- sr-users - kamailio.org

rtpengine - optional srtp
by Daniel-Constantin Mierla 06 Oct '17

06 Oct '17

Hello, wondering if anyone has a quick answer to spare some time searching the web or source code -- is there a way to offer optional SRTP (just SDES is also fine) in SDP with RTPEngine? The use case is when the target device is not yet known to support (or not) SRTP. There are couple of ways in specs or practices (e.g., RFC5939/6871, or advertising crypto attributes on RTP/AVP stream, or offering two streams one RTP and one SRTP), I am looking to find if it is possible to do it with RTPEngine and which of the options are supported/what are the parameters for doing it... Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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SIP -
by Subba Krishna MG 06 Oct '17

06 Oct '17

Hello, How do i include Expires header for 200 OK message? One of the manual says this.. how do i modify my existing config file to include this? I am using : /usr/local/etc/kamailio/kamailio.cfg for configuration. Rgds, MGSK 1.4.9. expires_offset (int) The value that should be subtracted from the expires value when sending a 200OK for a publish. It is used for forcing the client cu send an update before the old publish expires. *Default value is "0".* *Example 1-9. Set expires_offset parameter* ... modparam("presence", "expires_offset", 10)

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Re: [SR-Users] Subscribe Reinvites totag
by Soltanici Ilie 06 Oct '17

06 Oct '17

I found that the problem is because of this header:

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A "success_route" as counterpart to "failure_route" available?
by Kristijan Vrban 05 Oct '17

05 Oct '17

Hello, i need a "success_route" which does the same as "failure_route", but only for positiv 2xx reply. Is something like this available? Background: i forward REGISTER to external registrar server, and only what to save the location on positiv reply to REGISTER. Without doing the authentication myself. I tried to save the location in onreply_route, but then the contact of the external registrar server is saved. So i need a status check in the REQUEST_ROUTE. Best Kristijan

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call disconnects after 32 seconds
by Adesh Pandey 05 Oct '17

05 Oct '17

Hi, I have set up the Kamailio as load balancer using dispatcher module but all the calls get disconnected after 30 seconds (call is forwarded to Asterisk Box and answered as well). I have used example configuration from the below URL: http://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#idp51014772 Please help me to understand what is wrong and what shall I do it to fix. Thank You!! Adesh P. -- Adesh Pandey Sr. Software Developer [image: phone] +91 92129 92129 Ext: 56 [image: moible] +91 8527384897 [image: Profile Pic] [image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter] <https://twitter.com/MyOperator>[image: Twitter] <https://www.youtube.com/user/MyOperatorCo>[image: Linkedin] <https://www.linkedin.com/company/voicetree-technologies>[image: Google+] <https://plus.google.com/+MyoperatorCo/posts> [image: Logo] <http://myoperator.co/>

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Parse Error (event_route)
by James DePhillip II 04 Oct '17

04 Oct '17

I am trying to setup the internal XCAP service and kamailio will not start. The messages log states "kamailio: CRITICAL: <core> [cfg.y:3372]: yyerror_at(): parse error in config file //etc/kamailio//kamailio.cfg, line 1009, column 1-11: syntax error" This is the line "event_route[xhttp:request] {" from the config file. I am not sure why this is happening any ideas? Thanks,Jim

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issue when trying to start kamailio using db_postgres and dialplan modules
by David Escartín 04 Oct '17

04 Oct '17

hello all we are having an issue when trying to start kamailio using these modules and a kamailio version higher than 4.2 we install libpq5 and libpq-dev and we always have this error 0(10800) ERROR: <core> [core/sr_module.c:582]: load_module(): could not open module </usr/local/kamailio-5.0/lib64/kamailio/modules/db_postgres.so>: /usr/local/kamailio-5.0/lib64/kamailio/modules/db_postgres.so: undefined symbol: PQconnectdbParams do you know which lib or packet would we miss? we are trying to install it on centos6, ubuntu 10 or centos7 best regards david

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Re: [SR-Users] Manipulate From URI username
by Gertjan Wolzak 04 Oct '17

04 Oct '17

Thank you guys. I am using the s.substring solution, as the uac_replace is done in another route part. Rgds, Gertjan

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Manipulate From URI username
by Gertjan Wolzak 04 Oct '17

04 Oct '17

Hello Kamailions, Im trying to manipulate the username from the from uri, but not having any luck, I must be doing something wrong. I receive a from uri username that is prefixed with 888, now I am trying to remove the triple 8. I have configured the following: line 1090 if($fU=~"^888"){ 1091 strip(3); 1092 } 1093 route(GETGROUP); But I do not get any result, the debug shows the following: 29(9610) exec: *** cfgtrace:request_route=[MOMANX] c=[/usr/local/etc/kamailio/kamailio.cfg] l=1093 a=16 n=if 29(9610) exec: *** cfgtrace:request_route=[MOMANX] c=[/usr/local/etc/kamailio/kamailio.cfg] l=1091 a=45 n=strip 29(9610) exec: *** cfgtrace:request_route=[MOMANX] c=[/usr/local/etc/kamailio/kamailio.cfg] l=1093 a=5 n=route Does anyone see what I am doing wrong? Help would be appreciated. Rgds, Gertjan Wolzak

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"python_exec" not work
by ‪mahsa _a_‬ ‪ 03 Oct '17

03 Oct '17

Iwant to work with "app_python" module,i set “script_name”,“mode_init_function”, “child_init_function” and “python_exec”in kamailio.cfg ,but I have a problem in “python_exec”section. This section not work in correctly, please guide me.thanks

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Debian Stretch
by Jurijs Ivolga 03 Oct '17

03 Oct '17

Hi Guys, I recently tried to use Debian Stretch in production and it didn't went well. On load(500k-700k SIP messages per day) I get a problem that at some point there was a pike of CLOSE_WAIT connections(up to 2k of CLOSE_WAIT connections) and no new connections was possible, I tried to figure out where problem is, but I didn't found anything in system or Kamailio logs, Kamailio just stopped to receive traffic via TCP, but UDP continued to work. Test environment where was no load I didn't faced such issue. Just curious is there anybody who are using Debian Stretch in production without issues? Jurijs

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event_route[usrloc:contact-expired] not called after contact expires
by Yuriy Gorlichenko 02 Oct '17

02 Oct '17

Hi I added this eventroute for manipulate temprorary entires in my subscriber table but it is not called in my cfg. May be i didn't setted up in scf but didnf found any special params for this. I use kamailio 5.0.3 version: kamailio 5.0.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 09:39:55 Sep 8 2017 with gcc 4.9.2 My registrar and usrloc modules setted up as usual modparam("registrar", "method_filtering", 1) modparam("registrar", "max_expires", 3600) modparam("registrar", "gruu_enabled", 0) modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) thx for advice

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sigcomp usage with kamailio
by Ram 02 Oct '17

02 Oct '17

Hi, I need to enable sigcomp with kamailio. Can any one guide me how proceed with that. And its ok to integrate any third party tools like libsigcomp for the same. Regards Raman

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Kamailio htable mod-init route: Main process exited before writing to pipe
by George Diamantopoulos 02 Oct '17

02 Oct '17

Hello all, I've been trying to utilise event_route[htable:mod-init] in order to load information from the database into an htable on startup. However, Kamailio exits with an error for no obvious (to me) reason. Here's the log output: DEBUG: htable [htable.c:226]: child_init(): rank is (-127) DEBUG: htable [htable.c:259]: child_init(): executing event_route[htable:mod-init] (2) DEBUG: <core> [core/parser/msg_parser.c:600]: parse_msg(): SIP Request: DEBUG: <core> [core/parser/msg_parser.c:602]: parse_msg(): method: <OPTIONS> DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): uri: < sip:you@kamailio.org> DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): version: <SIP/2.0> DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=2 DEBUG: <core> [core/parser/msg_parser.c:491]: parse_headers(): Via found, flags=2 DEBUG: <core> [core/parser/msg_parser.c:493]: parse_headers(): this is the first via ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe And here's the routing block in kamailio.cfg. What I need to do is get the username for all non-expired contacts in the "location" table, perform a query on another database for each one of them and store the result in an htable: event_route[htable:mod-init] { sql_pvquery("ck", "select username from location where now() <= expires", "$avp(registered_ng)"); $var(i) = 0; while ( $(avp(registered_ng)[$var(i)]) != $null ) { sql_pvquery("ca", "select property from tablename where username = $(avp(registered_ng)[$var(i)])", "$var(ng_property)"); $sht(isnguser=>$var(ng_property)) = 1; $var(i) = $var(i) + 1; pv_unset("$var(ng_property)"); } pv_unset("$var(i)"); } Any hints as to what I might be doing wrong? Thanks! Best regards, George

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Regarding ASYNC module in kamailio - handling SIP replies
by Rajagopal Maddi 02 Oct '17

02 Oct '17

Hi, We are checking the kamailio source code for the ASYNC module. The system is working for REQUEST_ROUTE, but not for REPLY_ROUTE. We added the ANY_ROUTE in the cmd_export_t, but still the async_route is not able to suspend for SIP Reply (For example, 401 - Unauthorized). Can we get some help on how to perform this. Some hints on how to perform would really help our development. static cmd_export_t cmds[]={ {"async_route", (cmd_function)w_async_route, 2, fixup_async_route, 0, REQUEST_ROUTE|FAILURE_ROUTE|ANY_ROUTE}, {"async_sleep", (cmd_function)w_async_sleep, 1, fixup_async_sleep, 0, REQUEST_ROUTE|FAILURE_ROUTE|ANY_ROUTE}, {"async_task_route", (cmd_function)w_async_task_route, 1, fixup_async_task_route, 0, REQUEST_ROUTE|FAILURE_ROUTE|ANY_ROUTE}, {0, 0, 0, 0, 0, 0} }; Regards, RajaGopal Maddi

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Hello
by Subba Krishna MG 29 Sep '17

29 Sep '17

Hello, i am using Kamailio for SIP Presence server. When ever the Presence PUBLISH is received on kamailio, kamailio is sending 200 OK with out Expires IE. I want Kamailio to send Expires IE with 200 OK. How do i modify it. Can you please help me in this regard. Thank u MGSK

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Retrieve remote IP and port
by Iskren Hadzhinedev 29 Sep '17

29 Sep '17

Hi list, How can I reliably get the sender’s IP address? |$si| and |$sp| are returning the server IP and Port. I also tried using |$Ri| and |$Rp| but it yields the same results. Inspecting the packet shows the sender’s true IP:Port pair in the |Via:| header, but the |From:| and |To:| contain the kamailio server’s public IP address. Kind regards, -- /Iskren Hadzhinedev/

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Perl module missing in kamailio 5.0.3
by Ginhoux, Patrick 29 Sep '17

29 Sep '17

Hi, I'm thinking to upgrade my kamailio 5.0.1 running on a Rhel 7.1 server to 5.0.3 . Looking at the repository for Rhel7 to download the rpm, I don't find the perl rpm. For 5.0.1, it is kamailio-perl-5.0.1-1.1.x86_64.rpm. Can you help to get it for 5.0.3 ? Thanks in avance. Cordialement Patrick GINHOUX

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how to debug kamailio c code
by Sergey Safarov 28 Sep '17

28 Sep '17

I have issue with kazoo module call sprintf <https://github.com/kamailio/kamailio/blob/master/src/modules/kazoo/kz_pua.c…> not work properly. body variable contains values like \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 Must be XML string <?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" entity="2496"> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t8No1hQnp65U33jS1oqFMgxbIRPq-iFP"><status><basic>open</basic></status></tuple><note xmlns="urn:ietf:params:xml:ns:pidf">On the Phone</note><dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="1"><rpid:activities><rpid:on-the-phone/></rpid:activities><dm:note>On the Phone</dm:note></dm:person></presence> But i cannot find/understand where is must be executed conversion hex2string I want use gdb inspect memory vars but i cannot. Please advice how to do it.

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Re: [SR-Users] Question about Kamailio Performance
by Jiankai Zhu 28 Sep '17

28 Sep '17

Thank you guys for your quick response. I am new to Kamailio so all your information are very helpful for me. My setting: children=8 in kaimailio.cfg . I did not change it and leave it by default on my 4 cores system. My thought: I think it may related to Mysql since cpu and iowait is low and the problem happens only when I add more subscribers to mysql database. I will try according to : 1: Dmitri combination of mysql and htable as mysql cache 2: Mathias tune MySQL server 3: I think ram database should help on this. But it seems these is little document on redis. I am looking into SQLite3 memory mode but if there is any related reference / link will be appreciated. Thanks with regards, David David 2017年9月28日下午02:59，Daniel-Constantin Mierla <miconda(a)gmail.com> 写道： Hello, it seems that interaction with mysql is a bottleneck. What is the value of children parameter in kamailio.cfg? Cheers, Daniel On 28.09.17 04:00, Jiankai Zhu wrote: My goal: make sure when using kamailio as engine I can achieve: on systems with 4GB memory, Kamailio can serve a population over 300 000 online subscribers <from: https://www.kamailio.org/w/features/ > Test bed: CPU: 4 cores , 4G RAM Kamailio 5 with Mysql. Test source：another server running SIPp 10k subscriber in Mysql . 6000 Register Per Second , reuse 10k user to register , it works fine. If register 50 every second for per user it seem 300 k subscribers should work. (about 300% cpu used of totally 400% cpu. All requests finished without retransmit and fail. ) add 90k subscriber into Mysql . 100k subscirbers in Mysql . 1000 Register Per Second , a lot of failure and can not work . (cpu are not used much at that time. IO is not high.) My question : What is the root cause of bad performance when there is 100k subscribers' data in the Mysql database. How can I deploy a kamailio with 300k subscriber in database, and support 6000 register per second at the same time ? Maybe use ram database instead of Mysql? Thanks with regards, David _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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mysql installation error
by Timothy oladapo olawuyi 28 Sep '17

28 Sep '17

Dear All, Please help me with error messages. Am trying to configure kamailio MYSQL as DBEGINE base on the tutorial available in the SIP ROUTING WITH KAMAILIO book, when I executed the /usr/local/sbin/kamdbctl create and got the Error messages below. ERROR: could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' Here this the version of kamailio I installed. ./kamailio -I. Print out of kamailio internals Version: kamailio 4.4.6 (x86_64/linux) 75f13d Default config: /usr/local/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_LISTEN=16 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 75f13d Compiled with: gcc 4.9.2 Compiled on: 10:56:51 Sep 28 2017 Thank you for flying kamailio! Regards Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 |Skype: daptims | <htttp://www.pethahiah.com/> htttp://www.pethahiah.com

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How to do replace_body_all with Kemi python version kamailio 5.0.x
by Muhammad Zaka 28 Sep '17

28 Sep '17

Hi We are experiencing issue with python script KSR. textops.replace_body_all using branch 5.0.2: - Example Code: KSR.textops.replace_body_all("a=sendonly", "a=sendrecv") It looks like that the module textops and textopsx are not available. Using $rb or $msg(body) pseudo variables isn't working as they are readonly. Is there any other way to replace the body? Many Thanks Regards Muhammad Zaka

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CentOS 7 TLS Unable to set the memory allocation functions
by Ryan Wagoner 28 Sep '17

28 Sep '17

I had this issue with CentOS 7.3 and chalked it up to the openssl 1.0.1e bug mentioned on the list https://lists.kamailio.org/pipermail/sr-users/2013-August/079381.html However I just updated to CentOS 7.4 with openssl 1.0.2k-8 and the issue still exists with Kamailio 4.4.6-2.2. I've tried moving the loadmodule "tls.so" to the top as well. ERROR: tls [tls_init.c:526]: tls_pre_init(): Unable to set the memory allocation functions ERROR: tls [tls_init.c:528]: tls_pre_init(): libssl current mem functions - m: 0x7fc56d19c0c0 r: 0x7fc56d19c5a0 f: 0x7fc56d19c4c0 ERROR: tls [tls_init.c:530]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) Does anybody have TLS working with RHEL 7 or CentOS 7? Thanks, Ryan

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Question about Kamailio Performance
by Jiankai Zhu 28 Sep '17

28 Sep '17

My goal: make sure when using kamailio as engine I can achieve: on systems with 4GB memory, Kamailio can serve a population over 300 000 online subscribers <from: https://www.kamailio.org/w/features/ > Test bed: CPU: 4 cores , 4G RAM Kamailio 5 with Mysql. Test source：another server running SIPp 10k subscriber in Mysql . 6000 Register Per Second , reuse 10k user to register , it works fine. If register 50 every second for per user it seem 300 k subscribers should work. (about 300% cpu used of totally 400% cpu. All requests finished without retransmit and fail. ) add 90k subscriber into Mysql . 100k subscirbers in Mysql . 1000 Register Per Second , a lot of failure and can not work . (cpu are not used much at that time. IO is not high.) My question : What is the root cause of bad performance when there is 100k subscribers' data in the Mysql database. How can I deploy a kamailio with 300k subscriber in database, and support 6000 register per second at the same time ? Maybe use ram database instead of Mysql? Thanks with regards, David

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Subscribe Reinvites totag
by Soltanici Ilie 27 Sep '17

27 Sep '17

Hi All, I have system setup where there are few Asterisk servers behind Kamailio. Everything is working properly, the issue only is with the Incoming Subscribe's with to-tag from endpoints. Below you can see the invite: ---------------------------------------------------------------------------------------------------------- 2017/09/27 19:40:29.995598 192.168.2.33:5060 -> 10.25.23.1:5060 SUBSCRIBE sip:10000211@10.25.23.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.33:5060;branch=z9hG4bK-fc65f67d From: "Cisco" <sip:211@10.25.23.1>;tag=258a856c69c288 To: <sip:10000211@10.25.23.1>; tag=02cca2b6-1d9b-4edc-b1de-fdd78f8f9ab0 Call-ID: 4ce04a70-6128897b(a)192.168.2.33 CSeq: 1003 SUBSCRIBE Max-Forwards: 70 Route: <sip:10.25.23.1;lr;r2=on;ftag=258a856c69c288;nat=yes>, <sip:10.1.1.101;lr;r2=on;ftag=258a856c69c288;nat=yes> Proxy-Authorization: Digest username="211",realm="10.25.23.1",nonce="Wcvq0VnL6aUbQ3c+nZa5xbh8KMVoOwAq",uri="sip:10000211@10.25.23.1",algorithm=MD5,response="155f68d66b3c6ad035f7db1ec5732020" Contact: "Cisco" <sip:211@192.168.2.33:5060> Accept: application/dialog-info+xml Expires: 1800 Event: dialog User-Agent: Cisco/SPA504G-7.6.2 Content-Length: 0 ---------------------------------------------------------------------------------------------------------- Receiving such Invite Kamailio are routing to RELAY route (from WITHINDLG) and showing such error: ERROR: rr [loose.c:107]: find_first_route(): failed to parse Route HF It seems that Kamailio cannot find such Dialog, but why is happening? where is it destroyed? How can i solve this? Route (WITHINDLG) is as default, below you can see it as well. ----------------------------------------------------------------------------------------------------------- route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if ( is_method("ACK") ) { route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); } exit; } ----------------------------------------------------------------------------------------------------------- Thank You.

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Next IRC devel meeting
by Daniel-Constantin Mierla 27 Sep '17

27 Sep '17

Hello, to sync properly for the next major release (v5.1.0) and ongoing development, I propose an IRC devel meeting for next week, on Wednesday, Sep 27, 2017. An alternative would be the following day, Sep 28, or if there are many devs that want to attend and cannot do it these days, we can look at another date. Just propose new day and time. I created a wiki page to collect the topics that are wanted to be discussed: - https://www.kamailio.org/wiki/devel/irc-meetings/2017b Fell free to add there or reply to the mailing list with what you think it is relevant to discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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dialog:active_dialogs counter is steadily increasing
by Andrew Pogrebennyk 27 Sep '17

27 Sep '17

Hi, using kamailio 4.4.5, we have noticed that the active_dialogs counter is not accurate and there seems to be some kind of "leak" present at least on higher loaded systems. For example, here active_dialogs was reporting just a bit over 1000 dialogs, but the numbers of dialogs confirmed and not just do not sum up together to that value: # grep -c 'state:: 1' dialogs_20170919_1712.txt 5 # grep -c 'state:: 2' dialogs_20170919_1712.txt 62 # grep -c 'state:: 3' dialogs_20170919_1712.txt 0 # grep -c 'state:: 4' dialogs_20170919_1712.txt 531 # grep -c 'state:: 5' dialogs_20170919_1712.txt 27 What could be the possible reasons or issues that lead to that? We are thinking now about switching statistics collection to: kamctl fifo dlg_list | grep -c 'state:: 4' or reading the count directly from the database, as it seems to only write the dialogs with state 4 to mysql and those numbers seem OKish. Any ideas? Regards, Andrew

3 6

How to do msg_apply_changes with Kemi python version 5.0.x
by Muhammad Zaka 27 Sep '17

27 Sep '17

Hi We are experiencing issue with python script KSR. textopsx.msg_apply_changes using branch 5.0.2: - Example Code: KSR.textopsx.msg_apply_changes() Error: Sep 22 15:25:19 sbc99-kam-dev /usr/sbin/kamailio[23141]: PYT {INVITE}: ERROR: app_python [python_support.c:142]: python_handle_exception(): python_exec2: Unhandled exception in the Python code:#012Traceback (most recent call last):#012 File "/etc/kamailio/kamailio-kemi-python.py", line 64, in ksr_request_route#012 if self.ksr_route_reqinit(msg)==-255 :#012 File "/etc/kamailio/kamailio-kemi-python.py", line 173, in ksr_route_reqinit#012 KSR.textopsx.msg_apply_changes()#012AttributeError: 'module' object has no attribute 'textopsx' Is anybody have any idea how to workaround and apply message changes in python? Many Thanks Regards Muhammad Zaka

2 1

topos and record-route headers
by Giuseppe Pandolfi 27 Sep '17

27 Sep '17

Hi, I'm trying to use topos module but I have a problem with record-route header in re-invite for on hold. Uac is a subscriber of sipwise platfrom (vm open source version); my kamailio version is: /# kamailio -v// //version: kamailio 5.1.0-dev7 (ppc/linux) // //flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES// //ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB/ and uas is a subscriber of kamailio. The call flow is: UAC Sipwise Kamailio UAS | | (topos active) | |-------- Invite -------->|---------- (1)Invite ----->|------- (2)Invite ----->| |<------- 180/200 ------|<--- (4)180/(4)200 ----|<------- 180/200 ------| |---------- Ack --------->|----------- (5)Ack ------>|-------- (6)Ack ------->| |--- Invite (hold) ----->|-- (7)ReInvite(hold)-->|-- (8)Invite (hold)--->| |<--------- 200 ----------|<--------- (9)200 -------|<--------- 200 ----------| | corrupt | corrupt | | |---- No Ack Sent-------|---- No Ack Sent-------|---- No Ack Sent-------| *The problem:* the 200 ok of re-invite is corrupt by topos, it hasn't via header and contact isn't modified, then the uac can't send the ack. In the kamailio trace I see the ERROR below: *E!: U 22/09/2017 10:24:19:880 proxy0: ERROR: topos [tps_storage.c:1254]: tps_db_update_dialog(): no valid dlg uuid* *The questions are:* Can I use topos for this scenario? Is it possible that topos have bug when receive a re-invite with record-route headers? Infact, in simple scenario without record-route (call and hold between tow subscriber) everything works! *My opinion is*: when topos receive a request with record-route generates from record-route the new aleg and bleg tags (atpsh-xxxxx, btpsh-yyyyyy) but don't save their in internal database if the request is internal at dialog. Then a reintive whit record-route headers, generate new aleg/bleg tags but they aren't in database, then the response 200 ok it isn't elaborate. below messages details, attached kamailio log and pcap trace. tanks in advance for support, Giuseppe* * * * *(1)Invite:* INVITE sip:100@192.168.110.79:5060;transport=udp SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79> CSeq: 10 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri P-Asserted-Identity: <sip:390735594284@192.168.110.132> Content-Type: application/sdp Content-Length: 276 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> [sdp] *(2)Invite:* INVITE sip:100@192.168.1.73:5080;line=4d3148ee1d889aa SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK1a43.c9b996baf80be68d288d2de06cd9b212.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79> CSeq: 10 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri P-Asserted-Identity: <sip:390735594284@192.168.110.132> Content-Type: application/sdp Content-Length: 276 Contact: <sip:btpsh-59c55b4a-4fa6-1@192.168.110.79> [sdp] *(3) 180 Ringing* SIP/2.0 180 Ringing From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 10 INVITE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0,SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 Contact: <sip:atpsh-59c55b4a-4fa6-1@192.168.110.79> Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes>,<sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> *(4) 200 OK* SIP/2.0 200 OK From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 10 INVITE Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 181 Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0,SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 Contact: <sip:atpsh-59c55b4a-4fa6-1@192.168.110.79> Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes>,<sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> [sdp] *(5) ACK* ACK sip:atpsh-59c55b4a-4fa6-1@192.168.110.79 SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.4446035c867218358dbd848f773b5d09.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKspF2zas~;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 10 ACK Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Content-Length: 0 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> *(6) ACK* ACK sip:100@192.168.1.73:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK1a43.b1cce71dc1b004e442faca53cd6d8994.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 10 ACK Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Content-Length: 0 Contact: <sip:atpsh-59c55b4a-4fa6-2@127.0.0.1> *(7) Re-Invite * INVITE sip:atpsh-59c55b4a-4fa6-1@192.168.110.79 SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK2a43.3723a1a3f343f1f1974a66f4a62f41a9.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKf6peiap7;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 11 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Content-Length: 276 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> [sdp] *(8) Re-Invite* INVITE sip:100@192.168.1.73:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK2a43.a4e5f5baf85414d4471a1e817bdc1e00.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 11 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Content-Length: 276 Contact: <sip:atpsh-59c55b4a-4fa6-3@127.0.0.1> [sdp] * **(9) 200 OK* SIP/2.0 200 OK From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 11 INVITE Contact: <sip:100@192.168.1.73:5080> Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 211 [sdp] ------------------------------------------------------------------------

2 5

UACREG - answering to 401 on invite
by Jean Cérien 26 Sep '17

26 Sep '17

Hello I'm using Kamailio 5.0.1 With the UACREG module, I am registering to a remote provider. Register goes out, 401 back, Register goes out with nonce & co, OK Later, when I send an invite, the provider issues an 401 Unauthorized. I guess it expects me to resubmit an INVITE with the authentication data, but I dont see how to do that. Any help would be appreciated ! J.

2 6

How to change SDP in 200 Ok reply
by Jurijs Ivolga 26 Sep '17

26 Sep '17

Hi, I have difficult times to edit SDP in 200 ok replies, I need to change RTP/SAVPF to UDP/TLS/RTP/SAVPF in all 200 Ok. Here is code snippet: onreply_route { if ( $rm == "INVITE" && status=="200") { if (search_body("a=fingerprint")) { if (search_body(" RTP/SAVPF ")) { xlog("L_INFO","200 OK DTLS call.\n"); replace_body_all(" RTP/SAVPF "," UDP/TLS/RTP/SAVPF "); } } } } What I'm doing wrong? I think my code is executed against SDP of Invite, but not 200 OK. Jurijs

2 2

Best variant to share variables between two dialogs?
by Kristijan Vrban 26 Sep '17

26 Sep '17

Hi, based on Daniels example from: http://blog.miconda.eu/2010/01/best-of-new-in-kamailio-300-19-htable.html ... modparam("htable", "htable", "acalls=>size=8;") ... route { ... if(is_method("INVITE") && !has_totag()) { # a new call if($shtcv(ht=>^$fU$)>=3) { send_reply("403", "limit exceeded"); exit; } $sht(acalls=>$ci) = $fU; t_on_failure("NEW_INVITE"); } if(is_method("BYE")) { $sht(acalls=>$ci) = $null; } ... } failure_route[NEW_INVITE] { $sht(acalls=>$ci) = $null; } I use this htable variant to share data between two dialogs (store Contact from one dialog, and re-write other R-URI with it). Referenced by call-id: ... $avp(referenceID) = $ci + "-" + $avp(direction); $sht(a=>$avp(referenceID)) = $ct; ... ... $avp(referenceID) = $ci + "-" + $avp(direction); avp(newru) = $sht(a=>$avp(referenceID)); avp_subst("$avp(newru)", "/(<)(.*)(>)/\2/"); xlog("L_NOTICE", ">>>>>>>> ReWrite ru with:$avp(newru)\n"); $ru = $avp(newru); ... Is this the best method to share data between two dialogs? Because you have to take care yourself with failure_route and so on, to free (set back to $null) them. Or is there a more elegant way for that task? Best Kristijan

1 0

Send 407 for Invite
by Logeshwaran G 26 Sep '17

26 Sep '17

Hi, How to send the 407 proxy authentication for Invite message? Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 3

Memory allocation issue
by Logeshwaran G 26 Sep '17

26 Sep '17

Hi, I am having the below Error while executing the kamailio cfg file. Before an hour it executed successfully, There is no changes in cfg. loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(27575) : <core> [core/mem/shm.c:105]: shm_getmem(): could not attach shared memory segment: Cannot allocate memory 0(27575) : <core> [core/mem/q_malloc.c:1203]: qm_malloc_init_shm_manager(): could not initialize qm shm memory pool Too much qm shm memory demanded: 67108864 bytes 0(27575) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/config/kamailio_final.cfg, line 302, column 39: Can't initialize shared memory ERROR: bad config file (1 errors) Input please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

1 0

Docker image
by Sergey Safarov 25 Sep '17

25 Sep '17

Hello guys Yes i packaged kamailio to docker container <https://hub.docker.com/r/safarov/kamailio3/>. This container is ready to run kamailio in docker environment. Could you make review of this. Dockerfiles you can review at https://github.com/kamailio/kamailio/tree/master/pkg/kamailio/alpine_docker Build process logs you can review at https://hub.docker.com/r/safarov/kamailio3/builds/bl9rv6ygeubwnydh4an2tuc/ Sergey

3 3

Astricon 2017
by Daniel-Constantin Mierla 25 Sep '17

25 Sep '17

Hello, Kamailio will be present at Astricon 2017 having a stand in the expo area (Orlando, FL, during October 3-5 -- more details at www.astricon.net). Should you participate at the event and have flyers/other materials with products/solutions/services that use Kamailio or giveaways, you are welcome to join the booth or just drop them there, we are more than happy to promote and spread them. Also, you are very welcome to just stop by and have a chat. Besides the activity in the exhibition area, I and Fred Posner will have presentations about using Kamailio together with Asterisk or other VoIP applications. Be sure you do not miss them, the agenda is available at: - http://www.asterisk.org/community/astricon-user-conference/schedule Expect a consistent group from Kamailio community, it's a great event to be at! Looking forward to meeting many of you next week in Orlando! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 3

Re: [SR-Users] [sr-dev] SIP ping, sockets and multi homed
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, After deep investigation I found out that the issue is happening because of the mechanism Kamailio uses to find the most appropriate socket combined with the virtual IP usage.... For socket selection, Kamailio first asks Linux which would be the routing for a certain destination, then matches this outcome with the list of listened sockets. By default Linux will always return the local IP instead of the virtual one. So if Kamailio only binds to the virtual IP, the Linux's returned IP won't match our socket list and then the socket selection fails.... So currently Kamailio active/stand-by HA setup with virtual IP in a multi-homed environment does not work properly... We can overcome this by listening to all addresses, but then Kamailio will decide to use the local IP instead of the virtual one, and then the HA idea end up broken.... We can also force socket for everything. For regular call routing, this options actually works fine if the script always force some socket, but it seems to be not always possible for local generated messages. By disabling mhomed and using the dispatcher socket attr to send the OPTIONS message it works for example. But for the LCR module we have no option because apparently the LCR module has only one socket option, and so I am not able have a different socket per gateway. We can also explicitly set the src address for a routing rule to be the virtual IP instead of the local one. This makes Linux to return the virtual IP instead of the local one. This solution seems fine, but since it is done at linux level, every other service on the box would be affected and every route on the box would needed to be changed.. Another problem I found out is that I am not able to force the socket for local generated messages. It actually crashes Kamailio, and I always get a "core was not generated" msg.... anyone aware of that? The actual fix for this, would be having Kamailio to search the most appropriate socket in a different manner. It could loop the sockets and see which one reaches the destination instead of asking Linux to do it, or it could have some routing directives which could be consulted before selecting a socket.... (just ideas, not sure how feasible they are...) I am considering using the Linux routing change option, but has anyone faced such situation before? Regards, Patrick Wakano On 24 August 2017 at 09:18, Patrick Wakano <pwakano(a)gmail.com> wrote: > Hello Daniel, > I still couldn't make this SIP probing work.... So currently my setup > cannot actively check for GW availability... > Any idea about what may be causing it? > > Thanks for your time, > Patrick Wakano > > On 16 August 2017 at 13:01, Patrick Wakano <pwakano(a)gmail.com> wrote: > >> Forgot to mention it is a centos7 box... >> >> On 16 Aug. 2017 11:02 am, "Patrick Wakano" <pwakano(a)gmail.com> wrote: >> >>> Hi Daniel! >>> Very grateful for your attention! >>> >>> Apologies for lacking important info, check below the version and >>> sockets. >>> It may be worth mentioning these IPs are virtual ones and I am using net.ipv4.ip_nonlocal_bind >>> = 1(in this case the virtual IP is assigned to the box) >>> Interesting that it has a "mhomed: no" for all sockets but I do have >>> mhomed=1 in my cfg file.... could that be the issue? >>> Also there is no message before this one: /usr/sbin/kamailio[6164]: >>> ERROR: <core> [core/forward.c:181]: get_out_socket(): no socket found >>> This is the first one that pops when dispatcher tries to ping.... >>> >>> Thanks, >>> Patrick >>> >>> kamailio -v >>> version: kamailio 5.0.2 (x86_64/linux) >>> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, >>> USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, >>> TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, >>> USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, >>> HAVE_RESOLV_RES >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> id: unknown >>> compiled on 12:03:49 Jun 26 2017 with gcc 4.8.5 >>> >>> kamcmd> core.sockets_list >>> { >>> socket: { >>> proto: udp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: udp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 8080 >>> mcast: no >>> mhomed: no >>> } >>> } >>> >>> >>> On 15 August 2017 at 17:24, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> what are the sockets kamailio is listening on? See `kamcmd help` for >>>> the command that lists the sockets. >>>> >>>> Also, what is the log messages before the first one you pasted, there >>>> should be another one printed from get_out_socket(). >>>> It is always important to provide kamailio version (kamailio -v) and >>>> the operating system you are using. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 14.08.17 14:41, Patrick Wakano wrote: >>>> >>>> Hello Kamailio list, >>>> >>>> Hope you all doing well! >>>> >>>> I am stuck with one problem that should be incredibly easy to be done, >>>> but apparently it is not.... I want my dispatcher, LCR and nathelper >>>> modules to always SIP ping my SIP clients. I have a Kamailio based SBC that >>>> is separating two networks, and so I am using the 'mhomed' flag, but none >>>> of these SIP pings are working.... for now I will only ask about the >>>> dispatcher.... >>>> My dispatcher module has to talk with SIP servers in both networks, the >>>> thing is when I enable SIP ping for it (modparam("dispatcher", >>>> "ds_probing_mode", 1)), it gives me this type of error: >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[6164]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> After research it seems I need to manually set a "socket" attribute for >>>> each of my GWs in dispatcher list. After putting the socket parameter for >>>> the GW, I got another error: >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/socket_info.c:2046]: >>>> parse_protohostport(): bad port number in udp:192.168.33.100:5060 >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[14190]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> What can be wrong in my port info in the socket config? (I tried the >>>> socket with and without quotes but had no luck...) >>>> select * from dispatcher where id=2; >>>> id | setid | destination | flags | priority >>>> | attrs | description >>>> ----+-------+-------------------------+-------+----------+-- >>>> -------------------------------------------+--------------------- >>>> 2 | 2 | sip:192.168.33.110:6060 | 0 | 0 | socket="udp: >>>> 192.168.33.100:5060";rweight=50 | >>>> (1 row) >>>> >>>> Anyhow, I was expecting the 'mhomed' parameter to do the socket >>>> selection for me. Regular call routing either by the dispatcher or by LCR, >>>> also seems to require me to explicitly set the socket otherwise Kamailio >>>> (or the OS) may end up using the wrong socket. So, another question is why >>>> 'mhomed' does not dictate the whole socket selection Kamailio has to do >>>> regardless the module it is using to send SIP requests? >>>> >>>> Sorry for the long e-mail.... >>>> Best regards, >>>> Patrick Wakano >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Development Mailing Listsr-dev@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Kamailio Advanced Training - www.asipto.com >>>> Kamailio World Conference - www.kamailioworld.com >>>> >>>> >>> >

2 2

LCR Ping GWs
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, I was trying to activate continuous SIP ping from the LCR module, which apparently was available in the past, but currently it is not possible anymore. As I could check It can only ping inactive GWs. Question is, why such change was promoted? We have to send a call to the GW to detect it has died instead of actively monitoring them and so avoid this useless call attempt. It is not that big deal, but it does gives me a better monitoring of my providers (specially the ones not used so frequently) health. Regards, Patrick

1 0

Enabling Redirection in Kamailio SIP Server
by Anas Rasheed 23 Sep '17

23 Sep '17

How can we enable redirection in Kamilio SIP Server using Siremis User Interface for it? Can anyone provide us with the tutorial of it?

1 0

Memory configuration
by Joe Baran 23 Sep '17

23 Sep '17

I feel fairly confident I’m running into a memory tuning issue, but I’m struggling to determine the next step. Can someone give me some guidance on how to adjust settings to allow the system to stop crashing kamailio? Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7fa9eeeea000, 144); Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fa9eeeea000, 144) called from kazoo: kz_amqp.c: kz_amqp_async_query(1357), module: kazoo; Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: kazoo [kz_amqp.c:1359]: kz_amqp_async_query(): failed to allocate kz_amqp_cmd in process 29468 Sep 14 18:25:15 server1 kamailio[29468]: INFO: <script>: kGddp3V0bA6nitnk25dB4w..|log|failed to send Kazoo query for authentication xxxxxxxxxxxxxxxxxx Sep 14 18:25:15 server1 kamailio[29468]: ERROR: tm [t_reply.c:494]: _reply_light(): cannot allocate shmem buffer Sep 14 18:25:15 server1 kamailio[29468]: ERROR: sl [sl.c:271]: send_reply(): failed to reply stateful (tm) Sep 14 18:25:16 server1 kamailio[29483]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 108 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:743]: handle_sigs(): child process 29438 exited by a signal 11 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Thanks, Joe Baran

3 6

Shared htable between servers
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi list, Is it possible to have a htable shared between servers on different machines? I have two kamailio proxies on different servers for resilience and I'm storing call information in a hash table. Then I can route calls for the same user to the respective PBX. The reason for this is to have attended transfers working reliably (attended transfer between PBXes are a mess). Thank you! Kind regards, -- /Iskren Hadzhinedev/

2 1

Call transfer between pbx servers behind kamailio proxy
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi everyone, I'm having a hard time transferring calls when the users are on different FreeSWITCH servers behind kamailio. The users are distributed as follows: user1(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user2(a)sip.test.com-> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user3(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH2 When user1 calls user2 and user2 does an attended transfer to user3, the following happens: 285 67.220453 USER2_IP -> KAMAILIO_IP SIP 726 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 286 67.220674 KAMAILIO_IP -> FreeSWITCH1_IP SIP 877 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 287 67.223651 FreeSWITCH1_IP -> KAMAILIO_IP SIP 925 Status: 202 Accepted | 288 67.223753 KAMAILIO_IP -> USER2_IP SIP 838 Status: 202 Accepted | 289 67.225502 FreeSWITCH1_IP -> KAMAILIO_IP SIP/SDP 1225 Request: INVITE sip:user3@sip.test.com | 290 67.387323 KAMAILIO_IP -> 69.172.200.109 SIP/SDP 1420 Request: INVITE sip:user3@sip.test.com | Turns out 69.172.200.109 is the actual IP address of sip.test.com (I've probably should have gone with sip.example.com or something else). What is the proper way to route that INVITE back to the other FreeSWITCH server so that the transfer would work? Thanks! Kind regards, -- /Iskren Hadzhinedev/

2 2

how to play ring tune when callee decline s
by 赵国杰 22 Sep '17

22 Sep '17

Hi guys, I want the caller to play a short audio(like "the number your are calling is busy") when the callee declines the call. How can i do that? Best Regards Jesse

4 21

Typos in Kamctl Stats results
by Noah Mehl 21 Sep '17

21 Sep '17

Hey, I’m noticing that the results from the websockets module when running # kamctl stats there are some odd results that may just be typos? "websocket:ws_msrp_max_concurrent_connections = 0", "websocket:ws_msrp_max_concurrent_connectons = 0”, "websocket:ws_sip_max_concurrent_connections = 0", "websocket:ws_sip_max_concurrent_connectons = 0", Thanks! ~Noah

3 2

Encryption of voice stream
by Vasan VS 21 Sep '17

21 Sep '17

Hello all, Kindly let us know if we can have a mechanism to have the voice stream in using the SIP server completely encrypted as well as prove that it is encrypted using any speific mechanism. Thanks in advance Vasan

3 2

COALESCE Transformation?
by Brandon Armstead 20 Sep '17

20 Sep '17

Hello all, I've checked the docs and did some brief searching, curious if this transformation or method exists currently? Such as: https://dev.mysql.com/doc/refman/5.7/en/comparison-operators.html#function_… Thanks! -- Sincerely, Brandon Armstead CTO / CRYY.com

1 0

private memory leak - but where?
by Sebastian Damm 20 Sep '17

20 Sep '17

Hi, I'm trying to debug a Kamailio which runs out of memory after a while. But following the hints at https://www.kamailio.org/wiki/tutorials/troubleshooting/memory doesn't help. I know it is the private memory that I'm missing. I already increased the pkg size to 16M, but somehow that seems to be not enough. The Kamailio receives HEP messages, sends them to a lua script where it is processed. Overall there are not more than 10 custom private variables used. In the log I'm seeing these messages: Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: app_lua [app_lua_api.c:726]: app_lua_run_ex(): error executing: handle_packet (err: 2) Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <script>: SCRIPT: failed to execute lua function! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7f9744eea010, 232); Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f9744eea010, 232) called from core: core/parser/msg_parser.c: get_ hdr_field(116), module: core; Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:118]: get_hdr_field(): out of memory Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:330]: parse_headers(): bad header field [Via: SIP/2.0/WSS o6m] Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: pv [pv_core.c:647]: pv_get_callid(): cannot parse Call-Id header kamcmd pkg.stats for this process gives me this output. { entry: 61 pid: 19134 rank: 61 used: 8748080 free: 4384 real_used: 16772832 total_size: 16777216 total_frags: 30 } Now I tried to get a dump of what is in use, following the guidelines on the wiki page mentioned above, but without success. I can easily start and stop the traffic coming in to Kamailio (by starting or stopping the captagent sending it there), but I can't get any output. Here's what I did: captagent stop kamcmd cfg.set_now_int core memlog 1 kamcmd cfg.set_now_int core mem_dump_pkg 19134 captagent start ; sleep 1 ; captagent stop (I started captagent so that the process actually had something to do because that's what the wiki page says it needs for printing memory information.) During that second, there were quite a lot of packets running through this process, but I did not get a memory dump. A shm_summary dump however works as expected. What am I missing? How do I get more information about what this process is using its memory for? TIA Sebastian

2 6

sipp audio file
by Diego Nadares 19 Sep '17

19 Sep '17

Hi guys, Any body know how to create pcap (with audio) file to use with sipp?  <nop> <action> <exec play_pcap_audio="../scenarios/pcap/demo.cap"/> </action> </nop> I tried with wireshark capturing rtp but I'm doing something wrong i think. Thanks! Diego

4 4

jsonrpcc problem
by Adam Kuśmirek 19 Sep '17

19 Sep '17

Hello, I have a problem with jsonrpcc module. OS is Debian Jessie. I tried with packeged kamailio 4.4 and compiled kamailio 5.0. Kamailio starts properly. Routing block is very simple: route { jsonrpc_request("get_source", "{'ipv4': '$si', 'port': $sp}", "RESPONSE", "ERROR", "$var(result)"); exit; } route[RESPONSE] { xlog("Result received: $var(result)"); } route[ERROR] { xlog("Error received: $var(result)"); } If any message arrives, kamailio sends it to my rpc_server (netstring encoded): 106:{'id': 2, 'jsonrpc': '2.0', 'method': 'get_source', 'params': {'ipv4': '10.20.121.231', 'port': 5060}}, My server decodes the message and immediately answers with netstring encoded response: 47:{"jsonrpc": "2.0", "result": "200 OK", "id": 2}, At this moment kamailio logs errors and disconnects from rpc server: root@kamailio-test:/etc/kamailio# 4(925) DEBUG: <core> [parser/msg_parser.c:595]: parse_msg(): SIP Request: 4(925) DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:kamailio.conpeek.com> 4(925) DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> 4(925) DEBUG: <core> [parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1033201528>; state=16 4(925) DEBUG: <core> [parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 4(925) DEBUG: <core> [parser/msg_parser.c:486]: parse_headers(): Via found, flags=2 4(925) DEBUG: <core> [parser/msg_parser.c:488]: parse_headers(): this is the first via 4(925) DEBUG: <core> [receive.c:178]: receive_msg(): After parse_msg... 4(925) DEBUG: <core> [receive.c:221]: receive_msg(): preparing to run routing scripts... 4(925) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 4(925) DEBUG: <core> [parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 4(925) DEBUG: <core> [parser/msg_parser.c:171]: get_hdr_field(): <To> [40]; uri=[sip:1000@kamailio.conpeek.com] 4(925) DEBUG: <core> [parser/msg_parser.c:172]: get_hdr_field(): to body ["1000" <sip:1000@kamailio.conpeek.com> ] 4(925) DEBUG: <core> [parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:183]: get_hdr_field(): content_length=0 4(925) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 4(925) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=43020, isACK=0 4(925) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(925) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 4(925) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: 920bfe28c581e1bf18554e3a9b234ceb 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 4(925) DEBUG: <core> [receive.c:318]: receive_msg(): cleaning up *0(921) ALERT: <core> [main.c:740]: handle_sigs(): child process 929 exited by a signal 11* * 0(921) ALERT: <core> [main.c:743]: handle_sigs(): core was generated* 0(921) INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD 0(921) DEBUG: <core> [main.c:757]: handle_sigs(): terminating due to SIGCHLD 7(928) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 6(927) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 5(926) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 4(925) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 3(924) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 2(923) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 1(922) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 0(921) DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start 0(921) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table 0(921) DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fb99a822350 from h_table.c:449 0(921) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores 0(921) DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists 0(921) DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done 0(921) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 0(921) DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock 0(921) DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc 0(921) DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc The problem occures also if rpc server does not reply, i think when trying to start ERROR route. If I try with jsonrpc_notification, everything is OK. Any help ? Regards Adam

2 2

See how many workers are actually processing packets?
by Sebastian Damm 19 Sep '17

19 Sep '17

Hi, I think this has been answered before, but I don't know the answer. Is there a way to find out, how many workers are busy right now processing packets? For a new setup I don't know whether I need 16 or 96 UDP listeners, but I'd like to scale it correctly. The setup just inspects HEP messages and sends some of them to a RabbitMQ. So TM or SL stats probably won't help. Best Regards Sebastian

1 0

Proxy Invite to registered natted user
by eskornev 19 Sep '17

19 Sep '17

Hi All Trying to set Kamailio as proxy in front of Asterisk. When natted user registers is remembered in Kamailio via save("location"). Kamailio forwards Register to * with new contact field (pointing to Kamailio). But when server originates Invite to user, kamailio transfer request to "fake" address that was in original Register contact field (getting it from lookup("location")). How can i set Kamailio to send Invite to source ip and port of user?

2 1

kamailio jsonrpc client
by Aidar Kamalov 19 Sep '17

19 Sep '17

Hello, cannot get working rpc client n kamailio. [root@sipchel kamailio]# grep 'json\|jansson' kamailio-default.cfg loadmodule "jsonrpcs.so" loadmodule "jansson.so" loadmodule "janssonrpcc.so" # jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 3) # jansson modparam("janssonrpcc", "server", "conn=user_db;addr=127.0.0.1;port=5060;priority=10;weight=10"); .... request_route { ... janssonrpc_request("user_db", "core.version", '', "route=RESPONSE;retry=1"); ... } .... event_route[xhttp:request] { if ($hu =~ "^/jsonrpc") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } With this config i got error Sep 19 08:52:15 sipchel /usr/sbin/kamailio[39891]: INFO: janssonrpcc [janssonrpc_connect.c:294]: bev_connect(): Connecting to server 127.0.0.1:5060 for conn user_db. Sep 19 08:52:15 sipchel /usr/sbin/kamailio[39891]: INFO: janssonrpcc [janssonrpc_connect.c:277]: bev_connect_cb(): Connected to host 127.0.0.1:5060 ... Sep 19 08:52:20 sipchel /usr/sbin/kamailio[39891]: WARNING: janssonrpcc [janssonrpc_request.c:242]: schedule_retry(): Number of retries exceeded. Failing request. Sep 19 08:52:20 sipchel /usr/sbin/kamailio[39891]: WARNING: janssonrpcc [janssonrpc_io.c:218]: fail_request(): Request timeout: #012{"id":3,"jsonrpc":"2.0","method":"core.version"} But [root@sipchel kamailio]# curl -s -H 'Content-Type: application/json' --data-binary '{"jsonrpc":"2.0", "method": "core.version"}' http://localhost:5060/jsonrpc { "jsonrpc": "2.0", "result": "kamailio 5.0.3 (x86_64/linux) " } How to get working rpc client in kamailio? Is it possible to define url for janssonrpcc module? -- Aydar A. Kamalov

1 0

rtpproxy sdp
by Isravel Raja Thangamani 18 Sep '17

18 Sep '17

Hi Thanks in advance if anyone can point me in the correct direction . I have kamailio running in front of some asterisk. SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I'm running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider no audio from external caller to the asterisk ext neither asterisk to external caller I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs I think my mistake in somewhere in the cfg below. Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints? This e-mail message and any documents attached to it are confidential and may contain information that is protected from disclosure by various federal and state laws, including the HIPAA privacy rule (45 C.F.R., Part 164). This information is intended to be used solely by the entity or individual to whom this message is addressed. If you are not the intended recipient, be advised that any use, dissemination, forwarding, printing, or copying of this message without the sender's written permission is strictly prohibited and may be unlawful. Accordingly, if you have received this message in error, please notify the sender immediately by return e-mail or call (insert phone number), and then delete this message.

3 2

ringing calls dialogs
by Aidar Kamalov 18 Sep '17

18 Sep '17

Hello, is it possible from route get info about all calls in ringing state? (for call pickup implementation) Is it better to do via dialogs or presence modules? -- Aydar A. Kamalov

3 4

Issue in ACC_CDR
by Logeshwaran G 18 Sep '17

18 Sep '17

Hi, I have enabled both cnxcc and ACC cdr flags. The call is dropped when the credit is got over (its handling by cnxcc correctly.) In this case the details are not storing in the ACC_cdr table. I am not sure what I am missing. in cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails Error: 2(19484) ERROR: acc [acc_cdr.c:373]: write_cdr(): message is empty! 2(19484) ERROR: acc [acc_cdr.c:656]: cdr_on_end_confirmed(): failed to write cdr! Please Guid me. Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 1

Siremis - version from git don't work
by Yu Boot 18 Sep '17

18 Sep '17

Downloaded new Siremis version via "git clone" and replaced /var/www/ content by new files. Now any menu entry gives the following error: https://pastebin.com/hV8ngRVx If if copy old Siremis files back, everything works fine.

2 8

Polycom Presence Routing for PUBLISH Messages
by John Janke 18 Sep '17

18 Sep '17

So I'm using a Polycom VVX 310 with Presence enabled (both on Phone and Kamailio) and the phone is sending SUBSCRIBE messages but not PUBLISH. I have tried various phone configurations without success. The closest thread to my situation I found online said this: "The thing is Polycom phones perform end-to-end subscription to presence event. It worked when I changed the routing configuration to simply relay the SUBSCRIBE request to the monitored device. (I'm using OpenSER as SIP proxy.)" I'm not sure if this is correct as it's from 2012, but in the case that it is I don't quite understand how to implement this. Any help on how to edit the config file? John Janke Progressive Innovation and Automation, Inc. PO BOX 309 Plainfield, IL 60544 Direct: 847.316.9028 Office: 800.688.9051 x 55328 Fax: 630.860.1017

2 1

Re: [SR-Users] Siremis graph time format is a bug!?
by Humberto Rodríguez 18 Sep '17

18 Sep '17

Thanks Daniel-Constantin Mierla, I followed your recommendation, and began to use a date column as INT. Then, I started to insert the date as unix_timestamp, like this (unix_timestamp(now())), which was the only way the graphic showed the date correctly (Active_Failed_Dialogs_4_hours - From 2017-09-15 12:04:01 To 2017-09-15 16:00:01), I failed inserting (now()+0), siremis didn't understand this format too. Cheers, Hto Now No Matter How Hard The Past, You Can Always Begin Again ~ -------------------------------------------- On Thu, 9/14/17, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Subject: Re: [SR-Users] Siremis graph time format is a bug!? To: "Humberto Rodríguez" <humberj77(a)yahoo.es>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, sr-users(a)lists.sip-router.org Date: Thursday, September 14, 2017, 4:09 AM Hello, try to set the type of the time column to int, so it is retrieved as numeric timestam, not as formated datetime. Cheers, Daniel On 13.09.17 05:39, Humberto Rodríguez wrote: Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had siremis 4.2 and never saw it correctly, I just updated to Siremis 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file <?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- <ChartGroup name="sumdialogs"> <Chart name="sumdialogs" table="active_dialogs" title="Active_Failed_Dialogs" type="area" order="reverse" orderby="ORDER BY id DESC" limit="LIMIT 10080"> <XAxis>  <Item name="date_created" data="date_created" title="date_created" type="timestamp"/> </XAxis> <YAxis> <Item name="active_dialogs" data="active_dialogs" title="active_dialogs" type="dataset" color="#D02020"/> <Item name="failed_dialogs" data="failed_dialogs" title="failed_dialogs" type="dataset" color="#DFC329"/> </YAxis> </Chart> </ChartGroup> --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" ------------------------------------------------------------------------------------------ # id, date_created, active_dialogs, early_dialogs, expired_dialogs, failed_dialogs, processed_dialogs, '1', '2017-04-11 14:40:40', '220', '0', '1219', '37', '6879491' '2', '2017-04-11 14:42:02', '206', '0', '1219', '37', '6879604' '3', '2017-04-11 14:43:01', '215', '0', '1219', '37', '6879685' '4', '2017-04-11 14:44:01', '191', '0', '1220', '37', '6879744' '5', '2017-04-11 14:45:02', '203', '0', '1220', '37', '6879837' '9', '2017-04-11 14:46:01', '211', '0', '1220', '37', '6879929' '11', '2017-04-11 14:47:01', '234', '0', '1220', '37', '6880019' '12', '2017-04-11 14:48:01', '222', '0', '1220', '37', '6880087' ---------------------------------------------------------------------------------------------------------------------------------- No Matter How Hard The Past, You Can Always Begin Again ~ _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 1

Kamailio Offline Messages
by Jango 17 Sep '17

17 Sep '17

I need help about offline messages this is my full working kamailio.cfg with audio/video call & instant messaging but i need features to receive user offline messages. kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_NAT #!define WITH_TLS #!define WITH_ACCDB # # Kamailio (OpenSER) SIP Server v4.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="jsrver.tk" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # -- for siremis CDRs -------------- loadmodule "rtimer.so" loadmodule "sqlops.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "//etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@localhost/kamailio") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # ====================================================== # Populate CDRs Table of Siremis # ====================================================== route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 1

Dispatcher Module with ds_is_from_list function
by Ilie Soltanici 17 Sep '17

17 Sep '17

Hello, "ds_is_from_list" function from Dispatcher module give false result - when actually it should be true. modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") request_route { xlog("L_INFO","[=Start Routing=]") if(ds_is_from_list()) { xlog("L_INFO","[=FROM ASTERISK=]"); } else { xlog("L_INFO","[=TO ASTERISK=]"); } if($si=="192.168.1.10") { xlog("L_INFO","[=FROM ASTERISK BY IP=]"); } else { xlog("L_INFO","[=TO ASTERISK BY IP=]"); } root@kamailio:/var/log# kamctl dispatcher show dispatcher gateways +----+-------+---------------------+-------+----------+-------+-----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+---------------------+-------+----------+-------+-----------------+ | 1 | 1 | sip:192.168.1.10:5060 | 0 | 3 | | Main Asterisk | | 2 | 2 | sip:192.168.1.11:5060 | 0 | 2 | | Backup Asterisk | | 3 | 3 | sip:192.168.1.12:5060 | 0 | 1 | | Test Asterisk | +----+-------+---------------------+-------+----------+-------+-----------------+ root@kamailio:/var/log# cat /var/log/kamailio.log [=Start Routing=] [=TO ASTERISK=] [=FROM ASTERISK BY IP=] Additional Information - SIP Packet 2017/09/15 19:21:57.589451 192.168.1.50:5061 -> 192.168.1.1:5060 REGISTER sip:192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5061;branch=z9hG4bK-5219-1-0 From: sip:10@192.168.1.50;tag=1 To: sip:10@192.168.1.50 Call-ID: 1-5219(a)192.168.1.50 CSeq: 1 REGISTER Contact: sip:10@192.168.1.50:5061 Max-Forwards: 10 Expires: 120 User-Agent: SIPp/Win32 Content-Length: 0 - Kamailio Version - output of kamailio -v root@kamailio:/var/log# kamailio -v version: kamailio 5.0.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 12:12:14 Sep 7 2017 with gcc 6.3.0 root@kamailio:/var/log# - Operating System: Debian 9.1/ Linux kamailio1 4.9.0-3-amd64 #1 SMP Debian 4.9.30-2+deb9u3 (2017-08-06) x86_64 GNU/Linux Is this a bug or something wrong with configuration? Thank you

2 1

Implementing C or Java file in kamailio
by Logeshwaran G 15 Sep '17

15 Sep '17

Hi, Is there any way to implement my own C or Java program in kamailio.cfg to Validate my own scenarios? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 2

Replacing the From header user part.
by vivek＠advaitamtech.com 15 Sep '17

15 Sep '17

Hi All, Am trying to replace the user part of From header with own user name in kamailio.cfg I tried to assign directly, Eg: $fU = "abcd" Sometime when I receive From header as ' "12345" <12345(a)1.1.1.1>;tag=xxxx '. In this case it works as ' "12345" <abcd(a)1.1.1.1>;tag=xxxx ' . So i tried using replace_all() function. But this ended up adding extra characters(in my case "abcd") infront of content_type header. I dont know why. Can someone please help me out in replacing the from header in this case? Thanks, Vivek

2 2

Maximum Registration on Kamailo??
by amit Kumar 14 Sep '17

14 Sep '17

Hi All, kamctl ul show => show below result, does that mean max 1024 contact can be registered, if that is the case how to increase the max registration limit. Domain:: location table=*1024 *records=2 max_slot=1 Thanks

3 2

kamailio - multiple asterisk and chanspy
by Жан Базаров 14 Sep '17

14 Sep '17

Hello! how i can use asterisk's chanspy_app in kamailio-loadbalance-asterisk system?

2 1

Killing Calls for a Profile
by Sunil More 14 Sep '17

14 Sep '17

Hello All, Is there a method / command to kill calls related to one dialog profile. Right now i have to find all dialogs for a customer and loop through all the dialog hashes to run kamcmd dlg.end_dlg till i can kill all calls. Is there a easier way to do this. Thanking You, Sunil More Ph : 9503338275

2 4

Siremis "Chat Service" time format has bug or is only config issue??
by Humberto Rodríguez 14 Sep '17

14 Sep '17

Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had version 4.2 and never saw it correctly, I just updated to version 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file ?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- > <ChartGroup name="sumdialogs"> > > <Chart name="sumdialogs" table="active_dialogs" > title="Active_Failed_Dialogs" type="area" order="reverse" > > > orderby="ORDER BY id DESC" limit="LIMIT 10080"> > > <XAxis> >  > > <Item > name="date_created" data="date_created" title="date_created" > type="timestamp"/> > > </XAxis> > > <YAxis> > > <Item > name="active_dialogs" data="active_dialogs" > title="active_dialogs" type="dataset" color="#D02020"/> > > > <Item > name="failed_dialogs" data="failed_dialogs" > title="failed_dialogs" type="dataset" color="#DFC329"/> > > </YAxis> > </Chart> > </ChartGroup> > --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" > ------------------------------------------------------------------------------------------ > > > # id, date_created, active_dialogs, > early_dialogs, expired_dialogs, failed_dialogs, > processed_dialogs, > '1', '2017-04-11 14:40:40', '220', '0', > '1219', '37', '6879491' > '2', '2017-04-11 14:42:02', '206', '0', > '1219', '37', '6879604' > '3', '2017-04-11 14:43:01', '215', '0', > '1219', '37', '6879685' > '4', '2017-04-11 14:44:01', '191', '0', > '1220', '37', '6879744' > '5', '2017-04-11 14:45:02', '203', '0', > '1220', '37', '6879837' > '9', '2017-04-11 14:46:01', '211', '0', > '1220', '37', '6879929' > '11', '2017-04-11 14:47:01', '234', > '0', '1220', '37', '6880019' > '12', '2017-04-11 14:48:01', '222', > '0', '1220', '37', '6880087' > > ----------------------------------------------------------------------------------------------------------------------------------

2 1

Siremis graph time format is a bug!?
by Humberto Rodríguez 14 Sep '17

14 Sep '17

Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had siremis 4.2 and never saw it correctly, I just updated to Siremis 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file <?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- <ChartGroup name="sumdialogs"> <Chart name="sumdialogs" table="active_dialogs" title="Active_Failed_Dialogs" type="area" order="reverse" orderby="ORDER BY id DESC" limit="LIMIT 10080"> <XAxis>  <Item name="date_created" data="date_created" title="date_created" type="timestamp"/> </XAxis> <YAxis> <Item name="active_dialogs" data="active_dialogs" title="active_dialogs" type="dataset" color="#D02020"/> <Item name="failed_dialogs" data="failed_dialogs" title="failed_dialogs" type="dataset" color="#DFC329"/> </YAxis> </Chart> </ChartGroup> --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" ------------------------------------------------------------------------------------------ # id, date_created, active_dialogs, early_dialogs, expired_dialogs, failed_dialogs, processed_dialogs, '1', '2017-04-11 14:40:40', '220', '0', '1219', '37', '6879491' '2', '2017-04-11 14:42:02', '206', '0', '1219', '37', '6879604' '3', '2017-04-11 14:43:01', '215', '0', '1219', '37', '6879685' '4', '2017-04-11 14:44:01', '191', '0', '1220', '37', '6879744' '5', '2017-04-11 14:45:02', '203', '0', '1220', '37', '6879837' '9', '2017-04-11 14:46:01', '211', '0', '1220', '37', '6879929' '11', '2017-04-11 14:47:01', '234', '0', '1220', '37', '6880019' '12', '2017-04-11 14:48:01', '222', '0', '1220', '37', '6880087' ---------------------------------------------------------------------------------------------------------------------------------- No Matter How Hard The Past, You Can Always Begin Again ~

2 1

Cannot msg or call
by Костик 14 Sep '17

14 Sep '17

Hello, I have setup Kamailio 5.0.3 under Ubuntu 16.04. I can connect to the server but cannot call/text my contacts and don't see them online. When I try to call them, I get errors like ERROR: <core> [core/tcp_main.c:4264]: tcpconn_main_timeout(): connect 10.8.8.218:51079 failed (timeout) For texts, the error is 405 Method not allowed What is the problem and how to fix it? Thanks Serj

2 2

kamailio 4.4 registering to a SIP trunk provider using UAC then outbound routing to it.
by Jonathan Hunter 13 Sep '17

13 Sep '17

Hi Guys, I have a SIP provider who only accept registrations from users, so I have registered my kamailio Server with them using the uac module. It is registered fine, and I can get inbound calls on the trunk. My query is the best way to send outbound calls out via the registered sip gateway? I would normally load balance using dispatcher if it was to IP authenticating providers, but as they are using registration, what would be best practice to send outbound calls via the registration which is establised with the uac module? Is it just a question of doing it within the configuration file, as again they will look to auth requests as well with a 401/407 so I assume I can use uac_auth() for that. Just wondering what other people have done if anything, is my other option is to register using a B2Bua on the same server, and leave kamailio just as a proxy. Many thanks Jon

2 2

dialog keep-alive don't work
by Yu Boot 13 Sep '17

13 Sep '17

Hello. I'm using following dialog module settings: modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "profiles_with_value", "limithash ; caller") modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 300) modparam("dialog", "send_bye", 1) and I expect some OPTIONS within dialogs. But that don't happen. I don't see any OPTIONS messages in sngrep in established calls. What's wrong? I call dlg_manage() right before main t_relay, everything including CDRs are working fine...

2 3

Issue in Acc module
by Logeshwaran G 13 Sep '17

13 Sep '17

Hi, I am getting the below Error while Call Ended. 2(28504) ERROR: acc [acc_cdr.c:373]: write_cdr(): message is empty! 2(28504) ERROR: acc [acc_cdr.c:656]: cdr_on_end_confirmed(): failed to write cdr! Am I missing something? Please Guide me Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

1 1

kill the call using callid
by Logeshwaran G 13 Sep '17

13 Sep '17

Hi, Is it possible to kill the call using callid, I have referred the dialog module, I found the command: kamcmd dlg.end_dlg 342 56 But I need to kill the call with Callid, Input Please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 3

Can not register on kamailio
by Adesh Pandey 13 Sep '17

13 Sep '17

I am new to kamailio and I want to setup Kamailio as dispatcher for two Asterisk boxes but SIP registrations must be happening on the Kamailio itself. I have enabled mysql module and used auth module but when I try to register it says unauthorized access. In subscribers list I have added an entry for sip user 1 with domain as kamailio IP. Please help me to understand where can I improve so that I can implement register on Kamailio. -- Adesh Pandey Sr. Software Developer [image: phone] +91 92129 92129 Ext: 56 [image: moible] +91 8527384897 [image: Profile Pic] [image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter] <https://twitter.com/MyOperator>[image: Twitter] <https://www.youtube.com/user/MyOperatorCo>[image: Linkedin] <https://www.linkedin.com/company/voicetree-technologies>[image: Google+] <https://plus.google.com/+MyoperatorCo/posts> [image: Logo] <http://myoperator.co/>

2 4

Problem implementing usrloc via dmq
by Aleksandar Sosic 13 Sep '17

13 Sep '17

Hi everyone, we're trying to make work usrloc over dmq without a db on our nodes. Our `kamailio.cfg` code to make it work on the nodes is like this: ``` ... loadmodule "usrloc.so" ... loadmodule "dmq.so" loadmodule "dmq_usrloc.so" ... modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:routers.ourcloud.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) ``` Where `DMQ_ADDRESS="sip:<some_address>:5060"` is the public IP address of the running node. `nslookup` of `routers.ourcloud.com`: ```➜ ~ nslookup routers.ourcloud.com Server: 8.8.8.8 Address: 8.8.8.8#53 Non-authoritative answer: Name: rrouters.ourcloud.com Address: <IP1> Name: routers.ourcloud.com Address: <IP2> ``` and obviously IP1 and IP2 are pubblic IPs of kamailio nodes that respond to my pings. When i register a client to node IP1 it's ul dump is: ``` { "jsonrpc": "2.0", "result": { "Domain": "location", "Size": 1024, "AoRs": [{ "Info": { "AoR": "0409828030", "HashID": 1970914268, "Contacts": [{ "Contact": { "Address": "sip:0409828030@172.19.0.255:56328;ob", "Expires": 289, "Q": -1, "Call-ID": "5mclwiX4oYAVK0z-iT7M0p9RBX5-02g6", "CSeq": 59427, "User-Agent": "Telephone 1.2.6", "Received": "sip:<PROXY_IP>:5060", "Path": "[not set]", "State": "CS_NEW", "Flags": 0, "CFlags": 192, "Socket": "udp:<IP1>:5060", "Methods": 8159, "Ruid": "uloc-599d8fa8-f-1", "Instance": "[not set]", "Reg-Id": 0, "Last-Keepalive": 1503498176, "Last-Modified": 1503498176 } }] } }], "Stats": { "Records": 1, "Max-Slots": 1 } }, "id": 1 } ``` On node with IP2 the ul dump is: ``` { "jsonrpc": "2.0", "result": { "Domain": "location", "Size": 1024, "AoRs": [], "Stats": { "Records": 0, "Max-Slots": 0 } }, "id": 1 } ``` What are we doing wrong? Are we missing something? -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

3 7

Help to understand kamailio error msgs
by José Seabra 12 Sep '17

12 Sep '17

Hello there, I'm facing some error msg on my kamailio server in production that i would like to understand what they means in order to avoid them. The msg are: - *ERROR: tm [t_reply.c:1270]: t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200* - *dispatcher [dispatch.c:1509]: ds_load_replace(): cannot find load for (16720.mydomain.com <http://16720.mydomain.com>)* - *dispatcher [dispatch.c:2270]: ds_next_dst(): cannot update load distribution* - *tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 487 reply when a final 603 was sent out* - *tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 487 reply when a final 480 was sent out* - *WARNING: tm [t_lookup.c:245]: ack_matching(): WARNING: ack_matching() attempted on a transaction with no E2EACK callbacks => the results are not completely reliable when forking is involved* Thank you Best regards -- José Seabra

2 3

jsonrpc execution error
by David Villasmil 12 Sep '17

12 Sep '17

Hello guys, i configured jsonrpc on kamailio.cfg and RPCFIFOPATH on kamctlrc to the same file, seems to work for some commands, but not for others: root@ip-172-31-30-160:/home/admin# kamctl trusted show +----+--------------+-------+--------------+--------------+- -----+----------+ | id | src_ip | proto | from_pattern | ruri_pattern | tag | priority | +----+--------------+-------+--------------+--------------+- -----+----------+ | 1 | 1.2.3.4 | udp | | NULL | | 0 | | 2 | 5.4.3.2 | udp | | NULL | | 0 | +----+--------------+-------+--------------+--------------+- -----+----------+ root@ip-172-31-30-160:/home/admin# kamctl trusted dump { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32022 } root@ip-172-31-30-160:/home/admin# kamctl trusted reload { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32041 } Any help? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Multiple Htable Reload
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, I have multiple Htables, Is it possible to Reload all the Htables at one time via jsonrpc: As of Now I am using the below command to Reload one htable at a time. {"jsonrpc": "2.0","method": "htable.reload","params": ["b"],"id": 123} Input Please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 1

Map Technique(Key value pair) in kamailio
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, Is there any way to use the Mapping technique in kamailio. For Example: ========== Key = 91, Value = India Key = 92, Value = Srilanka How to store Key Value Pairs into kamailio and how to Retrieve? If I pass 91 it should Retrieve India, Is it possible? Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 8

Passive Open TCP sessions increases and kamailio runs out of memory
by Niklas Larsson 11 Sep '17

11 Sep '17

Hi, I have a kamailio 4.4.5 (x86_64/linux), running as a proxy in front of asterisk, handling udp/tcp/tls registration, using udp towards asterisk. Over time memory is consumed and the Passive Open TCP sessions is increasing until we run out of memory and a restart is needed. Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f9d55184000, 34120); Free fragment not found! Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f9d55184000, 34120) called from tls: tls_init.c: ser_malloc(275), module: tls; Free fragment not found! This image contains graphs that show the behaviour: https://imgur.com/68PvmyA Should it behave like this? Is there a knob to twist to change behaviour? tcp / tls stuff from kamailio.cfg: enable_tls=yes tcp_max_connections=8196 tls_max_connections=8196 tcp_no_connect=yes # Don't try to connect - drop if there is no session tcp_connection_lifetime=3605 modparam("tcpops", "closed_event", 1) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) in the registrar route I have: tcp_keepalive_enable("60", "5", "5"); /niklas

1 0

kamailio asterisk queue
by Жан Базаров 11 Sep '17

11 Sep '17

hello! how i can forward presence peer state to asterisk queue? Thank's

2 1

Q: purpose of t_newtran() in kamailio-basic.cfg route[LOCATION]?
by Anthony Alba 09 Sep '17

09 Sep '17

In kamailio-basic.cfg route[LOCATION] we have the following: --snip-- $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: --snip-- What is the purpose of t_newtran() here? Since we have not hit t_relay() yet (and don't have a transaction) is this to ensure the caller sees/acks the 404/405? Earlier in the mailing list https://lists.kamailio.org/pipermail/sr-users/2015-April/088062.html I read, although this is in reply to a different situation: "Why do you need to create new transactions yourself with t_newtran()? Nonexotic applications of Kamailio usually don't require this. ..." # User location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } Thanks AAlba

2 1

Re: [SR-Users] append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

After I add route(DISPATCH) to branch_route branch calls are going throught asterisk # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); xnotice("BRANCH: $rm $ru [$ci]"); route(DISPATCH); route(NATMANAGE); } but if I set $ru = "sip:34200@domain.org"; append_branch("sip:34201@domain.org"); first main ($ru) INVITE resolve to contact of ip phone INVITE sip:34200@172.16.3.7:5060 but branch doesnt' resolve INVITE sip:34201@domain.org Is there any way to resolve it? -- Aydar A. Kamalov

2 1

append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

Hello! I have a simple route route[HUNT] { $ru = "sip:34200@172.16.3.7:5060"; append_branch("sip:34201@192.168.2.71"); } 172.16.3.7 and 192.168.2.71 ip phones 192.168.10.54 is kamailio 192.168.10.47 is asterisk If I called to this route kamailio will call to 34200 through dispatcher via asterisk. But branch going directly to the ip phone. Is it possible to call all branches through dispatcher and asterisks? U 192.168.10.54:5060 -> 192.168.10.47:5060 INVITE sip:34200@172.16.3.7:5060 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.0. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. U 192.168.10.54:5060 -> 192.168.2.71:5060 INVITE sip:34201@192.168.2.71 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.1. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. -- Aydar A. Kamalov

2 1

need some help about acc module
by Vidja Prashant 08 Sep '17

08 Sep '17

Hello, I have some feature required changes in acc module. In acc module we getting cdrs in database only which have METHOD=INVITE METHOD=BYE but in my setup I also get METHOD=OPTIONS so I need this entry also in acc table in kamailio database. so which changes i need to do in kamailio acc module.... or other...... my cdrs log I paste below which have mehod=options "Sep 7 07:16:53 sip /usr/local/sbin/kamailio[29264]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: request accounted: timestamp=1504783013;method=OPTIONS;from_tag=eUC9UgGrD;to_tag=j90H6yy5pU6Xc;call_id=2xJd0KCt-P;code=200;reason=OK;calling=46566464645;called=you;egressIP= kamailio.org;src_ip=1.0.0.127;to=sip:+918866213320@sip.call.org ;tag=j90H6yy5pU6Xc;from=sip:918888888888@localhost;contact=;pai=;uri=sip: you(a)kamailio.org;onnet=0;rate=0.000333333" Thanks.. # #

2 1

crash after using async_route()
by Vitaliy Aleksandrov 07 Sep '17

07 Sep '17

Hello kamailio list, Recently found a problem in my configuration that uses async_route() functionality. It crashes after several calls when wait_timer fires. #0 0xb74a8556 in raise () from /lib/libc.so.6 #1 0xb74a9d78 in abort () from /lib/libc.so.6 #2 0x08293ae2 in qm_free (qmp=0xad65d000, p=0x3d64692d, file=0xb6216a16 "tm: h_table.c", func=0xb621663c <__FUNCTION__.18751> "free_cell_helper", line=187, mname=0xb621664d "tm") at core/mem/q_malloc.c:471 #3 0xb613f103 in free_cell_helper (dead_cell=0xae2cd210, silent=0, fname=0xb6239ea5 "timer.c", fline=655) at h_table.c:187 #4 0xb61e7758 in wait_handler (ti=557858937, wait_tl=0xae2cd258, data=0xae2cd210) at timer.c:655 #5 0x0826a2cc in timer_list_expire (t=557858937, h=0xad6b9668, slow_l=0xad6ba144, slow_mark=312) at core/timer.c:874 #6 0x08267cb1 in timer_handler () at core/timer.c:939 #7 0x0826a4d3 in timer_main () at core/timer.c:978 #8 0x08069575 in main_loop () at main.c:1721 #9 0x080707ca in main (argc=11, argv=0xbf85f044) at main.c:2723 When crash happens, kamailio prints the following message: Sep 4 16:15:38 [18938]: : <core> [core/mem/q_malloc.c:469]: qm_free(): BUG: qm_free: bad pointer 0x70707553 (out of memory block!) called from tm: h_table.c: free_cell_helper(187) - aborting Also had a few crashes in retransmission_handler(): #0 0xb750b556 in raise () from /lib/libc.so.6 #1 0xb750cd78 in abort () from /lib/libc.so.6 #2 0xb6249b5a in retransmission_handler (r_buf=0xae036674) at timer.c:367 #3 0xb6247558 in retr_buf_handler (ticks=1234464444, tl=0xae036688, p=0x1f40) at timer.c:594 #4 0x0826a2cc in timer_list_expire (t=1234464444, h=0xad71c668, slow_l=0xad71cd44, slow_mark=2232) at core/timer.c:874 #5 0x08267cb1 in timer_handler () at core/timer.c:939 #6 0x0826a4d3 in timer_main () at core/timer.c:978 #7 0x08069575 in main_loop () at main.c:1721 #8 0x080707ca in main (argc=11, argv=0xbff64134) at main.c:2723 ERROR: tm [timer.c:366]: retransmission_handler(): transaction 0xae0365e0 scheduled for deletion and called from RETR timer (flags 6d) Both timers fired for an INVITE transaction that was previously suspended by async_route(), then resumed, sent out and received a 4xx reply (407). This configuration worked fine with kamailio 4.2.x and problem appeared after upgrading to 5.0.2. Trying to figure out how to narrow down the problem. Any input is appreciated.

2 6

alias_db muliple usernames for alias
by Aidar Kamalov 07 Sep '17

07 Sep '17

Hello, I have modparam("alias_db", "append_branches", 1) and table dbaliases +----+----------------+-----------------+----------+-----------------+ | id | alias_username | alias_domain | username | domain | +----+----------------+-----------------+----------+-----------------+ | 1 | 1111 | sipchel | 34200 | sipchel | | 2 | 1111 | sipchel | 34201 | sipchel | +----+----------------+-----------------+----------+-----------------+ If I call to 1111 ringing only 34200, because first invite is right (resolving to username contact), but branch inivite is wrong. Is it bug or it is my my fault? First: INVITE sip:34200@172.16.3.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.0 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> Second: INVITE sip:34201@sipchel SIP/2.0 Record-Route: <sip:192.168.10.54;lr=on;ftag=1898fba2a5729040o0;did=958.fd11> Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.1 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> -- Aydar A. Kamalov

1 0

handle_tcpconn_ev(): connect failed
by Donat Zenichev 07 Sep '17

07 Sep '17

Hi. Recently I've come across with TCP connection problem. The topology is as following: DNS srv load balancer - two kamailio proxy servers - one routing server. Client appeals to NAPTR record like: sip.domain.com So dns returns one of the proxy servers to client (depending on weight/priority). Now both kamailio have the same priority and weight (the goal is load balancing). Routing server (now it is asterisk) working with chan_pjsip.so, that supports NAPTR/SRV records. He is able to resolve Record-Route / Route headers with value - sip.domain.com (that proxy servers add to record-route headers while relaying requests to him). This topology is done to support present dialogs, even if proxy that recently processed it, is dead. But the problem comes, when routing server (asterisk) sends in-dialog requests to the proxy, that wasn't used to establish the dialog. Example, routing server obtains 200 OK from endpoint (relayed by kamailio1 to him) and he sends back ACK, but not to the kamailio1, he sends it to kamailio2 (because he resolves NAPTR sip.domain.com and gets ip of second kamailio). Kamailio2 processes the request as usual, because both kamailio have the same db for dialog module, but when he tries to relay the request to endpoint, he gots the error: ERROR: <core> [tcp_main.c:4070]: handle_tcpconn_ev(): connect XXX.XXX.XXX.XXX:52185 failed The port that kamailio2 tries to use to relay the ACK, is port that endpoint used to establish the dialog with kamailio1 and actually his TCP connection is now established with kamailio1. So kamailio2 tries to use the same port and gets the error. And this is proper behavior I think. There is no problem with UDP transport. Has anyone seen the similar problem? That indeed is not a problem, but proper behavior. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

2 4

cr failure routes
by Nelson Migliaro 07 Sep '17

07 Sep '17

Hello, I am using carrierroute in Kamailio (great module btw). All calls go the same gateway (rewrite_host). I am routing over differents vendor using a prefix. Prefix: 21 dnis: 999999999 $rU after rewrite_host is 21999999999 My problem is when trying to use a failure route, next domain use a different prefix but the host. So I get the: Candidate destination <gateway> was previously used Is there a flag or something to prevent this verification ? Thank you Nelson.-

2 1

How can client get registered users list from Kamailio server or server may broadcast this actively?
by Jack Wang 07 Sep '17

07 Sep '17

Hey guys, As indicated by the title, is there any module providing the related function? My server version is 4.2. Thanks. :)

2 1

Cost deduction in kamailio
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, How to deduct the call cost or credits for the subscribers with the cnxcc module? Input Please Thanks & Kind Regards, Logeshwaran G

3 15

Query in LCR
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, I am handling Two Different Account holders and Two Different Gateways with the different lcr_ids: id :1 lcr_id:1 gw_name:car1 ip_addr : 1.2.3.4 id :2 lcr_id:2 gw_name:car2 ip_addr : 1.2.5.3 In kamailio config file I am using load_gws(1), Here Comes my doubt, I need to Pass the lcr_id value inside the load_gws(lcr_id) Dynamically, Is it Possible , Please Guide me. Thanks & Kind Regards, *Logeshwaran G* *"Be Hunger, Stay Foolish"*

2 2

Doubt in Lcr_table
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, May I know the Difference between the "prefix" column in Lcr_gw table and lcr_rule table. I am using the prefix value in lcr_rule table, like 00091. But What is the use of Prefix Column in lcr_gw table. Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hunger, Stay Foolish"*

2 2

Loading keys with key_type="1" from database for htable - Module params help needed
by George Diamantopoulos 06 Sep '17

06 Sep '17

3 3

Call pickup with feature code
by Cyrille Demaret 06 Sep '17

06 Sep '17

Hi, I'm trying to implement call pickup with a feature code for phones that doesn't support "Replaces" header. User has to dial for example *8 to pickup a call who is in the same group pickup. My idea is to loop for each members of the group (coming from a SQL table using sqlops) and check for calls in early state. I was thinking to use the dialog module to search for active calls but I don't find any function allowing to do this. Does anyone know a module with a function allowing to find an active calls by callee? Thank you, Cyrille

1 0

SQL for ordered LCR rules
by Patrick Wakano 06 Sep '17

06 Sep '17

Hello list, Hope you all doing well! I am trying to ease the management of LCR routing rules, since once we begin to have multiple prefixes, multiple GWs and so on, the visualization and management of the rules priorities becomes exponentially hard to do. So first thing I am trying to achieve is an easy way of retrieving the rules in an ordered manner. I couldn't find any tool to do such thing and source code was not very friendly.... so I've come up with this Postgresql query that I think retrieves all rules in the same order I expect LCR to select the GWs. SELECT lr.lcr_id, lr.prefix, lrt.priority, lg.gw_name, lg.ip_addr FROM lcr_rule lr JOIN lcr_rule_target lrt ON lrt.lcr_id = lr.lcr_id AND lrt.rule_id = lr.id JOIN lcr_gw lg ON lg.lcr_id = lr.lcr_id AND lg.id = lrt.gw_id WHERE lr.enabled = 1 AND lg.defunct = 0 AND lr.lcr_id = ID AND lr.prefix SIMILAR TO '(|PREFIX%)' ORDER BY lr.lcr_id, LENGTH(lr.prefix) DESC, lrt.priority; It is missing the weights calculation, but it is rather complex and I am not using it anyway.... Other than that does anyone did something similar to check if my query really matches what LCR engine does? Thanks, Patrick Wakano

4 3

Re: [SR-Users] SQL for ordered LCR rules
by Patrick Wakano 06 Sep '17

06 Sep '17

Thanks for the response guys! The link https://github.com/dimitri/prefixIt is returning 404.... Regarding the performance itself I am not worried since this select it is just for management and I don't expect having millions of rules. The idea is just to have an easy way to have a picture of how the LCR will order and select the gateways based on a given prefix. The three LCR tables are not so easy to handle and manage from command line so my idea was to have a single SELECT or VIEW to return me all I need at once! >From what I could check, I think the select I sent pretty much translates what LCR module does internally, I am just trying to verify if it has some flaw, which could mislead me in the rules management. Cheers, Patrick Wakano On 6 September 2017 at 00:32, Dmitry Sinina <dmitry.sinina(a)onat.edu.ua> wrote: > https://yeti-switch.org/demo.html > > > On 9/5/17 5:29 PM, Dmitry Sinina wrote: > >> And you can try our opensource LCR engine. We use kamailio as load >> balancer and SEMS as SBC. >> >> On 9/5/17 3:02 AM, Patrick Wakano wrote: >> >>> Hello list, >>> >>> Hope you all doing well! >>> I am trying to ease the management of LCR routing rules, since once we >>> begin to have multiple prefixes, multiple GWs and so on, the visualization >>> and management of the rules priorities becomes exponentially hard to do. >>> So first thing I am trying to achieve is an easy way of retrieving the >>> rules in an ordered manner. I couldn't find any tool to do such thing and >>> source code was not very friendly.... so I've come up with this Postgresql >>> query that I think retrieves all rules in the same order I expect LCR to >>> select the GWs. >>> >>> SELECT lr.lcr_id, lr.prefix, lrt.priority, lg.gw_name, lg.ip_addr >>> FROM lcr_rule lr >>> JOIN lcr_rule_target lrt ON lrt.lcr_id = lr.lcr_id AND lrt.rule_id = >>> lr.id <http://lr.id> >>> JOIN lcr_gw lg ON lg.lcr_id = lr.lcr_id AND lg.id <http://lg.id> = >>> lrt.gw_id >>> WHERE lr.enabled = 1 AND lg.defunct = 0 AND lr.lcr_id = ID AND lr.prefix >>> SIMILAR TO '(|PREFIX%)' >>> ORDER BY lr.lcr_id, LENGTH(lr.prefix) DESC, lrt.priority; >>> >>> It is missing the weights calculation, but it is rather complex and I am >>> not using it anyway.... Other than that does anyone did something similar >>> to check if my query really matches what LCR engine does? >>> >>> Thanks, >>> Patrick Wakano >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >

2 4

real-time update question
by Stent, Bob 05 Sep '17

05 Sep '17

I am just starting out with Kamailio. One of the things I want to be able to do is add a domain to the configuration and have Kamailio pick it up without having to restart. I see there is a runtime update framework but I am having trouble navigating the documentation to find out what this actually does for me. Can somebody point me to the right set of docs to read to figure this out? Do I simply add an alias to the kamalio.cfg file and somehow send an RPC command to Kamalio to re-read configuration? Bob Stent Aspect Software 5 Technology Park Drive Suite #9 Westford, MA 01886 This email (including any attachments) is proprietary to Aspect Software, Inc. and may contain information that is confidential. If you have received this message in error, please do not read, copy or forward this message. Please notify the sender immediately, delete it from your system and destroy any copies. You may not further disclose or distribute this email or its attachments.

3 2

5.0 Version compiling gives warning
by Yasin CANER 05 Sep '17

05 Sep '17

3 3

Dialplan Module
by Nelson Migliaro 05 Sep '17

05 Sep '17

Hello, Do you know if its possible to

2 2

Re: [SR-Users] Incorrect value when setting number like values in Htable using Kamcmd
by Tony - Telecomy 05 Sep '17

05 Sep '17

Prefixing the numbers with "s:" fixed the issue. Thank you for your help. Tony

1 0

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