NONCE? Mit freundlichen Grüssen -… [View More]

2 1

How to Print Global Parameter Values and Log JSONRPC Requests Received over UDP in Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Mar '23

16 Mar '23

Dear everyone, I am wondering if there is a way to print the Global Parameter values such as debug, children, flags, etc. just to make sure that they have been changed. I am aware that I can use the `kamcmd cfg.get core debug` command to check the value of the debug parameter for example, but I am wondering if there is a way to do this in the configuration file of Kamailio. Additionally, I would like to know if there is a way to print in the log of Kamailio that it received a JSONRPC request … [View More]

1 0

Increment in number of "200" responses while calling procedures over jsonrpc with http
by sadik.oualla.mohamed＠gmail.com 16 Mar '23

16 Mar '23

Hello, Thank you very much! I've used your suggestion to accomplish some amazing things. However, I'm having an issue when calling the procedures over jsonrpc with http. I've noticed that the number of "200" responses gets incremented by 2, which is not the case when using the `*kamcmd*` tool. I suspect that this may be due to jsonrpc using *http/https*, which iscausing Kamailio to treat it as a "200" http responses code as "200" SIP codes. Could you advise on how to avoid this issue? Is … [View More]

3 8

Kamailio store call statistics
by georgy.grigoryev＠snapcom.fr 16 Mar '23

16 Mar '23

Hello everyone, I'm trying to store the call statistics in my Kamailio server. I'm using SQLite just to understand what type of information i can store. I'm using the dialog module with the classic parameters: enable_stats => 1 dlg_flag => 4 hash_size => 4096 db_url => sqlite:///usr/local/etc/kamailio/kamailio.sqlite db_mode => 1 I'm able to open the DB but everything is empty, am i using the wrong modules? or i didn't configure the server well? The goal is to store CALL CDR … [View More]

2 1

webrtc audio issue
by Arun K R 15 Mar '23

15 Mar '23

Hello, I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in the same server. I have configured kamailio to work as webrtc server and it forwards the registration to asterisk. Now when I am trying call from jssip webrtc client it reaches kamailio and route it through private interface to asterisk server. Asterisk then route it to the provider server. When i make a call asterisk server recives rtp from the provider and convert the rtp to srtp and sending back to kamailio. but … [View More]

2 1

Dispatcher behavior
by Jawaid Bazyar 15 Mar '23

15 Mar '23

Hi, is this expected behavior? I have two target hosts in a dispatcher set, 10.20.20.118 (prio 5) and 10.20.20.252 (prio 10). When I have 10.20.20.252 turned off, dispatcher still tries to send packets to that host, even when the status flags are IP. root@vs-rs01:/etc/kamailio# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { … [View More]

5 14

Error on transfer when calling from WebRTC with TOPOH
by Arsenijs Kabihno 15 Mar '23

15 Mar '23

Hello, We have an issue which appears when TOPOH module is enabled, call is made from WebRTC, then transferred by calee: Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) INFO: {2 2 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK66807520, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address via-branch=extra rtcp-mux-offer … [View More]generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) NOTICE: {2 2 INVITE g2rgvnioadg8m7l6i7k6} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881327;method=INVITE;from_tag=p9qqi7hrua;to_tag=ea7d2850-3e76-4c11-821c-a0d1e3cbb011;call_id=g2rgvnioadg8m7l6i7k6;code=200;reason=OK;src_user=92010;src_domain=10.181.0.1;src_ip=10.5.0.145;dst_ouser=27221;dst_user=27221;dst_domain=10.181.0.1 Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 31(37) INFO: {1 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} <script>: RTPENGINE branch_id:0 ruri: sip:97093@10.128.150.67:46058;rinstance=d403efb922420649;transport=UDP, method:BYE, status:<null>, extra_id: <null>, rtpengine_delete Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 28(34) NOTICE: {2 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881338;method=BYE;from_tag=0c29be45-6aff-4d4e-8503-650893a5a5ac;to_tag=b1a8a401;call_id=e034a187-0e4d-4c4e-ab4b-f2e5f1281330;code=200;reason=OK;src_user=22628;src_domain=10.5.2.192;src_ip=10.5.2.192;dst_ouser=97093;dst_user=97093;dst_domain=10.128.150.67 Mar 15 11:55:39 ap07stvp.xxxxxxx.net docker[2062]: 32(38) INFO: {1 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: sip:jgqtuev6@2idi5cucan2v.invalid;transport=ws;ob, method:INVITE, status:<null>, extra_id: <null>, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address direction=priv1 direction=pub1 rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 27(33) INFO: {1 4728 INVITE 133ccd4c-031b-4d10-b073-745f35433132} <script>: RTPENGINE branch_id:0 ruri: sip:0037126048570@193.138.189.220, method:INVITE, status:<null>, extra_id: z9hG4bKPjedcf2d96-01cd-43fd-bed0-4e601e3e542f0, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 via-branch=extra Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 50(56) INFO: {2 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bKPj1cd87bda-7862-4ca9-8475-ac6b342c45ef0, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=pub1 direction=priv1 SIP-source-address via-branch=extra rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/resolve.c:1730]: sip_hostport2su(): could not resolve hostname: "2idi5cucan2v.invalid" Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/forward.c:516]: forward_request(): bad host name 2idi5cucan2v.invalid, dropping packet Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} sl [sl_funcs.c:418]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL) We have the following setup: WebRTC client behind load-balanser -> kamailio -> asterisk. When call is transferred, asterisk makes re-invite. If TOPOH is disabled, it happens this way and issue is not present: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> WebRTC ACK example: 03/09/2023 09:35:48.126 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:98dn3met@0j62ln34srjn.invalid;transport=ws;ob;alias=10.5.0.145~27562~6 SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPj1d11dae2-10bf-466e-9221-5710f801d725 From: <sip:27221@10.181.0.1>;tag=b6b73135-efe1-4d86-bbc8-bf4313130eb5 To: <sip:92010@10.181.0.1>;tag=namgu127j9 Call-ID: t0hmmk50h5i4ok88asjj CSeq: 28632 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Route: <sip:10.181.0.1:4444;transport=ws;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 If TOPOH is enabled, kamailio doesn't pass this ACK to WebRTC: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> -> -> ACK example: 03/09/2023 09:33:03.544 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:10.0.0.1;line=Bn7Ipzh8eb8DxMSFyk.rWzitxzVFBoirxRG5xYeOxb7hpoXTlMGDBbuTCk3oBDAqlH** SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPjf00fc541-f011-4e52-b63a-2c569478ca39 From: <sip:27221@10.181.0.1>;tag=7d402872-7355-4269-b912-e52ee28ed415 To: <sip:92010@10.181.0.1>;tag=t0q4g9am9g Call-ID: t0hmmetickvlvn5k7nco CSeq: 25565 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=t0q4g9am9g;did=4c8.a182;rtp=bridge;rtp=ws> Route: <sip:10.0.0.1;line=Bn7IpzdIwzd8Vf8Iwzd9WkyFWkAFBELJBo.qBYyuCoVgBz1uxn8gx61uxn8geYXOeD3FV6dFeD7Oxj7YpnX5ekFFlDHJljd8VzATC60ulYi5ebC7poiFBk3oBI**> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 We have an assumption that issue might be related to the way how TOPOH handles alias in contact. Did someone have encountered something like this? [View Less]

1 0

Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Adil Mafzool Zaka 15 Mar '23

15 Mar '23

Hi I am using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected … [View More]

2 1

Use Kamailio and RTPEngine to play wav file as early media to announce a server maintenance
by jhayhoe＠liquidweb.com 15 Mar '23

15 Mar '23

I am looking to configure Kamailio and RTPEngine to play a wav file for any new calls during a scheduled asterisk server maintenance. I have set up the following within route{} to send the call to a early_media route based on a variable that I can set on the server using kamcmd. route { ... if ($sht(config=>maint)=="1") { if (is_method("INVITE") && !has_totag()) { xlog ("L_INFO","Ongoing maintenance. Call-ID:$ci | Method: $rm"); … [View More]

3 3

stuck with webrtc on kamailio
by Arun K R 14 Mar '23

14 Mar '23

hello, I have issue of no voice on kamailio configured for webrtc. I have configured rtpengine with public and private interface. My kamilio has two interface of pubic and private ip. Somebody has suggested me that there is no ice trickling happens. I am using jssip webrtc client. Kindly suggest me or help me with this. I have send another email for same issue to this group and didnt get any reply. Kindly help me

1 0

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 March 2023