September 2019 - sr-users

Kamailio stop to process incoming SIP traffic via TCP.
by Kristijan Vrban 16 Jan '20

16 Jan '20

Hi kamailios, i have a creepy situation with v5.2.1 stable Kamilio. After a day or so, Kamailio stop to process incoming SIP traffic via TCP. The incoming TCP network packages get TCP-ACK from the OS (Debian 9, 4.18.0-15-generic-Linux) but Kamailio does not show any processing for the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via UDP is working just totally fine. When i look via command "netstat -ntp" is see, that the Recv-Q get bigger and bigger. e.g.: Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED 31347/kamailio After Kamailio restart, all is working fine again for a day. We have maybe 10-20 devices online via TCP and low call volume (1-2 call per minute). The only settings for tcp we have is "tcp_delayed_ack=no" How to could we debug this situation? Again, no error, no warings in the log. Just nothing. Kristijan

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WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

Hello, I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking: INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0 Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060> Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE User-Agent: Proxy Date: Wed, 03 Apr 2019 17:11:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Content-Type: application/sdp Content-Length: 596 Server: SIP Proxy and then WebRTC app is replying with 200 OK: SIP/2.0 200 OK Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Proxy-WEBRTC Content-Type: application/sdp Content-Length: 901 and finally, Kamailio is trying to send this ack through UDP protocol: ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes> Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 ACK User-Agent: Proxy Content-Length: 0 If i'm trying to force it through TLS, i'm receiving error: get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443, to udp:22.22.22.22:23317) Can someone point me in the right direction, please? Thank you.

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rtpproxy does not response, disable it
by Zodiac 03 Jan '20

03 Jan '20

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

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lookup(aliases) issues with 5.2
by igor.potjevlesch＠gmail.com 13 Dec '19

13 Dec '19

Hello! I have notice some issues with lookup(aliases) since I upgraded to 5.2 from 5.1: First with kamctl: If this default command is executed: ctl_cmd_run ul.add "$USRLOC_TABLE" "$OSERUSER@$OSERDOMAIN" "$2" \ "$UL_EXPIRES" "$DEFAULT_Q" "$UL_PATH" "$UL_FLAGS" "$BR_FLAGS" "$ALL_METHODS" as $ALL_METHODS is not defined, there is a huge value in methods and then lookup(aliases) fails with -2. I have to update kamctl script and put -1 ("-1") instead of the variable. So, now, methods is a NULL value in DB and my alias is not ignored. But I use to make my alias like this: sip:number@blabla_1.local Which is no longer works: ERROR: registrar [common.c:62]: extract_aor(): failed to parse AoR [sip:number@blabla_1.local] Any idea of the difference since 5.2 with this behaviour? Without the underscore, the AoR seems to be valid but I need to support underscore as before. Regards, Igor.

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Re: [SR-Users] Question on Database Version Table
by JR Richardson 15 Nov '19

15 Nov '19

Yep version_table Set the name of the table holding the table version. Useful if the proxy is sharing a database within a project and during upgrades. Default value is “version”. Example of usage: version_table="version44" That's what I was looking for, perfect! Yes, I added accountcode and notes to my table for internal use, was just making notes on columns I need on the new version upgrade. Thanks. JR On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > like Alex said, you can have two version tables, one for each versions > of kamailio you play with and use in config the core parameter to set > the name of that table. > > I am actually writing to say that address table never had accountcode > and notes columns. Likely they were added in your deployment for other > purposes or either the default column names were changed. The default > column names are listed at: > > - > https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address > > These were the same for rather long time, iirc the recent version number > changed mainly due to with modifications done to the size, not due to > new columns. > > Cheers, > Daniel > > On 18.06.19 20:50, JR Richardson wrote: > > Hey All, > > > > I'm doing some upgrades and noticed database table version numbers > > have increased. > > > > database table adjustments: > > | address | production ver 3 | upgrade 5.2 ver 6 | need to add > > accountcode and notes column > > | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column > > | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add > > ruri_pattern and priority column > > > > I run mysql master/slave so I'm concerned about the master database > > not having the correct version numbers or can I just update the > > version number in the table when I do the upgrade of the kamailio > > versions? > > > > Thanks. > > > > JR

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Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

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Kamailio Developers Meeting, Nov 14-15, 2019, in Dusseldorf
by Daniel-Constantin Mierla 09 Oct '19

09 Oct '19

Hello, Kamailio SIP Server project is organizing another meeting of its developers during November 14-15, 2019, hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join. Please note we have a limited capacity of 20 seats in the meeting room. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

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Branching 5.3
by Daniel-Constantin Mierla 02 Oct '19

02 Oct '19

Hello, if there are no strong opinions against, then it is time to branch 5.3 on git repository. I am considering to do it by the end of the day on Wednesday, October 2, 2019. After that, the master will be again open for new features, while 5.3 branch will continue to get the only fixes and improvements to documentation and tools. 2-3 weeks later we should release v5.3.0, the first stable version in 5.3.x series. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

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BLF 'NOTIFY' not working
by Petarr Jones 02 Oct '19

02 Oct '19

Hi I am using Kamailio 5.2 with the following PRESENCE modules and parameters: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_reginfo.so" modparam("presence", "db_url", DBURL) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "disable_bla", 0) modparam("presence_xml", "integrated_xcap_server", 1) I have a Yealink T28P phone which has two lines/extensions successfully registered with Kamailio 5.2. When I setup the BLF light to monitor one of the extensions, I get the correct SUBSCRIBE record but the NOTIFY record is sent over and over again until it fails. ----------------------------------------------------------------------------------------------------------- NOTIFY sip:101@192.168.1.123:5062 SIP/2.0 Via: SIP/2.0/UDP 45.45.45.45;branch=z9hG4bKd16c.03b5bc97000000000000000000000000.0 To: <sip:101@45.45.45.45>;tag=1366853594 From: <sip:104@45.45.45.45>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-e6c3 CSeq: 2 NOTIFY Call-ID: 594386350(a)192.168.1.123 Content-Length: 0 User-Agent: kamailio (5.2.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:45.45.45.45:5060;transport=udp> ----------------------------------------------------------------------------------------------------------- When I register the phone with Freeswitch instead and run the same test, the NOTIFY works and I get a much longer NOTIFY message ----------------------------------------------------------------------------------------------------------- NOTIFY sip:100@192.168.1.123:5064 SIP/2.0 Via: SIP/2.0/UDP 66.66.66.66;rport;branch=z9hG4bK1mXej9r2605UB Route: <sip:55.55.55.55:5064>;transport=udp Max-Forwards: 70 From: <sip:102@66.66.66.66>;tag=xUoVghM5syHc To: "100" <sip:100@66.66.66.66>;tag=1720892352 Call-ID: 2583553127(a)192.168.1.123 CSeq: 1178415701 NOTIFY Contact: <sip:102@66.66.66.66:5060> User-Agent: FreeSWITCH-mod_sofia/1.10.1-release-12-f9990221e6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1800 Content-Type: application/dialog-info+xml Content-Length: 149 xml version="1.0"?> dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:102@66.66.66.66"> dialog-info> ----------------------------------------------------------------------------------------------------------- Should the message coming from Kamailio contain more data or is that not the issue. Should I be including other modules also? All guidance would be appreciated.

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Call limit through shared database
by Olli Attila 01 Oct '19

01 Oct '19

Hello, I have 2 kam instances using one shared database and both kam servers running the same routing logic code. Kamailio servers are configured to use the same database as dialog backend. I am limiting the concurrent dialogs in the following way but it seems that kamailio is using its own memory to read profile size rather than the db. Second kam is not aware of the dialog profiles that the first kam wrote to the shared db. Is there a way to get multiple kam instances to read profile size from the same database when calling the get_profile_size function? modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) route[LIMIT_CALLS] { if (!dlg_isflagset("1")) { # $td – gateways, $fu - user if (get_profile_size("concurrent_calls", "$fu", "$avp(calls)")) { if ($avp(calls) >= 3) { xlog("L_INFO", "Concurrent calls $fu at limit"); send_reply("503", "Calls limit reached"); exit; } else { dlg_manage(); dlg_setflag("1"); set_dlg_profile("concurrent_calls", "$fu"); } } } } Cheers, Olli

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