sr-users June 2020

sr-users@lists.kamailio.org

71 participants
105 discussions

insert new timestamp value to cdr table
by mohsen khashei 05 Jun '20

05 Jun '20

Hi I want to add a new column to acc table.I created a new column in the acc table of type timestamp and named it ring_time. in every call i put the ring time to a $dlg_var like this: $dlg_var(ringtime) = $Ts; then I add a extra column in config like this: modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;ring_time=$dlg_var(ringtime)") but when I try to test it always i get: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '1591361996' for column `kamailio`.`acc`.`ring_time` at row 1 (1292) Jun 5 17:29:59 kamailio /usr/sbin/kamailio[22901]: ERROR: {2 102 INVITE 105a0f4a3d99a0a5558355e54b43f4e1@192.168.1.121:5060} <core> [db_query.c:244]: db_do_insert_cmd(): error while submitting query Jun 5 17:29:59 kamailio /usr/sbin/kamailio[22901]: ERROR: {2 102 INVITE 105a0f4a3d99a0a5558355e54b43f4e1@192.168.1.121:5060} acc [acc.c:477]: acc_db_request(): failed to insert into database

2 1

How to overwrite DisplayName
by Stefano Larosa 05 Jun '20

05 Jun '20

Hi and first of all thanks for your amazing work. I have a kamailio 5.1.1 and i need to change the DisplayName before sending INVITE to another Asterisk server. I need that "Sergio"<+33558700000(a)80.169.227.18> Becomes "Simona"<+33558700000(a)80.169.227.18> I've tried just changing $fn value: $fn="Simona" Or using uac_replace_from("Simona ",""); But INVITE msg is always From: "Sergio"<sip:+33558700000@80.169.227.18;transport=UDP>;tag=1ac9e066 I've tried this options in different position in kamailio.cfg but i can figure out the problem. Do you have any suggest? Thank you, Stefano.

3 5

Kamailio under NAT
by Aidar Kamalov 05 Jun '20

05 Jun '20

Hello, I am trying to start kamailio+rtpengine+asterisk in docker with docker-compose (not in network host mode). And I faced behavior that I can't understand. When my phone(fanvil) sends a register to kamailio with his ip in contact - 192.168.88.55 (to host's ip address 192.168.88.55), kamailio 200OK answers with "received=172.18.0.1" param in Via. (172.18.0.1 it is host's docker interface, kamailio's ip is 172.18.0.4) After that my phone unregisters with Expires: 0 and sends a new register with 172.18.0.1 in contact. Is that correct? Is there any way to fix this? 2020/06/05 12:05:51.764617 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK14305253915361123;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 1 REGISTER Contact: <sip:1000@192.168.88.100:5756> Max-Forwards: 70 Expires: 3600 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.765716 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK14305253915361123;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.59da Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 1 REGISTER Contact: <sip:1000@192.168.88.100:5756>;expires=3600 Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 2020/06/05 12:05:51.779447 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK251317807114669601;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 2 REGISTER Contact: <sip:1000@192.168.88.100:5756> Max-Forwards: 70 Expires: 0 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.780479 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK251317807114669601;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.f669 Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 2 REGISTER Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 And new register: 2020/06/05 12:05:51.780287 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK2696168232620424454;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=2649930762 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 257831362525663-2610472014851(a)172.18.0.1 CSeq: 1 REGISTER Contact: <sip:1000@172.18.0.1:46142> Max-Forwards: 70 Expires: 3600 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.781208 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK2696168232620424454;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=2649930762 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.4b3b Call-ID: 257831362525663-2610472014851(a)172.18.0.1 CSeq: 1 REGISTER Contact: <sip:1000@172.18.0.1:46142>;expires=3600 Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 -- Aydar A. Kamalov

1 0

Kamailio 5.3.0 and HEPv3 oversized traffic
by Karsten Horsmann 04 Jun '20

04 Jun '20

Hi list, I saw some strange behavior in Kamailio 5.3.0 with HEPv3. My Kamailio serves webrtc and sip tls and captures that to an Homer 7.7 instance. I see only the UAC Parts of the message and not the answers from Kamailio in that scenario. Maybe I reached the max Paket size for udp. Hard to debug. In production I saw everything (but there I used and older Kamailio 5.2.x and Homer 5.5 which can handle fragment udp). Any ideas how to hunt the root cause of the problem? Cheers Karsten

5 11

kamcmd mtree.match
by Volker Denneberg 03 Jun '20

03 Jun '20

Hi, given this mtree kamcmd mtree.list siteId { tname: siteId tprefix: 00492419697994 tvalue: { 00492419697994 2388 } } { tname: siteId tprefix: 492415310752 tvalue: { 492415310752 2388 } } i wonder why this lookup fails: kamcmd mtree.match siteId 49241531075212 0 error: 500 - Invalid Parameters My expectation was that 2nd tprefix should match. I am using modparam("mtree", "db_table", "mtrees") modparam("mtree", "char_list", "0123456789+") modparam("mtree", "mt_tree_type", 0) # payload type for tree data : string Thanks in advance, Volker

3 5

Re: [SR-Users] Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Cheers Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Wednesday, June 3, 2020 9:39:36 PM To: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: RE: Request documentation of Path Flow of Kamailio Hello, you can find many presentations (which includes diagrams etc..) in the archive of past Kamailioworld conferences on kamailio.org and also on youtube. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt> Sent: Wednesday, June 3, 2020 9:38 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Henning Westerholt <hw(a)skalatan.de> Subject: Re: Request documentation of Path Flow of Kamailio One more thing if you have some diagrams that also be considered as how communication should pass from one client to another from a different network please can you send it to me and thanks. Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt<mailto:Daniel.Cassar.a100127@mcast.edu.mt>> Sent: Wednesday, June 3, 2020 9:33:19 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Subject: Re: Request documentation of Path Flow of Kamailio Thanks :) Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Sent: Wednesday, June 3, 2020 9:20:42 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Cc: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt<mailto:Daniel.Cassar.a100127@mcast.edu.mt>> Subject: RE: Request documentation of Path Flow of Kamailio Hello Daniel, thanks for your interest in Kamailio. Looking forward to see the results of your work, please free to give some feedback if its finished. ?? Being an open source project, there is many documentations available e.g. in our wiki. But most of its is more reference character and probably not something that you can directly use in an academic setting. But there is already some work outside in academia which might be useful for you. Check e.g. this diploma thesis out, which describe the implementation of Kamailio as a telephone system for CERN: https://cds.cern.ch/record/1560053/files/CERN-THESIS-2013-073.pdf?subformat… You can find more works that is using Kamailio e.g. with google scholar: https://scholar.google.com/scholar?hl=en&as_sdt=0%2C5&q=kamailio&btnG= Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Daniel Cassar Sent: Wednesday, June 3, 2020 8:08 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Request documentation of Path Flow of Kamailio Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need the path communication of how Kamailio works meaning the traffic, protocols being used and these are multicast or unicast when transfer from one site to another. Is this information available somewhere as I seem not to find it anywhere which is why I'm sending you to find out if you have these documentations that I request. I hope to hear from you soon and thanks. Sincerely, Daniel Cassar

1 0

Re: [SR-Users] Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Thanks :) Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Wednesday, June 3, 2020 9:20:42 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt> Subject: RE: Request documentation of Path Flow of Kamailio Hello Daniel, thanks for your interest in Kamailio. Looking forward to see the results of your work, please free to give some feedback if its finished. ?? Being an open source project, there is many documentations available e.g. in our wiki. But most of its is more reference character and probably not something that you can directly use in an academic setting. But there is already some work outside in academia which might be useful for you. Check e.g. this diploma thesis out, which describe the implementation of Kamailio as a telephone system for CERN: https://cds.cern.ch/record/1560053/files/CERN-THESIS-2013-073.pdf?subformat… You can find more works that is using Kamailio e.g. with google scholar: https://scholar.google.com/scholar?hl=en&as_sdt=0%2C5&q=kamailio&btnG= Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Daniel Cassar Sent: Wednesday, June 3, 2020 8:08 PM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] Request documentation of Path Flow of Kamailio Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need the path communication of how Kamailio works meaning the traffic, protocols being used and these are multicast or unicast when transfer from one site to another. Is this information available somewhere as I seem not to find it anywhere which is why I'm sending you to find out if you have these documentations that I request. I hope to hear from you soon and thanks. Sincerely, Daniel Cassar

1 1

Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need the path communication of how Kamailio works meaning the traffic, protocols being used and these are multicast or unicast when transfer from one site to another. Is this information available somewhere as I seem not to find it anywhere which is why I'm sending you to find out if you have these documentations that I request. I hope to hear from you soon and thanks. Sincerely, Daniel Cassar

1 0

Trouble loading multiple javascripts with KEMI
by Benjamín Visón 03 Jun '20

03 Jun '20

I am using the module *app_jsdt *with *KEMI *and I am having trouble loading multiple javascript files. The interpreter only notices the last file that's loaded. What I am trying to do is separate my logic so that everything is not inside the kamailio.conf.js (or the main file the contains the necessary functions if you will). modparam("app_jsdt", "load", "/etc/kamailio/file1.js") modparam("app_jsdt", "load", "/etc/kamailio/file2.js") cfgengine "jsdt" In the example above, only functions in file2 are recognized. I am aware that I can use jsdt_run or jsdt_load but from KEMI I don't see a clear way to link this with things like events and timers or replies that don't necessarily go through the *request_route*. (As a plus, I am wondering on performance on Javascript vs Python/Lua and if it's viable on high load, if anyone can throw 2 cents on that it would be helpful!) Saludos, [image: image.png] [image: image.png] <https://www.facebook.com/bvisonl> <https://twitter.com/benjaminvison> [image: image.png] <https://instagram.com/bvisonl/> Benjamín Visón / IT Engineer / Software Developer bvisonl(a)gmail.com / (829)-664-5163

1 0

Polycom looses registration within cluster scenario.
by Zhan Bazarov 03 Jun '20

03 Jun '20

Hello. We faced to an issue with polycom, namely - polycom phones looses registration within SRV/DNS based cluster scenario.. So we have round-robin cluster of SIP-proxy instances behind the same SRV/A records(each instance has the same weight). The initial registration works fine, but after the polycom sends a SUBSCRIBE request to another one sipProxy instance(because of round-robin scenario) - polycom stops sending re-register by expires which we are providing in 200Ok message... So the idea is to keep polycom located on the instance where initial register request came to. But polycom is sending SUBSCRIBE in shuffle(not to the server where registration is located) We can solve this issue by keeping registration on lines not on main SIP configuration itself, otherwise polycom looses registration from cluster at all. But it lead to rejecting some calls by polycom with '400 bad request.. We can solve it by disabling validation. But it leads to additional things like: polycom has multiple registrations on the device itself.. I mean, for example, first one registration isn’t expired, and polycom has two registrations at the same time... any ideas? How we can avoid this issue right? Thanks. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

3 4

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 June 2020