- sr-users - kamailio.org

Negative value returned when using sql_pvquery
by Jonathan Hunter 06 Jan '16

06 Jan '16

Hi Guys, Wonder if anyone has experienced this before, on kamailio 4.1. I am using an sql_pvquery and it works fine, unless I pull a value from the database that is defined as 10 digits and starts with a 3, actually its populated as 03785702370 but due to the table it removes the leading 0 (bigint(20) unsigned). Running a manual query in my database for a DestinationMsisdn I get; select DestinationMsisdn,SourceMsisdn from MsisdnPoolAllocations where PoolMsisdn='+44162489xxxx';+-------------------+--------------+| DestinationMsisdn | SourceMsisdn |+-------------------+--------------+| 3785702370 | 4474526xxxxx |+-------------------+--------------+ However running in the kamailio.cfg an sql_pvquery to do the same query as below; sql_pvquery("cd","select DestinationMsisdn,SourceMsisdn from MsisdnPoolAllocations where PoolMsisdn='$rU'","$var(MOdest),$var(NewSourceMSISDN)"); However this returns a value of -509264926 for $var(MOdest) which should just be the 3785702370 number. What can cause kamailio to interpret this as a negative value? Has anyone seen this before? Many thanks Jon

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Re: [SR-Users] Fosdem 2016
by DanB 06 Jan '16

06 Jan '16

Hey Daniel, Please also count me in (and possible 2 other friends) for dinner/event out. Thanks, DanB

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Kamailio World 2016 – Call For Speakers
by Daniel-Constantin Mierla 06 Jan '16

06 Jan '16

Hello, should anyone here be interested in submitting a proposal for a presentation at Kamailio World 2016, note that the process is open now. More details were posted on a news article at: - http://www.kamailio.org/w/2016/01/kamailio-world-2016-call-for-speakers/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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Kamailio Password Encryption
by micho fr 05 Jan '16

05 Jan '16

Hi Guys, I have a kamailio server 1.3 with clear text Password in Subscriber table.... I wish I can encrypt that password by doing some kind of combinations between username, domain and let kamailio server be reading the encrypted password Is there a way to do it? modparam("auth_db", "db_url", DBURL)modparam("auth_db", "calculate_ha1", 0)modparam("auth_db", "password_column", "password")#modparam("auth_db", "password_column", "ha1")modparam("auth_db", "password_column_2", "ha1b")modparam("auth_db", "load_credentials", "")modparam("auth_db", "use_domain", MULTIDOMAIN)

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(no subject)
by Safdar Khan 05 Jan '16

05 Jan '16

I have made a calling service through kamailio and rtpengine.In this setup of calling service i have also included webRTC for cross platform(like android phone to web browser).Same calling service is not working after addition of freeswitch.Additional integration of Freeswitch in this current setup is making trouble. I have followed below link to setup my rtpengine module. https://github.com/caruizdiaz/kamailio-ws <https://github.com/caruizdiaz/kamailio-ws> My requirement is to integrate all these three to make cross platform calling system Kamailio<-->Rtpengine<-->Freeswitch But the calls are not relaying to freeswitch via rtpengine. Help me out to relay calls from rtpengine to freeswitch in this setup. Thanks and Regard Md Safdar Khan

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Re: [SR-Users] Parser error when enabling siptrace with HEP functionality
by Sergey Basov 05 Jan '16

05 Jan '16

Thanks for reply. I check one more time, and does not see any non SIP (HEP) packets received on port 5060 of kamailio. May bee parser trying to parse (HEP) packet when it going out via socket to capture server? When I turn of topoh module I does not see any errors in log. And in both cases I see HEP packets on kamailio capture server. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com SATB-RIPE SATB-UANIC tel: (+38067) 403-62-54

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Configure Kamailio to generate OPTIONS
by Mititelu Stefan 05 Jan '16

05 Jan '16

Hello, What is the simplest way to configure Kamailio to generate periodical OPTIONS to the caller, for an established dialog? Can you give me some starting references? Thanks, Stefan

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Is it possible to use kamailio as a Webrtc gateway and integrate it into to a conference bridge for handling webrtc clients in conference bridge?
by suganthi karthick 05 Jan '16

05 Jan '16

Hi all, I need to implement a WebRTC gateway for an existing conference bridge. The WebRTC gateway has to support Signaling, ICE and DTLS. The webrtc clients can be JsSIP or any webrtc client. The conference bridge is an existing working one for SIP clients, and I am trying to add webrtc support for that. The webrtc gateway needs to be implemented in a way like a library because it needs to be integrated into the existing platform. There are some init functions and config function from the existing conference platform, based on which the webrtc gateway has to be configured. Also, when a webrtc call come from a webrtc client, it needs to handle the signaling and the media(RTP) has to go to the conference bridge platform. It would be really helpful if you suggest whether I can use openSIPS for this purpose and use it as a library and integrate into the exiting platform? Your suggestions will be more helpful. Thanks.

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Parser error when enabling siptrace with HEP functionality
by Sergey Basov 05 Jan '16

05 Jan '16

Hi, All. I have a strange issue occuring, when I enable the siptrace functionality within the routing logic, I begin to get a lot of parser error as show below. Is this normal behavior? This errors appears only when topoh module is enabled... As I see with wireshark, there is no duplicates or returned HEP packets... I enable siptrace as follows: request_route { sip_trace(); setflag(22); ... } Configuration of the topoh and siptrace modules: # ----- topoh params ----- modparam("topoh", "mask_key", "123456789") modparam("topoh", "mask_ip", "127.0.0.8") #Siptrace modparam("siptrace", "duplicate_uri", "sip:10.1.23.20:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) modparam("siptrace", "force_send_sock", "sip:10.1.23.23:5060") modparam("siptrace", "traced_user_avp", "$avp(s:user)") output from kamailio.log with debug enabled: Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [th_msg.c:180]: th_mask_via(): body: 120: [SIP/2.0/UDP 10.10.206.39:5060 ;received=10.10.206.39;TH=dcv;branch=z9hG4bK-d8754z-2e9df22b7c5cb6c9-1---d8754z-;rport=5060] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [th_msg.c:191]: th_mask_via(): +body: 199: [SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-s7wTDLa0zUfYZXl5zpl0zpl0z.lRD.zok.sEG.lvBJY.euwReuTfzpl0zpl0z.lRD.zokqZaxuZ.P.c3BJX01R5fO.wCZLZ3WSqMkhBqGgCczJsoeV1SzJa21LY.1.e.kW7jDW7ceh52GpZKDpcSBVfSPh7qzh1E] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [th_msg.c:322]: th_mask_record_route(): no record route header Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: tm [t_hooks.c:266]: run_trans_callbacks_internal(): DBG: trans=0x7fcfacccb7a8, callback type 4194304, id 0 entered Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:766]: sip_trace_store_db(): database connection not initialized Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1875]: pipport2su(): the port string is 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1875]: pipport2su(): the port string is 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [proxy.c:265]: mk_proxy(): doing DNS lookup... Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [resolve.c:1208]: srv_sip_resolvehost(): 10.1.23.20:9060 proto=1 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [resolve.c:1329]: srv_sip_resolvehost(): returning 0x9da020 (10.1.23.20:9060 proto=1) Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1687]: trace_send_hep_duplicate(): setting up the socket_info Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1694]: trace_send_hep_duplicate(): force_send_sock activated, grep for the sock_info Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==11 && [10.1.23.23] == [10.56.41.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==11 && [10.1.23.23] == [10.56.42.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.1.23.23] == [10.1.23.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1701]: trace_send_hep_duplicate(): found socket while grep: [10.1.23.23] [10.1.23.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [parser/msg_parser.c:602]: parse_msg(): parse_msg: invalid message Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023�#023�#0128*#027#0128*#024zx�V�#016#003> Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [topoh_mod.c:214]: th_prepare_msg(): outbuf buffer parsing failed! Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: tm [t_funcs.c:362]: t_relay_to(): SER: new transaction fwd'ed Is there any way I can disable seeing these messages? What are these messages? Any thoughts are appreciated. Thanks in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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Kamailio-IMS icscf erro
by Kranti Kumar 05 Jan '16

05 Jan '16

Hi, We are using attached two configuration file to start icscf node. But while we run "kamailio start" command getting below errors. loading modules under config path: /usr/lib/i386-linux-gnu/kamailio/modules/ ERROR: bad config file (1 errors) Segmentation fault Can anyone help me here to resolve this issue. Regards, -kranti

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Too many packets in rtpengine UDP receive queue
by Zodiac 04 Jan '16

04 Jan '16

Hi friends: I am running rtpengine daemon on a CentOS machine functionally. SIP server is Kamailio. Vedio and audio calls are both OK with rtpengine. But there is always prompts like following in rtpengine’s log: Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30136] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30156] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30156] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible So I am wondering whether this prompt is normal. Otherwise, what should I do to prevent this to happen? Thx a lot! ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

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Re: [SR-Users] Presence - Subscription based NOTIFY to enable MWI
by Arsen Hovhanissian 03 Jan '16

03 Jan '16

Alright I guess this is one of those duhh… moments, all this time I was forgetting to add the content length to the NOTIFY event. Remark: DO NOT FORGET the content length! Thanks for the help, very much appreciated!

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Re: [SR-Users] Presence - Subscription based NOTIFY to enable MWI
by Arsen Hovhanissian 03 Jan '16

03 Jan '16

I see, what i’m trying to do is actually to limit querying the DB by updating the MWI only when there is activity. In other words to send a NOTIFY when the user receives a new message in his voicemailbox. I am not sure if it’s possible to send the NOTIFY for example 5-30 mins after the initial subscribe.

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Presence - Subscription based NOTIFY to enable MWI
by Arsen Hovhanissian 03 Jan '16

03 Jan '16

Hi everyone, I’m trying to send a NOTIFY event using “sipsak” to enable the MWI I read up a lot of documentation and didn’t really find the information needed to accomplish this. So I have: Phone 1 -> P1 Server 1 -> S1 (Kamailio 4.3.4) Server 2 -> S2 (sipsak) P1 registers to S1 and creates a new entry in the active watchers table with a "message-summary” event. At this point I am assuming that any request I would send will be out of dialog. Now the question is, How could I reply to that SUBSCRIPTION with a NOTIFY? Thanks in advance!

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Happy New Year!
by Daniel-Constantin Mierla 31 Dec '15

31 Dec '15

The amazing 2015 for Kamailio is done! Thank you everyone for contributing to it! Now it is the time to get prepared to fill the 2016, with interesting discussions on our forums, solving new RTC challenges, adding features in Kamailio and welcoming new contributors! I wish a healthy and prosperous year to all Kamailio friends! I am looking forward to meeting many of you at Kamailio World Conference, where we are going to celebrate 15 years of development for Kamailio project, as well as at other events around the globe! Happy New Year! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda http://www.kamailioworld.com

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tsilo and linphone android connected via TCP
by Vasiliy Ganchev 29 Dec '15

29 Dec '15

Hi folk! sorry for first sending to the wrong list ! Currently playing around with push notification in kamailio, have a bad case. Would appreciate any idea. Scenario (all devices connected via TCP): UA-A - sipphone, UA-B[1] - sipphone UA-B[2] - Linphone android Server - kamailio+asterisk UA-A call UA-B (UA-B[1] - online, UA-B[2] - offline) Kamailio make ts_store, and t_relay to UA-B[1], for UA-B[2] kamailio send push notification UA-B[2] appear registered, I make ts_append, and see kamailio sending INVITE to UA-B[2], but tansport used is UDP (expected TCP) If I use UDP on UA-B[2] - have no issue. I am using kamailio-4.2.5, looked in sources of current master, did not found changes that could help in my scenario. General question - does anyone has same problem using ts_append for TCP connected devices? If TCP in ts_append should work OK, any other idea? cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/tsilo-and-linphone-android-connecte… Sent from the Users mailing list archive at Nabble.com.

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kamailio wheezy config files
by Ramya Y 29 Dec '15

29 Dec '15

Hi , Greetings! i have configured mpath in kamailio.cfg ,when i run kamailio component of icscf (wheezy). i am facing below Errors. *error:* root@debian:/etc/kamailio# kamailio start root@debian:/etc/init.d# kamailio start loading modules under config path: /usr/lib/kamailio/modules:/usr/lib/kamailio/modules_s:/usr/lib/kamailio/modules_K loading modules under config path: usr/local/ims-kamailio/kamailio-4.2.0/output/lib64/kamailio/modules/ loading modules under config path: /.... .../:/usr/lib/i386-linux-gnu/kamailio/modules/ ERROR: bad config file (1 errors) Segmentation fault could you please suggest me which mpath i have to configured in kamailio.cfg for wheezy. Thanks in Advance! Regards, Ramyasri

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Append to reply in default action of failure_route
by Efelin Novak 28 Dec '15

28 Dec '15

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

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notify takes several seconds to generate
by Juha Heinanen 28 Dec '15

28 Dec '15

i noticed that kamailio 4.3 presence server takes several seconds to generate notify when it receives subscribe request. for example: Dec 27 15:42:01 lohi /usr/bin/sip-proxy[7363]: INFO: Routing SUBSCRIBE <sip:test@test.tutpro.com> to presence server at <sip:127.0.0.1:5080;transport=tcp> Dec 27 15:42:01 lohi /usr/bin/pres-serv[7113]: INFO: Handled SUBSCRIBE <sip:test@test.tutpro.com> Dec 27 15:42:05 lohi /usr/bin/pres-serv[7108]: INFO: Routing locally generated NOTIFY <sip:test@test.tutpro.com;gr=urn:uuid:47759291-c9db-4eb3-a540-5e68353414ee> to <sip:test@test.tutpro.com> from <sip:test@test.tutpro.com> the host is not loaded at all. these relevant presence server params are set: modparam("presence", "subs_db_mode", 3) modparam("presence", "db_table_lock_type", 0) and db is on localhost. any explanation why notify takes so long to generate? -- juha

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how to make secure websocket run on kamalio
by tintuc tv 28 Dec '15

28 Dec '15

Hi list is there any tutorial for making secure websocket work with kamailio. the SSL Certificates is bought from SSL Certificates thank you

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kamailio 4.0.1 websocket configuration
by Rupayan Dutta 28 Dec '15

28 Dec '15

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

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Best way to store RTPProxy statistics in database
by Gholamreza Sabery 28 Dec '15

28 Dec '15

I want to store RTPProxy statistics in $rtpstat variable (exported by RTPProxy module) into database. Currently I use app_python module to run a specific Python script with from_tag, to_tag, callid and $rtpstat as parameters. This module puts this information in a message queue which is red by another process. That process writes the data in the database. Is there a better, more efficient way to store arbitrary data like this into database in Kamailio? Regards

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Asterisk failover
by Phil Daws 28 Dec '15

28 Dec '15

Seasons Greetings! Am starting to learn about Kamailio fronting Asterisk and wish to ask how one performs multi-site failover. What I would like to be able to do is when the upstream trunk provider routes a SIP call to me if Kamailio detects that the backend Asterisk is down it will send to a secondary system ? Thanks, Phil

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Too many packets in rtpengine UDP receive queue
by Zodiac 27 Dec '15

27 Dec '15

Hi friends: I am running rtpengine daemon on a CentOS machine functionally. SIP server is Kamailio. Vedio and audio calls are both OK with rtpengine. But there is always prompts like following in rtpengine’s log: So I am wondering whether this prompt is normal. Otherwise, what should I do to prevent this to happen? Thx a lot! ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

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Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 25 Dec '15

25 Dec '15

With an amazing 2015 almost gone, I am using this moment to give my thanks and greetings to the people involved in Kamailio project, old and new friends, developers, contributors, the engaged and warm community members. Very soon 2016 will arrive, Kamailio will celebrate 15 years of development, therefore we are looking to a special year ahead! Merry Christmas and Happy Winter Holidays! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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AUTH doesn't work
by Evgeny Zhavoronkov 23 Dec '15

23 Dec '15

Hello! Here is my config: http://pastebin.com/X20JumYB Why can't I authorize? I get: 403 Not relaying (kamailio (4.3.4 (x86_64/linux))) Interesting that with previous version everything was ok. I just use my previous config. -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

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Issue with end to end MESSAGE over TCP
by Thierry Luo 23 Dec '15

23 Dec '15

Hi All, I have deployed a Kamailio proxy server (as IMS pCSCF) in Amazon EC2. UA registration over TCP and presence flow (PUBLISH/NOTIFY) are all fine. But MESSAGE from one UA to another is UA over TCP is always not successful. I captured the sip flow from the server, and found that at the final hop from Proxy to terminating UA, the Kamailio server send the MESSAGE over udp, regardless of the fact that UAs were registered over TCP. What is the problem maybe? Attached is the cfg file. Thanks. Thierry Luo

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rtpproxy does not response, disable it
by Zodiac 21 Dec '15

21 Dec '15

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

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lost mysql connections
by Juha Heinanen 21 Dec '15

21 Dec '15

for some reason once in a while kamailio processes fail to execute mysql queries while at the same time an mysql test script that is run once per second from cron works fine. kamailio executes mysql queries during normal sip message processing and also when it processes an OPTIONS ping test request: sql_query("sip_proxy", "SELECT table_version FROM version WHERE table_name='domain'", "result"); if ($rc == 1) { sql_result_free("result"); options_reply(); } else { xlog("L_ERR", "$rm <$ru> failed sql test\n"); sql_result_free("result"); send_reply("500", "Server Internal Error"); }; here is an example: ... Dec 15 00:51:08 proxy1 php: MySQL server test succeeded Dec 15 00:51:09 proxy1 php: MySQL server test succeeded Dec 15 00:51:10 proxy1 /usr/bin/sip-proxy[29894]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query Dec 15 00:51:10 proxy1 /usr/bin/sip-proxy[29894]: ERROR: db_mysql [km_dbase.c:145]: db_mysql_async_exec_task(): failed to execute query on async worker Dec 15 00:51:10 proxy1 php: MySQL server test succeeded Dec 15 00:51:11 proxy1 php: MySQL server test succeeded Dec 15 00:51:13 proxy1 php: MySQL server test succeeded Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29891]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29891]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29883]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:14 proxy1 php: MySQL server test succeeded Dec 15 00:51:14 proxy1 /usr/bin/sip-proxy[29886]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:15 proxy1 php: MySQL server test succeeded Dec 15 00:51:15 proxy1 /usr/bin/sip-proxy[29885]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:16 proxy1 php: MySQL server test succeeded Dec 15 00:51:17 proxy1 php: MySQL server test succeeded Dec 15 00:51:18 proxy1 php: MySQL server test succeeded ... all db_mysql module params are at their default values. any idea why kamailio at connection loss does not automatically create new connections? is there a means to somehow debug this further? -- juha

2 3

Kamailio and Skype for Business (MS Lync)
by Grant Bagdasarian 21 Dec '15

21 Dec '15

Hello, Does anyone have any experience configuring Kamailio as a gateway for Skype For Business (Microsoft Lync)? Regards, Grant

3 2

ACC module from_tag, to_tag and callid
by Gholamreza Sabery 21 Dec '15

21 Dec '15

In the ACC modules' database table (acc) in Kamailio sometimes from_tag, to_tag and callid of an INVITE and corresponding BYE completely match. But sometimes from_tag in INVITE is to_tag of BYE and to_tag of BYE will become from_tag of INVITE. Why this happens? Is this normal? Regards

3 6

crash in version 4.0
by Juha Heinanen 21 Dec '15

21 Dec '15

core dump occurred in kamailio version 4.0. does anyone remember if a bug causing such a crash would possibly have been fixed in some later kamailio version? -- juha ------------------------------------------------------------------------------ Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21089]: WARNING: db_mysql [km_dbase.c:85]: driver error on ping: Lost connection to MySQL server during query Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21116]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 90 Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21016]: ALERT: <core> [main.c:788]: child process 21089 exited by a signal 11 Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21016]: ALERT: <core> [main.c:791]: core was generated ------------------------------------------------------------------------------ Program terminated with signal 11, Segmentation fault. #0 run_trans_callbacks_internal (cb_lst=<value optimized out>, type=512, trans=0x7f093593a9b0, params=0x7fff7a5289c0) at t_hooks.c:286 286 t_hooks.c: No such file or directory. in t_hooks.c (gdb) bt full #0 run_trans_callbacks_internal (cb_lst=<value optimized out>, type=512, trans=0x7f093593a9b0, params=0x7fff7a5289c0) at t_hooks.c:286 cbp = 0x3030 backup_from = 0x8a49d0 backup_to = 0x8a49d8 backup_dom_from = 0x8a49e0 backup_dom_to = 0x8a49e8 backup_uri_from = 0x8a49c0 backup_uri_to = 0x8a49c8 backup_xavps = 0x8a4ad0 #1 0x00007f0941ad3ddc in run_trans_callbacks_with_buf (type=1887007839, rbuf=<value optimized out>, req=<value optimized out>, repl=0x6565705f74736423, flags=0) at t_hooks.c:336 params = {req = 0x7f0935b32360, rpl = 0xffffffffffffffff, param = 0x7f0935940900, code = 408, flags = 408, branch = 0, t_rbuf = 0x7f093593aa70, dst = 0x7f093593aac0, send_buf = { s = 0x7f093594d180 "From: xxx <sip:xxx@xxx:5060;user=phone>;tag=262063958\r\nTo: xxx <sip:xxx@xxx:5060;user=phone>;tag=as2db5b848\r\nCall-ID: 2864df7e-1d31c032-6dd726"..., len = 462}} trans = 0x246 #2 0x00007f0941aef8cb in relay_reply (t=0x7f093593a9b0, p_msg=<value optimized out>, branch=<value optimized out>, msg_status=<value optimized out>, cancel_data=<value optimized out>, do_put_on_wait=<value optimized out>) at t_reply.c:1880 relay = 0 save_clone = 0 buf = 0x7f0943b3d140 "SIP/2.0 408 Request Timeout\r\nVia: SIP/2.0/UDP xxx:5060;branch=z9hG4bKterm-2b3acd-xxx-xxx-72778\r\nFrom: xxx <sip:xxx@xxx:5060;user=phone>;tag=17"... res_len = 462 relayed_code = 408 relayed_msg = 0xffffffffffffffff reply_bak = 0x13c400000050c7d3 bm = {to_tag_val = { s = 0x7f0943b3d259 "0b5775687262b458dc9c8d93773b9caa-d95c\r\nCall-ID: 1be7fd61-4cf116a3-62498d9c-8757@xxx\r\nCSeq: 1 INVITE\r\nServer: Kamailio (4.0.1 (x86_64/linux))\r\nContent-Length: 0\r\n\r\n", len = 37}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7f093593aa70 to_tag = <value optimized out> reason = {s = 0x56dbc5 "Request Timeout", len = 15} onsend_params = {req = 0x0, rpl = 0x1, param = 0x7f093593ab18, code = 898869680, flags = 32521, branch = 0, t_rbuf = 0x7d9c88, dst = 0xfffffe22, send_buf = { s = 0x7f093593ab20 "", len = 1101857122}} #3 0x00007f0941af5ac1 in fake_reply (t=0x7f093593a9b0, branch=0, code=<value optimized out>) at timer.c:354 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 898870080}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 898870080}}}} reply_status = <value optimized out> #4 0x00007f0941af5ec8 in final_response_handler (ticks=1904446327, tl=0x7f093593ab40, p=<value optimized out>) at timer.c:526 prev_branch = -478 now = 0 #5 retr_buf_handler (ticks=1904446327, tl=0x7f093593ab40, p=<value optimized out>) at timer.c:584 rbuf = 0xfffffe22 fr_remainder = <value optimized out> retr_remainder = <value optimized out> retr_interval = <value optimized out> new_retr_interval_ms = <value optimized out> #6 0x00000000004fa080 in slow_timer_main () at timer.c:1151 n = 12 ret = 12 tl = 0x7f093593ab40 #7 0x0000000000458039 in main_loop () at main.c:1693 i = 16 pid = <value optimized out> si = 0x0 si_desc = "sctp receiver child=15 sock=xxx:5060\000\000\000\200\215Rz\377\177\000\000\000\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000\001\000\000\000\004\000\000\000\005\000\000\000\000\000\000\000\020\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000 \251\212", '\000' <repeats 13 times>, "\020\000\000\000\000\000\000" nrprocs = 16 #8 0x00000000004599fb in main (argc=17, argv=<value optimized out>) at main.c:2566 cfg_stream = <value optimized out> c = <value optimized out> r = 0 tmp = 0x7fff7a529f11 "" tmp_len = 0 port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 2572917976 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7fff7a528d80 p = <value optimized out> (gdb)

2 1

rtpengine module rtpengine_disable_tout question
by Juha Heinanen 19 Dec '15

19 Dec '15

rtpengine readme tells: 4.2. rtpengine_disable_tout (integer) Once an RTP proxy was found unreachable and marked as disabled, the rtpengine module will not attempt to establish communication to that RTP proxy for rtpengine_disable_tout seconds. Default value is "60". does this mean that after 60 seconds is passed, rtpengine module will again try to use that RTP proxy, i.e., the RTP proxy does never become permanently disabled unless disabled by nh_enable_rtpp mi command? does the this param apply also if all RTP proxies of a set (perhaps consisting of only one proxy) are unreachable? -- juha

2 2

Kamailio v4.1.9 Released
by Daniel-Constantin Mierla 18 Dec '15

18 Dec '15

Hello, Kamailio SIP Server v4.1.9 stable release is out. This is marking the end of maintenance for 4.1.x series. Kamailio project maintains the last two stable branches, at this moment these are 4.2 and 4.3. Therefore, anyone running older releases is advised to upgrade in the near future to the latest version of one of those branches, preferably 4.3, in order to benefit of proper maintenance. For more details about version 4.1.9 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/12/kamailio-v4-1-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.4. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

websocket sometimes log errors after ws connect has closed
by Vasiliy Ganchev 18 Dec '15

18 Dec '15

Hi list! firstly, the described below issue is harmless to us, and is just little bit annoying, but analysing the nature of issue decided to share the results, and hear the opinion of community. Scenario: - WebRTC client connected to Kamailio's websocket - after some time I close tab with WebRTC, in browser. - using "kamctl fifo ws.dump" and "netstat -npa |grep kama |grep tcp" I can see that the connection is closed - but Kamailio still triing to send keepalive to that connection, and in syslog I got errors: Dec 18 12:28:18 /usr/sbin/kamailio[4471]: WARNING: websocket [ws_frame.c:245]: encode_and_send_ws_frame(): TCP/TLS connection get failed Dec 18 12:28:18 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_frame.c:797]: ping_pong(): sending keepalive Analysing sources, and using debug level 3 I see such logs (after the webrtc client is closed, and ) Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:468]: wsconn_get_list(): counter wsc [0xb1acb270] prev => [(nil)] next => [(nil)] Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:495]: wsconn_get_list(): wsc [0xb1acb270] id [2] ref++ Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:504]: wsconn_get_list(): wsconn_get_list returns list [0xb666f410] with [1] members What is strange: in some tests, after unsuccessful sending of Ping, kamailio's "wsconn_get_list" clean up that connection, but in some tests, kamailio try sending Ping to old connection during 5-10 minutes (this time is not constant from test to test) Suddenly my current knowledge of C is not enough to find out were is the problem's root. General question is: is this correct list for such issuing? Or it should be in Dev list? once again - the problem is harmful (afetr some time, kamailio stop trying sending ping to inactive connection, but it is annoying to see errors in logs) Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/websocket-sometimes-log-errors-afte… Sent from the Users mailing list archive at Nabble.com.

1 1

issue with BYE
by gerry kernan 18 Dec '15

18 Dec '15

Hi I'm looking for any pointers to resolve and issue I have . When callee does a hang-up the call terminates on kamailio and asterisk but the caller's phone handset doesn't receive the BYE and doesn't hang-up. If the caller does the hang up then the call terminates and the callee's handset hands up . My setup is Kamailio WAN -> Kamailio Lan -> Asterisk LAN I'm using Path and dispatcher modules to route the SIP traffic to the backup end asterisk server my the domain in the SIP request. Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - www.infinityit.ie IP Telephony Asterisk Consulting - www.asteriskconsulting.com Contact Centre Total Interact - www.totalinteract.com

3 7

Releasing Kamailio v4.1.9
by Daniel-Constantin Mierla 18 Dec '15

18 Dec '15

Hello, I am going to release v4.1.9 later today, as last to be officially packaged version from branch 4.1. It is strongly recommended to migrate to a newer version, because the focus of developers is on the two most recent stable branches. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

kamailio as simple sip forwarder / gateway / proxy
by Peter Gervai 17 Dec '15

17 Dec '15

Hello, I've been browsing the net and the (not so friendly) documentation but found no readily available solution. I have tried to fabricate on myself but it seems it's not that easy without getting deeply familarised with SIP protocol internals which I don't quite want know. The problem seems simple enough, maybe you can easily tell the solution. There is kamailio at ip 256.10.10.1. It's task is to accept connections from SIP clients and forward them to PBX at 256.20.20.1, and accept PBX packets and forward them to the clients. PBX handles registration, options, invites, everything. Kamailio does not authenticate, authorise, just forwards everything back and forth, like a mindless proxy. No NAT involved. [The reason is that K. have to call external scripts whenever an outbound call (towards the clients) is initiated, which the PBX is unable to. Please don't advise other solutions (which isn't able to call executables in-session).] I have created a simplistic config which is not expected to work and it doesn't quite (plenty of security stuff left out now): route { if(src_ip==myself) { add_path(); $du="sip:256.20.20.1:95060"; } else { if(src_ip!="256.20.20.1") { $du="sip:256.20.20.1:95060"; } else { #from gateway pkts aren't changed } if(!t_relay()) { sl_reply_error(); break; } } There is several possible problems, but it seems Contact: is one main one since it contains the client instead of the proxy. Possibly this may even be the almost-good solution. I'm sure there's an easy and pretty solution so before I start to make ugly ones I ask the wise people. Thanks, Peter

2 1

Kamailio v4.2.7 Released
by Daniel-Constantin Mierla 17 Dec '15

17 Dec '15

Hello, Kamailio SIP Server v4.2.7 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.7 (or to 4.3.x series). For more details about version 4.2.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/12/kamailio-v4-2-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.4. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio as a SBC
by Guillaume 17 Dec '15

17 Dec '15

Hi everyone, I'm trying to make kamailio works as a SBC in front of Asterisk. I have read the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and it is close to what I'm looking for but still a little different. I want don't want kamailio to do auth and I want it to be a media gateway. The media and SIP path have to look like that: Phone 1 ----- kamailio -----Asterisk ---- Kamailio ---- Phone 2 First I have add an outboundproxy field in the Asterisk configuration to make all SIP messages from Asterisk passe through Kamailio. Then I started a minimal conf for Kamailio and it works at least for the SIP part. Call can be established perfectly. However the RTP proxy part is a mess. Kamailio changes the IP address in the sdp part only for INVITE messages and not for the 200 OK responses, so there is only one side of the media passing through Kamailio, the other side is sent directly to asterisk. But even for the half part rtpproxy craches with a segfault and Kamailio doesn't transmit this media. I have been searching some helps on google but nothing so far :/ I hope that someone here could help me. ----------------------------------------------- #!KAMAILIO ####### Defined Values ######### #!define ASTERISK 192.168.51.43 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=2 auto_aliases=no # to avoid simple attacks, use unconventional port port=5060 ####### Modules Section ######## mpath="/usr/local/lib/kamailio/modules" loadmodule "db_text.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.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 "path.so" loadmodule "rtpproxy.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) modparam("path", "use_received", 1) modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## route { xlog("L-ALERT", "new message !!!!!!!!!!!!!!!"); if (!sanity_check()) { exit; } if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); xlog("L_NOTICE", "$si $rm $ru -- too many hops\n"); exit; } if (af==INET) { force_rport(); if (src_ip != ASTERISK) { $du = "sip:192.168.51.43:5060"; add_path_received(); route(RTPPROXY); record_route(); } else { loose_route(); route(RTPPROXY); record_route(); } } forward(); } route[RTPPROXY] { rtpproxy_manage("cwei"); } onreply_route { if (af==INET) { rtpproxy_manage("cwei"); } return(1); } --------------------------------------------- Guillaume

2 2

Planning to release v4.2.7
by Daniel-Constantin Mierla 17 Dec '15

17 Dec '15

Hello, later this week, on Thursday, Dec 17, I am planning to release a minor version based on branch 4.2, respectively 4.2.7 -- it's been a while since v4.2.6. As usual, if there are issues not yet reported or patches to be backported, write to sr-dev mailing list. If time allows, in the same day or next days, I plan to do the last release of branch 4.1 to mark the end of official packaging in 4.1 series. At this moment, the last two stable branches are 4.3 and 4.2. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 1

Presence Module
by Loïc Boissy 16 Dec '15

16 Dec '15

Hello, I use kamailio 4.3.2 with the presence module. When the contact is able to manage its presence, No problem for me. But when a contact doesn't manage it presence, i would like to know if it is possible to generate a sip message (like as a simple presence message) with kamailio? The goal of that is to prevent my watcher when a "simple" contact (without presence option) is registered and unregistered on my kamailio server. Thank you for your help. Best regards. Loïc BOISSY

2 4

State-full failover Kamailio+RTPProxy
by Frank Costeira 16 Dec '15

16 Dec '15

Hi, Is there anyway to do state-full failover of Kamailio+RTPProxy from a primary server to a backup server? Regards, Frank

5 7

New logo for Kamailio project
by Daniel-Constantin Mierla 16 Dec '15

16 Dec '15

Hello, based on the feedback after the proposal, there were no voices against the new logo. Therefore, as of today, Kamailio project is officially using the new logo -- see it at: - http://www.kamailio.org/w/2015/12/new-logo-for-kamailio-project/ If you display one of the old Kamailio logos, we would encourage to use the new one at your earliest convenience. That will help to propagate it faster. Posting about the change on blogs, forums or social media channels would be appreciated as well. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

RFC: Renaming dialog_ng to ims_dialog
by Daniel-Constantin Mierla 16 Dec '15

16 Dec '15

Hello, I am under the impression that the name dialog_ng creates confusion out there and some people are using it instead of the classic dialog module. Although it was started with goals of reworking dialog module with a different concept (which was discussed mainly by some guys that afterwards changed their job to non-voip area), dialog_ng ended up to be tailored for IMS needs. Probably we should do that refactoring of the dialog module, but meanwhile dialog_ng doesn't refect that and some people are confused by the current naming of the two modules. Practically is more about convenience at this moment and if IMS developers and users think it is not going to be a big overhead for their deployments to be upgraded, I can take care to rename it. So, while general opinion matters, I think we should see first what IMS devs prefer. I am personally not affected that much, so I am fine to keep it like it is now -- in that case, proper notes should be added to documentation, stating that dialog_ng must be used only for IMS (or when the config writer knows very well what she/he is doing). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

4 3

IMC module
by Alexandru Covalschi 16 Dec '15

16 Dec '15

Hello again First of all I wanted to ask if someone ever implemented that http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC Second question is - I don't understand the logic. In description is said: Handles Message method.It detects if the body of the message is a conference command.If so it executes it, otherwise it sends the message to all the members in the room. But why in example (well however it has broken syntax) to IMC manager are sent only messages from chat-rooms? How message from client can possibly reach imc-manager then? Also - when I send message with body "1111111" from user 1001 and imc_manager catches it - I receive 500 command error. Why? :/ All that is working on top of sipjs-demo. -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 3

rtpengine and serial forking with multiple network interfaces
by Alex Balashov 15 Dec '15

15 Dec '15

Hi, It is not very clear to me how to handle serial forking scenarios with rtpengine where there are multiple RTP listeners attached to multiple network interfaces. That is to say, given: rtpengine ... -i net1/xxx.xxx.xxx.xxx -i net2/yyy.yyy.yyy.yyy 1. I attempt to reach SIP gateway 1 via net1, calling rtpengine_manage() in the process: rtpengine_manage("replace-origin replace-session-connection ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)"); 2. The call to gateway 1 fails, and I roll over to gateway 2, reached via net2, with dispatcher. I have some logic that checks how $rd has changed and sets $avp(rtp_dest) accordingly in the dispatcher failure_route[]. 3. In that failure_route[], I call: rtpengine_manage("replace-origin replace-session-connection ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)"); 4. When looking at the SDP for the INVITE on branch #2 going out to gateway 2, it does not appear to utilise media endpoint yyy.yyy.yyy.yyy, but rather continues to represent itself outwardly as xxx.xxx.xxx.xxx, even to gateway #2. I can't figure out what I'm doing wrong here, and the 'rtpengine' documentation only has one mention of serial forking, in the context of the discussion of the 'via-branch' parameter. As far as I can tell, this is not relevant to my use-case; I am not concerned about deleting the whole call from rtpengine, but rather about updating the SDP offer consonantly with serial forking to gateway #2. Any help would be deeply appreciated! Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

Relaying failure codes back to initial server
by Alexandru Covalschi 15 Dec '15

15 Dec '15

Hello everyone! I need to relay 486/408/... other failure codes back to initial INVITE server. Here http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is recommended just to exit failure_route, but that didn't work for me. I need that to let Freeswitch know which cause has ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

3 4

Dispatcher according to called subscriber
by Michael Nielsen 15 Dec '15

15 Dec '15

Hello everyone, I know kamailio can be configured into almsot everything for SIP - and that the dispatcher function is already build for dispatching between multiple sbc's etc... Is it possible to use the dispatcher for routing between multiple sbc's accroding to called subscriber? So I can make it somewhat multi-tentant. So subscriber A always is routed to 1 or more specific sbc's and subscriber B to 1 or more other spcific sbc's?

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MySQL Database installation issue
by Kranti Kumar 15 Dec '15

15 Dec '15

Hi All, I am Newbie to Kamailio and while we configuring IMS Kamailio , facing issues on MySQL installation. OS : Debian 7 (Wheezy) While we issue apt-get install mysql-server-5.6, getting below error: apt-get install mysql-server-5.6 The following packages have unmet dependencies: mysql-community-server : Depends: libaio1 (>= 0.3.93) but it is not installable Depends: libnuma1 (>= 2.0.8~rc4) but it is not installable Recommends: mysql-client (= 5.6.28-1debian7) E: Unable to correct problems, you have held broken packages. Note:in mysql list added below lines: deb http://repo.mysql.com/apt/debian wheezy mysql-5.6 deb-src http://repo.mysql.com/apt/debian wheezy mysql-5.6 deb http://repo.mysql.com/apt/debian wheezy mysql-apt-config Can any one help us to resolve this problem. Thank you so much in advance. Regards, -kranti

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Re: [SR-Users] MySQL Database installation issue
by Kranti Kumar 15 Dec '15

15 Dec '15

Ganesh, After issuing command $sudo apt-get update, adding the repo we are getting the below error. root@debian:/etc/apt/sources.list.d# apt-get install mysql-server-5.6 Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'mysql-community-server' instead of 'mysql-server-5.6' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: mysql-community-server : Depends: libaio1 (>= 0.3.93) but it is not installable Depends: libnuma1 (>= 2.0.8~rc4) but it is not installable Recommends: mysql-client (= 5.6.28-1debian7) E: Unable to correct problems, you have held broken packages. Regards, -kranti On Tue, Dec 15, 2015 at 8:20 AM, Infinicalls Infinicalls < infinicalls(a)gmail.com> wrote: > Hi, > Did you try issuing the command $sudo apt-get update after adding the > repo? What error message you are getting? > > Ganesh Kumar > > > On 12/15/15, Kranti Kumar <krantikumara(a)gmail.com> wrote: > > Hi All, > > > > I am Newbie to Kamailio and while we configuring IMS Kamailio , facing > > issues on MySQL installation. > > > > OS : Debian 7 (Wheezy) > > > > While we issue apt-get install mysql-server-5.6, getting below error: > > > > apt-get install mysql-server-5.6 > > > > > > The following packages have unmet dependencies: > > > > mysql-community-server : Depends: libaio1 (>= 0.3.93) but it is not > > installable > > > > Depends: libnuma1 (>= 2.0.8~rc4) but it is not > > installable > > > > Recommends: mysql-client (= 5.6.28-1debian7) > > > > E: Unable to correct problems, you have held broken packages. > > > > > > Note:in mysql list > > > > added below lines: > > > > deb http://repo.mysql.com/apt/debian wheezy mysql-5.6 > > > > deb-src http://repo.mysql.com/apt/debian wheezy mysql-5.6 > > > > deb http://repo.mysql.com/apt/debian wheezy mysql-apt-config > > > > > > Can any one help us to resolve this problem. > > > > Thank you so much in advance. > > > > Regards, > > > > -kranti > > > > > -- > > -- > http://www.infinicalls.com >

1 0

Issue with BYE and $du
by Rene Montilva 14 Dec '15

14 Dec '15

Hi list What could be the scenario, when hangup a call and kamailio show empty the $du var and can't send BYE to endpoint -- Ing. Rene Montilva *FOSS Developer and VoIP Engineer.*

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how to send BYE to outbound leg in kamailio?
by Dileep Nunna 14 Dec '15

14 Dec '15

Hi, After credit limit reached(ims_charging), tried to send BYE to both legs in kamailo (modules/dialog_ng/dlg_req_within.c).Bye sent to caller, but when sending bye to callee it is failing.(it is considering record-routes from caller invite, not in 200 OK from callee).Is there any changes necessary? Can you please reply. Thanks, Dileep

2 1

403 Not Relaying when default port changed
by Javohir Maxmarajabov 14 Dec '15

14 Dec '15

Hello fellow developers... We have been testing Kamailio for a week and it is working great... But some of the our friends reported that they can't connect to our server using their mobile Internet... and it seems default SIP port is blocked by Service Provider... So we decided to change ports... I opened kamailio.cfg and added these lines: listen=tcp:_PUBLIC_IP_:32850 listen=udp:_PUBLIC_IP_:32850 listen=tls:_PUBLIC_IP_:32851 listen=tcp:127.0.0.1:32850 listen=udp:127.0.0.1:32850 listen=tls:127.0.0.1:32851 listen=tcp:10.19.0.5:32850 listen=udp:10.19.0.5:32850 listen=tls:10.19.0.5:32851 port=32850 port=32851 Now clients throwing an error "403 Not relaying"... I tried to enable debugging to see cfgtrace... no errors detected But when I comment out lines above, (with default cfg) everything works fine Where and What am I doing wrong? Thanks in advance :) Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

2 1

IMS-P-CSCF: stripped down version of kamailio.cfg
by Franz Edler 12 Dec '15

12 Dec '15

Dear IMS-experts, I am trying to reduce the complexity of the IMS implementation by disabling the functions which are not required for a pure stripped down IMS core platform. Therefore I deactivate NAT handling, RTP proxy, SBC, dispatcher etc... in a first step. Later on I will add the functions step-by step. According to the requirements. Now I discovered an inconsistency in the configuration file (kamilio.cfg) of P-CSCF. In main routing logic there is the point where P-CSCF distinguishes an originating-initial request from a terminating-initial request. This is done by the code: if (!ds_is_from_list()) { # Originating from Subscriber: route(Orig_Initial); } else { # Terminating to Subscriber: route(Term_Initial); } Unfortunately the function ds_is_from_list() is part of the dispatcher module which I want to avoid in a first step. Therefore another criterion must be used to distinguish between originating- and terminating-initial request. I therefore propose to use the pseudo-user-name "term" in the Route-URI as a distinguisher. What do you think? I am actually thinking about the correct statement (regular expression) to do that. Maybe anyone can help. BR Franz

1 0

transformations {s.rm,match}
by Jonathan Hunter 11 Dec '15

11 Dec '15

Hi Guys, I have been using {s.rm,match} to remove occurrences of numbers, however I am aware it removes all occurrences. Is it possible to make it only remove the first occurrence and not all? For example If I am looking to modify 639157407639 I want to remove only the first 63, not both. So I end up with 9157407639 and not 91574079. Is it possible with a transformation of is regex the way ? thanks Jon

2 1

Dispatcher Module & ds_load_list enhancement
by jay binks 11 Dec '15

11 Dec '15

Just an observation and enhancement suggestion. VERY occasionally Ive been getting *ERROR: <core> [parser/parse_param.c:591]: parse_params2(): parse_params(): No memory left* Now I know this is the root of the problem BUT ... it would be nice if it didnt leave my kamailio with no dispatchers at all. Im just suggesting that rather than calling destroy_list before re-populating the list. Could this be changed to create a new list, swap over the pointers to the new distributor list ( IF it has > 0 entries ) and then destroy_list . its just a thought ? -- Sincerely Jay

2 1

Asynchronous processing increases branch count
by Alex Balashov 11 Dec '15

11 Dec '15

Hi, Normally with serial forking, the first branch ID corresponds to $T_branch_idx == 0, i.e. Via: SIP/2.0/UDP 172.30.110.4;branch=z9hG4bK7f3d.57593ee3a139f0ae6c4a8a0f531e342c.0 In the event of failure, a second branch attempt has a branch index of 1 and so on: Via: SIP/2.0/UDP 172.30.110.4;branch=z9hG4bK7f3d.57593ee3a139f0ae6c4a8a0f531e342c.1 However, when I use asynchronous processing with t_suspend() / t_continue(), resuming the transaction in an rtimer process, I notice that the $T_branch_idx of the first branch is 1, then 2. Where did 0 go? Is this a normal effect of TM suspend/continue? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

problem with cfgtrace
by Joao Alves 11 Dec '15

11 Dec '15

Hi, Have just installed kamailio 4.3.4 and for some reason not getting cfgtraces from debugger. I've picked the default kamailio-advanced.cfg, and did no changes other than setting #!define WITH_DEBUG. Any suggestions on what could be causing this to happen? Thanks! Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

2 1

problem with cfgtrace
by Joao Alves 10 Dec '15

10 Dec '15

Hi, Have just installed kamailio 4.3.4 and for some reason not getting cfgtraces from debugger. I've picked the default kamailio-advanced.cfg, and did no changes other than setting #!define WITH_DEBUG. Any suggestions on what could be causing this to happen? Thanks! Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

1 0

WebRTC no longer supports RTP
by Yuriy Gorlichenko 10 Dec '15

10 Dec '15

https://developers.google.com/web/updates/2015/10/chrome-47-webrtc So at 47 chrome we already have no sound. What kind of proto we must use and how to handle this with rtpengine? Do anyone have same problems with it?

2 2

info: kamailio.org updates - letsencrypt for https and dkim for email
by Daniel-Constantin Mierla 10 Dec '15

10 Dec '15

Hello, during the past few days I made some updates related to the security aspects of kamailio.org services. Two are relevant for the community. 1) First, kamailio.org uses now a TLS certificate signed by letsencrypt.org, a free trusted CA backed up by Mozilla and other internet companies, so browsing via HTTPS should no longer issue any warning of untrusted certificate (previously we used a CACert.org certificate which was not trusted automatically by browsers). Wiki and mailing lists portals use the letsencrypt certificate as well, so is no reason not to browse all kamailio.org and lists.sip-router.org pages only via HTTPS. Perhaps in the near future we will try to enable redirect of HTTP to HTTPS at least for the main page and login pages for wiki, mailing lists and other places that require sensitive data. Now SSLLabs test ranks https://kamailio.org with grade A: * https://www.ssllabs.com/ssltest/analyze.html?d=kamailio.org&latest As a side note, for those that haven't noticed it, for quite some time kamailio.org is available also over IPv6. 2) Second, emails forwarded by kamailio.org and lists.sip-router.org are having now a DKIM signature. Also, there are SPF records in DNS for these domains. Hopefully, those two will help getting the emails to be allowed by various spam filters out there, as their legit origin can be checked. If you check the sources of an email messages and the email server of receiving party is doing DKIM/SPF checks, you should see some headers like next (taken from an email I received to my gmail account from sr-users mailing list): """ Authentication-Results: mx.google.com; spf=pass (google.com: domain of sr-users-bounces(a)lists.sip-router.org designates 193.22.119.66 as permitted sender) smtp.mailfrom=sr-users-bounces(a)lists.sip-router.org; dkim=pass header.i=(a)lists.sip-router.org DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; c=relaxed/relaxed; d=lists.sip-router.org; s=20151206; h=Sender:Content-Type:List-Subscribe:List-Help:List-Post:List-Archive:List-Unsubscribe:List-Id:Reply-To:Subject:MIME-Version:Message-ID:To:From:Date; bh=lGjvCZYcxBHUHaJDnut1j2YTyPsXTnXHzUb0CgcDc1Q=; b=DlD+MKoEqyISB5Ba775t3zg70FC6ouC+tEo7j5zv4dn2Dhm4pWqkQXSfU4Kp1NqW1ZRYFC/mpg/7LEcGW2FlDL9J0FpUg1VjNmN7D1wvtW08hBBw91tsXImu9yf7KZjg/p4IbXu6vznldubrSxweIaV3q/xbrLgaqP5Dsrvs/9A=; """ Kamailio is not enforcing any of those policies on received email messages, so sending to the lists should not be affected. Should anyone discover problems when browsing the web portals or notices issues with emails from our mailing lists, report them to sr-dev mailing list. Also, if anyone has more hints on increasing the security/privacy for the web server and email systems we run for kamailio.org, do not hesitate to provide us suggestions. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

3 2

RTPPROXY unpredictable behavior
by Javohir Maxmarajabov 10 Dec '15

10 Dec '15

I am trying to enable RTPPROXY on debian, but it seems RTPProxy is ignoring my arguments... So, I am launching RTPPROXY using command: rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -p /var/run/rtpproxy.pid -R -a -P -r /tmp/rtppath -S /tmp/rtpspool -u rtpproxy rtpproxy Also tried with: rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -u rtpproxy rtpproxy And when I try to start Kamailio service I am getting error: 0(10562) ERROR: rtpproxy [rtpproxy.c:1681]: send_rtpp_command(): can't send command to a RTP proxy 0(10562) ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy does not respond, disable it 0(10562) WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP proxy 0(10562) WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy has been disabled temporarily p.s. I have checked RTPPROXY, and I am sure that it is launched ps aux | grep rtpproxy returned me: root 2770 0.0 0.0 19392 480 ? Ssl 04:59 0:00 /usr/bin/rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -p /var/run/rtpproxy.pid -R -a -P -r /tmp/rtppath -S /tmp/rtpspool -u rtpproxy rtpproxy root 2775 0.0 0.0 11740 932 pts/1 S+ 04:59 0:00 grep --color=auto rtp Which says everything is ok and rtpproxy is running with arguments that I declared But the problem is, When I check it with: netstat -tulpn | grep rtpproxy I am receiving: udp 0 0 127.0.0.1:22222 0.0.0.0:* 2853/rtpproxy As you can see rtpproxy is listening on port 22222 and completely ignoring arguments I declared :( Where and What am I doing wrong? Sorry for my English and Thanks in advance... -- Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

2 1

having a count of lost response for dialog keep alives
by Jayesh Nambiar 10 Dec '15

10 Dec '15

Hi, I'm using dlg_set_property(ka-src) and dlg_set_property(ka-dst) to keep alive my clients. In case of a network change the client does a Re-Invite with the new contact address and kamailio does keep-alives to the new address properly as the fix reported here: https://github.com/kamailio/kamailio/issues/273 The problem happens when the Re-Invite transaction on the client side is not complete and kamailio happens to send an OPTIONS keep-alive at that same time, the client fails to send back a 200 OK, because the client itself is trying to gain access on the new network. So as a work-around I thought would it be possible to end the dialog on missing at least 3 responses and not killing the dialog if one response is missed. Can it be a configurable parameter in the dialog module or something on those lines. Just a suggestion !! Thanks. - Jayesh

2 2

random incorrect method parsing
by Vasiliy Ganchev 10 Dec '15

10 Dec '15

Hi folk! Have a strange issue, and cannot understand what is wrong. Test scheme UA(sip) -> INVITE -> Kamailio The transport protocol used is TCP. The issue is reproduced randomly, in case of wrong INVITE, Kamailio does not parse Method from R-URI and answer "400 CSeq method does not match request method". The log with debug level 3 for such INVITE: https://gist.github.com/vance-od/c4e1c783adba02d80c58 my first confuse is why in line 2 different number of bytes: "read= 1025 bytes, parsed=1037" the second is in line 12: Method value is empty after this everithing brokes and kamailio answers 400. Just another call: the same setup (same device, kamailio instance etc), same conditions, just another call. Everything is OK, similar part for good call: https://gist.github.com/vance-od/01b5dff2d81f0878cff4 All my attempts to find the diff between to messages goes to nothing, the 2 INVITEs looks similar, the wireshark analyse of TCP level - also had no result (all is similar) The issue happens only with one of our devices and only in case of TCP/TLS (UDP constantly ok, this is why I am assured the problem is in device, but I need proofs) version: kamailio 4.2.5 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, 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. will appreciate any help or idea ,with such a mess! Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/random-incorrect-method-parsing-tp1… Sent from the Users mailing list archive at Nabble.com.

2 5

[Kamailio] Perfomance tests
by Prashant Desai 10 Dec '15

10 Dec '15

Hi, We are using Kamailio 4.2.3 and we have following query related to Performance tests. We see that kamailio wiki page has performance test results plotted for the Open SERV1.2.0. Would like to know is similar test are done for the later versions of kamailio till 4.2.3 ? If Yes please provide the link. Regards, Prashanth

2 1

routing design
by Slava Bendersky 09 Dec '15

09 Dec '15

Hello Everyone, Looking for some design suggestion. I am trying achieve routing based on domain.tld. I thought use drouting, but not sure if it suitable for this task. Example: Kamailio Server 1 | |__________________ PBX Group 1 domain1.tld | |____ PBX Server 1 | |____PBX Server 2 |__________________ PBX Group 2 domain2.tld | |____ PBX Server 1 | |____PBX Server 2 and so on. Slava.

2 1

Kamailio Redundancy Models
by Frank Costeira 08 Dec '15

08 Dec '15

Hi, I'm new to Kamailio and I'm trying to understand the architectural options and levels of redundancy that can be attained with Kamailio. In my particular scenario I would be using Kamailio as proxy server (no user registrations) to handle global prefix routing between 3 major regions; Americas, EMEA, and Aisa-Pacific. The idea would be to create a pool of Kamailio servers in each region. The pool would consist of two HA pairs. The HA pairs would be placed in two separate datacenter, and sessions would be distributed between the datacenter. In the event the Kamailio server failed it would have a local backup in the datacneter. In the event a datacenter failed, there would be an alternate datacenter available in the region. Can the LCR feature/module share a database, so that each kamailio server in the region have the same call routing information? Is there a feature or a way to automate the configuration of several Kamailio servers, when you want them to have the same routing logic/configuration. Is there a module, feature or API that would allow Kamailio to fetch a config, or possibly push the config via an API when the application is initialized. What is the best method for establishing a 1:1 back up with Kamailio? If we use TCP and route-via headers, the SIP session would be pinned through the active proxy server. Is there away to provide stateful failover so that the session state is maintained between primary and the backup server? Any pointers or recommendations would be highly appreciated. Regards, Frank

2 1

secure websocket error in kamailio
by Priyaranjan Nayak 08 Dec '15

08 Dec '15

Hi All, I am running the kamailio for secure websocket (WSS). While running traffic testing the kamailio is not able to take beyond 2300 connection. I am getting below error. 45(12831) ERROR: tls [tls_server.c:243]: tls_fix_connection_unsafe(): tls: ssl bug #1491 workaround: not enough memory for safe operation: 208 45(12831) ERROR: sl [../../forward.h:247]: msg_send(): tcp_send failed 45(12831) ERROR: websocket [ws_handshake.c:113]: ws_send_reply(): sending reply 46(12832) ERROR: <core> [tcp_main.c:3577]: handle_ser_child(): received CON_ERROR for 0x7f03e0b0fa18 (id 651), refcnt 3, flags 0x4018 45(12831) WARNING: <core> [tcp_read.c:1642]: handle_io(): F_TCPCONN connection marked as bad: 0x7f03e0b0fa18 id 651 refcnt 1 44(12830) ERROR: tls [tls_server.c:154]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: 304 44(12830) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading I have seen below data in kamailio at the time of getting error. shmem:fragments = 57 shmem:free_size = 16986416784 shmem:max_used_size = 193649976 shmem:real_used_size = 193452400 shmem:total_size = 17179869184 shmem:used_size = 188516920 Could you please suggest me how can I get more performance from kamailio while using WSS ? Thanks Priyaranjan

2 2

Proposing new logo for Kamailio project
by Daniel-Constantin Mierla 08 Dec '15

08 Dec '15

Hello, upon a process initiated before the summer during the IRC Devel Meeting, when we started to look for refreshing the logo of Kamailio project, we are now trying to finalize the decision and we are proposing a new logo for Kamailio project. The new proposed logo, as well as more details, can be seen at: * http://www.kamailio.org/w/2015/12/proposing-new-logo-for-kamailio-project/ We look forward to the feedback from community about this proposal - you like it or not, is too similar with other logos, etc... This announcement was sent to few of our mailing lists, but please reply to sr-users(a)lists.sip-router.org, being something concerning the community and definitely not something about development. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

2 6

lookup(location) offen dont work with websocket connected endpoints
by Yuriy Gorlichenko 08 Dec '15

08 Dec '15

long time ago looked this problem but resolve it by sql queries directly from confing file to location table. So while remember lookup(location) ofen can not find any contacts if its registers from websocket. it may work 10-15 times and then fails. So debug shows answer that is no connections in location table. Now at production system can not run this issue but at development system want to do it today or tomorrow.

2 1

Installing 4.3 version from git
by Jurijs Ivolga 07 Dec '15

07 Dec '15

Hi all, I'm trying to install 4.3 version from git as described here: http://www.kamailio.org/wiki/install/4.3.x/git As far as I understood latest stable version is 4.3.3, but after installing from git I'm getting 4.3.4 version which is I believe is development version. kamailio -version version: kamailio 4.3.4 (x86_64/linux) 583dc0 Could you please check this? With kind regards, Jurijs

3 2

crash on using http_query when response is 200
by Jayesh Nambiar 07 Dec '15

07 Dec '15

Hi, I'm using http_query to send a post request on a https URL. The weird part is it crashes with segmentation fault when the response from the server is 200. If the response was a 500, it works and continues processing. Here's the function: http_query("https://notify.abcd.com/onnet/call", "$avp(devices)", "Content-Type: application/json" "$var(response)"); xlog("L_INFO", "Response is $var(response)"); And here's the backtrace on crash: 0x00007f62f2bf8e38 in ?? () #1 0x00007f62ea22251b in http_query (_m=0x7f62f2c3a038, _url=0x7f62f2bf3548 "\330\064\277\362b\177", _dst=0x7f62f2bf9250 "\030\215\277\362b\177", _post=0x7f62f2bf35c8 "\200r\277\362b\177", _hdr=0x7f62f2bf91d0 "h\221\277\362b\177") at functions.c:226 #2 0x00007f62ea229e0b in w_http_query_post_hdr (_m=0x7f62f2c3a038, _url=0x7f62f2bf3548 "\330\064\277\362b\177", _post=0x7f62f2bf35c8 "\200r\277\362b\177", _hdr=0x7f62f2bf91d0 "h\221\277\362b\177", _result=0x7f62f2bf9250 "\030\215\277\362b\177") at utils.c:483 #3 0x000000000041f6b5 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2bf8a58, msg=0x7f62f2c3a038) at action.c:1079 #4 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2bf6fc0, msg=0x7f62f2c3a038) at action.c:1549 #5 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2bfb9b0, msg=0x7f62f2c3a038) at action.c:1045 #6 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2bb9388, msg=0x7f62f2c3a038) at action.c:1549 #7 0x000000000041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b8f248, msg=0x7f62f2c3a038) at action.c:678 #8 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b8e268, msg=0x7f62f2c3a038) at action.c:1549 #9 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b91040, msg=0x7f62f2c3a038) at action.c:1045 #10 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b8b7a8, msg=0x7f62f2c3a038) at action.c:1549 #11 0x000000000041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b3a2c0, msg=0x7f62f2c3a038) at action.c:678 #12 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b3a2c0, msg=0x7f62f2c3a038) at action.c:1549 #13 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b47f00, msg=0x7f62f2c3a038) at action.c:1045 #14 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b37e80, msg=0x7f62f2c3a038) at action.c:1549 #15 0x000000000041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b1c9c8, msg=0x7f62f2c3a038) at action.c:678 #16 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b1c9c8, msg=0x7f62f2c3a038) at action.c:1549 #17 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b1cc28, msg=0x7f62f2c3a038) at action.c:1045 #18 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b170a0, msg=0x7f62f2c3a038) at action.c:1549 #19 0x000000000042ca93 in run_top_route (a=0x7f62f2b170a0, msg=0x7f62f2c3a038, c=0x0) at action.c:1635 #20 0x000000000051c115 in receive_msg ( buf=0xab7520 <buf> "INVITE sip:bdbbf600000001b5b919d0@devsip.abcd.com:8321 SIP/2.0\r\nRecord-Route: <sip:203.153.53.145:8300;r2=on;lr=on>\r\nRecord-Route: <sip:203.153.53.145:8321;transport=tcp;r2=on;lr=on>\r\nVia: SIP/2.0/UDP"..., len=1901, rcv_info=0x7ffeb128cc10) at receive.c:196 #21 0x000000000062d427 in udp_rcv_loop () at udp_server.c:495 #22 0x00000000004b28c1 in main_loop () at main.c:1593 #23 0x00000000004b9aa5 in main (argc=5, argv=0x7ffeb128d118) at main.c:2597 Can someone please confirm the crash so that I'll go ahead and report this as an issue on the tracker. Basically when it crashes, kamailio doesn't even execute the xlog function after the http_query call. P.S. abcd.com is a changed domain for confidential reasons. Thanks, - Jayesh

3 2

PDB module long prefixes
by Igor Potjevlesch 07 Dec '15

07 Dec '15

Hi, I had configured my pdb module to accept 5 number prefixes by doing: In pdbt.c: bufsize = slen + 1 + 1 + *4* + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) ... ret = snprintf(p,* 7*, "%d\n", node->carrier); if (ret < 1 || ret > *6*) { And in common.h: #define MAX_PDB_CARRIERID *99999* And i hadn't had any problem until now. I had rows in my .csv like: 0645052050;10122 And it worked. The problem I have now is with prefixes bigger than 40000. I get segmentation fault when doiong a query with pdbt, however the pdbt build works good without any error. At first i thought of a buffer problem but eventhough testing with a .csv of 2 rows with 50000 prefixes will not work. Another curious effect is that for a prefix like 84238 the server changes the answer to 18754. I dont know why... Could anyone help me please? Thanks and regards, Igor.

1 0

Executing a C function from the Kamailio Script
by Mack Hendricks 07 Dec '15

07 Dec '15

We have some logic in a C function that we would like to execute from the Kamailio script. I see there are modules for calling functions in other languages. But, I don’t see anything for C. Is creating a module the only way to leverage this C function? Thanks, -Mack

4 4

SIP Trunking Kamailio directly to external SIP Trunk Provider
by amjadali_bb＠hotmail.com 07 Dec '15

07 Dec '15

Hi, I have got two Asterisk servers and both are communicating with the SIP Trunk provider (External Network) directly - the Asterisk servers are connected with the Kamailio and then Kamailio with the Agent Phones. I need some change in this architecture that I want Kamailio to communicate with the SIP Trunk provider (External Network) directly. Can any one please advise if this is possible and how ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/SIP-Trunking-Kamailio-directly-to-e… Sent from the Users mailing list archive at Nabble.com.

2 1

Announcement: Next Kamailio World - May 18-20, 2016, in Berlin
by Daniel-Constantin Mierla 07 Dec '15

07 Dec '15

Hello, a short message to announces that the date and location for next Kamailio World Conference were decided - May 18-20, 2016, in Berlin, Germany. Kamailio project is celebrating 15 years of development in 2016, therefore we plan to have a special edition. Website of the event and call for presentations will be launched very soon. Meanwhile, if you haven't participated at a past edition, you can check the previous edition website in order to get an idea about the structure and content of the event: - http://www.kamailioworld.com/k03/ Have a great week, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Kamailio radius authentication prolem
by Volkan Oransoy 07 Dec '15

07 Dec '15

Hi all, I try to authenticate my users via mod_radius, but I have problem. FreeRadius server gives this error: Auth: [digest] Cleartext-Password or Digest-HA1 is required for authentication. I think I need to send those attributes from kamailio but I couldn't figure out how to do it. Here is diff of my config with default config. Thanks, /Volkan ===================== diff /etc/kamailio/kamailio.cfg /etc/kamailio/kamailio.cfg.original < #!define WITH_DEBUG 294,297d292 < loadmodule "auth_radius.so" < modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") < loadmodule "avpops.so" < 739,783c734,739 < if (is_method("REGISTER")) < { < avp_print(); < if (!radius_www_authorize("example.com")) { < xlog("SCRIPT: www auth return code: $rc\n"); < switch ($rc) { < case -7: < send_reply("500", "Server Internal Error"); < exit; < case -1: < send_reply("400", "Bad Request"); < exit; < default: < }; < if (defined($avp(digest_challenge)) && < ($avp(digest_challenge) != "")) { < append_to_reply("$avp(digest_challenge)"); < }; < send_reply("401", "Unauthorized"); < exit; < }; < } < < if (from_uri==myself) < { < if (!radius_proxy_authorize("example.com", "$pU")) { # Realm and URI user are taken < switch ($rc) { # from P-Preferred-Identity < case -7: # header field < send_reply("500", "Server Internal Error"); < exit; < case -1: < send_reply("400", "Bad Request"); < exit; < default: < }; < if (defined($avp(digest_challenge)) && < ($avp(digest_challenge) != "")) { < append_to_reply("$avp(digest_challenge)"); < }; < send_reply("407", "Proxy Authentication Required"); < exit; < }; < < } < --- > #!ifdef WITH_IPAUTH > if((!is_method("REGISTER")) && allow_source_address()) { > # source IP allowed > return; > } > #!endif 784a741,753 > 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

2 1

No NOTIFY while PUBLISHing new presence status
by Thierry Luo 07 Dec '15

07 Dec '15

Hi All, I'm now using Kamailio as Presence Server for my IM system. All is fine except that, when one user publishes new presence status (in tuples) to the Presence Server, which does not NOTIFY the subscribers of the status. And the more weird is that, when I restart the subscriber UAs and re-subscribe, the Presence Server send out the NOTIFY messages to the subscribers that just re-subscribed. This is not compliant with RFCs. Anybody please help me out. Attached is the configuration file. Thanks. Thierry Luo

2 2

Table 'kamailio.version' doesn't exist
by Annus Fictus 07 Dec '15

07 Dec '15

Hello List, when I try to create Kamailio Database, I receive this error: *kamdbctl create* MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... ERROR 1146 (42S02) at line 1: Table 'kamailio.version' doesn't exist ERROR: Creating core tables failed at standard! Kamailio version 4.3.4 Any hint? Regards

2 3

UAC dynamic reg_contact_addr()
by Yuriy Gorlichenko 07 Dec '15

07 Dec '15

I have multiple ip addresses at my kamailio. I use uac module for registration to sip providers. I have one provider but want to register form different ip addresses used by my server. When register sends it takes ip address form reg_contact_addr(). But if I want to register from another interface with different ip i need to get to this parametr another ip address. So As I understand it imposibleto do with avp. Does kamailio have some methods to do this?

2 3

why is rtpenengine-delete deleting the whole call?
by Juha Heinanen 05 Dec '15

05 Dec '15

in my test sip proxy parallel forks the invite to two contacts and two rtpengine-offers are made with these kind of call-id and via-branch params: "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0" "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1", then the via-branch .0 callee answers and rtpengine-answer is sent to with this kind of call-id and via-branch params: "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0", after that sip proxy deletes the other branch by sending rtpengine-delete with this kind of params: Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Received command 'delete' from 127.0.0.1:40700 Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Dump for 'delete' from 127.0.0.1:40700: { "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1", "received-from": [ "IP4", "127.0.0.1" ], "from-tag": "3322319973f3909e", "command": "delete" } and rtpengine responds: Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Response dump for 'delete' to 127.0.0.1:40700: { "created": 1448583951, "last signal": 1448583955, "tags": { "3322319973f3909e": { "tag": "3322319973f3909e", "created": 1448583951, "in dialogue with": "0f6759297b8f1711", "medias": [ { "index": 1, "type": "audio", "protocol": "RTP/AVP", "streams": [ { "local port": 53864, "endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10458 }, "advertised endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10458 }, "last packet ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... ": 1448583951, "flags": [ "RTP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } }, { "local port": 53865, "endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10459 }, "advertised endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10459 }, "last packet": 1448583951, "flags": [ "RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } } ], "flags": [ "initialized", "send", "recv" ] } ] }, "0f6759297b8f1711": { "tag": "0f6759297b8f1711", ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0", "created": 1448583951, "in dialogue with": "3322319973f3909e", "medias": [ { "index": 1, "type": "audio", "protocol": "RTP/AVP", "streams": [ { "local port": 53844, "endpoint": { "family": "IPv4", "address": "10.0.2.15", "port": 10144 }, "advertised endpoint": { "family": "IPv4", "address": "10.0.2.15", "port": 10144 }, "last packet": 1448583951, "flags": [ "RTP", "RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... } }, { "local port": 53845, "endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "advertised endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "last packet": 1448583951, "flags": [ "RTCP", "fallback RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } } ], "flags": [ "initialized", "send", "recv", "rtcp-mux" ] } ] } }, "totals": { "RTP": { "packets": 0, "bytes": 0, "errors": 0 }, "RTCP": { "packets": 0, "bytes": 0, "errors": 0 } }, "result": "ok" } which then results in deletion of the whole call, i.e., also the branch that answered. question: why does rtpengine not do what it is asked to do in rtpengine-delete, i,e. delete the offer corresponding to "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1"? -- juha

3 11

Kamailio + Asterisk, same machine, same public IP.
by Douglas Adami 04 Dec '15

04 Dec '15

Hello everyone, I have Kamailio and Asterisk on the same machine with the same public IP. When I try to forward calls to the asterisk (PSTN GW on port 5080), returns the error 401, but does not reach the Asterisk any information. When I switch to an IP out, it works. Any idea? Regards, Doug

2 6

If we print out $fu, $fU & $fd on onsend_route, it prints out incorrect values.
by Jurijs Ivolga 04 Dec '15

04 Dec '15

Hi all, I opened a bug here: https://github.com/kamailio/kamailio/issues/430 But then I was advised to write to mailing list. In nutshell: I'm running Kamailio 4.3.3. When I'm trying to update $ru, $fu & $fd in Kamailio conf: $rU = "1000"; $fu = "sip:something@mydomain.com"; $fd = "newdomain.com"; And later I'm trying to print out $rU, $fu, $fU & $fd in onsend_route: xlog("ALERT: From URI $fu \n"); xlog("ALERT: From domain $fd \n"); xlog("ALERT: From user $fU \n"); xlog("ALERT: Request user $rU \n"); Kamailio 4.3.3 prints out following: ALERT: From URI sip:2000@some_ip ALERT: From domain some_ip ALERT: From user 2000 ALERT: Request user 1000 But I believe it should: ALERT: From URI sip:something@mydomain.com ALERT: From domain mydomain.com ALERT: From user something ALERT: Request user 1000 I'm using default Kamailio config with minor changes, you can check it in attachment. Is it proper behavior? Thank you! With kind regards, Jurijs

2 1

[permissions] allow_source_address_group() always return -1
by Iñaki Baz Castillo 04 Dec '15

04 Dec '15

Hi all, Kamailio 4.3 (Debian Wheezy package) with db_mysql and permissions modules enabled. Once started, this is the cached content of the "address" table: ~# kamctl mi "address_dump" 22 <20, 99.23.86.78, 0> [TEST SERVER] Within the script: ----------------------------- xlog("L_INFO", "request_route() [method:$rm, si:$si, sp:$sp]\n"); $var(group) = allow_source_address_group(); if ($var(group) != -1) { // ALWAYS HERE :( xlog("L_WARN", "invalid source IP [si:$si]\n"); } ----------------------------- Then I send an INVITE from such an allowed IP, and got: INFO: <script>: request_route() [method:INVITE, si:99.23.86.78, sp:7060] WARNING: <script>: invalid source IP [si:99.23.86.78] "version" table contains: table_name table_version ------------------------------------- address 6 trusted 5 I've used code similar to the above in ~30 kamailios (however never in 4.X). Do I miss something? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 2

Re: [SR-Users] About Siremis Login Page Problem
by Daniel-Constantin Mierla 04 Dec '15

04 Dec '15

Hello, do you have the mod_redirect enabled for apache? Can you look at the logs of apache and see if there is any error message? Also, you can look at the logs of siremis in the folder siremis/logs/. Cheers, Daniel On 03/12/15 12:49, Morshed wrote: > > Dear Concern, > > I have installed kamailo and siremis installed properly. But I cannot > login to the siremis, the login page is not loaded successfully. > Please have a look on the attached image and any suggestion from your end? > > > > Regards, > > > > Morshed Alam > > Support Engineer | IP Telephony > > > > Link3 Technologies Ltd. > > Bulu Ocean Tower, 17th Floor > > 40, Kemal Ataturk Avenue, Banani, > > Dhaka 1213, Bangladesh. > > Tel: +88-02-9822288 > > IP Phone : +88-09678997705 > > Fax: +88-02-9821332 > > Web: www.link3.net <http://www.link3.net> > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

LCR module : re load_gws uri errors
by Yasin CANER 04 Dec '15

04 Dec '15

3 7

Generating db-tables .html file
by smititelu 04 Dec '15

04 Dec '15

Hello, I want to add a new table to the [1]. I managed to create a new *kamailio-*.xml* in "lib/srdb1/schema/", run make and see the resulting **-dbschema.xml* in doc/databases/kamailio, along with the other .sql scripts. Here I got stuck; I can't find the scrip that generates the final .html. I tried grepping the code for some keywords found in page source without success. Also, couldn't find the answer reading [2]. Any ideas how the final .html is generated? Thanks, Stefan [1] http://kamailio.org/docs/db-tables/kamailio-db-devel.html [2] http://www.kamailio.org/dokuwiki/doku.php/development:db-schema

2 6

terminate call when dialog when remote contact disappears without BYE
by Thibault Gueslin 03 Dec '15

03 Dec '15

I would like to terminate the dialog when one party disconnects without sending a BYE (no more connctivity, client shutdown) I am using TCP connection. Should I terminate the dialog from event_route[usrloc:contact-expired] ? I have tried also to use SIP OPTION parameter from dialog module modparam("dialog", "ka_timer", 10 ) modparam("dialog", "ka_interval", 30 ) but nothing is sent) thibault

3 13

SQLOPS vs APP_LUA
by Luke Milbourne 03 Dec '15

03 Dec '15

Hi All, I need to run some SQL and then set the destination URI according to the results, if any. I've got this working using APP_LUA, however, I've noticed SQLOPS. Can anyone advise on which method is likely to produce better results. I'm assuming my Lua function is connecting to the DB and then disconnecting each time. If I use SQLOPS, does this mean Kamailio will keep a connection available to the DB? -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

2 2

About Siremis Login Page Problem
by Morshed 03 Dec '15

03 Dec '15

Dear Concern, I have installed kamailo and siremis installed properly. But I cannot login to the siremis, the login page is not loaded successfully. Please have a look on the attached image and any suggestion from your end? Regards, Morshed Alam Support Engineer | IP Telephony Link3 Technologies Ltd. Bulu Ocean Tower, 17th Floor 40, Kemal Ataturk Avenue, Banani, Dhaka 1213, Bangladesh. Tel: +88-02-9822288 IP Phone : +88-09678997705 Fax: +88-02-9821332 Web: <http://www.link3.net> www.link3.net

1 0

using APP_LUA for complex arithmetic operations
by Jayesh Nambiar 03 Dec '15

03 Dec '15

Hi, I understand that kamailio does not allow float-point values in variables and I think kamailio is not a great place to do complex arithemtic operations and so I thought of using app_lua for the same. So can I pass the control to app_lua and come back to config script with some values? For eg. can I call a lua script that does a complex arithemetic operation on my AVP values and based on certain conditions; return back a true or false and then I continue in my existing script. I came across an exmaple http://kb.asipto.com/kamailio:usage:k32-lua-routing but this is more like doing the entire thing inside LUA. What I'm looking at is getting certain things done in lua and continue the script processing. Thanks for any help here. - Jayesh

3 4

Kamailio - Asterisk, problem with SIP trunk mode
by Mickael MONSIEUR 02 Dec '15

02 Dec '15

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

3 6

PKG memory leak in 4.2.6
by Julia Boudniatsky 02 Dec '15

02 Dec '15

Hello, We are running Kamailio 4.2.6 ( db77ac) compiled with memory debugging. The free PKG memory has decreased on ~6.5 M after two weeks. Please see attached file with the log messages at kamailio shutdown. Thank you in advance, Julia.

2 5

Record-route on sequential requests
by Kristian F. Høgh 01 Dec '15

01 Dec '15

Hi list, Record-route on sequential requests doesn't have did/vsf parm. On initial INVITE, I call record_route() and set dialog flag. kamailio adds the following RR Record-Route: <sip:ww.xx.yy.zz;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2> UAC sends a re-INVITE, containing the following route: Route: <sip:178.21.251.54;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2> I call loose_route() followed by record-route(), and kamailio adds the following RR: Record-Route: <sip:ww.xx.yy.zz;lr;ftag=15af612df> I've got a phone which updates the routeset on sequential requests (which I don't think it should) and the did-matching fails. Should kamailio add did/vsf on sequential requests, when I call record_route()? Should I set dlg_match_mode to 1? (I use 0) Is it wrong to call record_route() from sequential requests? Which way is the best to resolve the problem? Regards, Kristian Høgh Uni-tel A/S

2 4

Problem with NATed UAC
by Vladimir Stepic 01 Dec '15

01 Dec '15

Hello everyone, this is my first message. I'm glad to become a part of this big community. I'm new in SER technology. I worked in Asterisk PBX only, until now. For the first time i installed Elastix MT witch have kamailio in front of asterisk, and now i have a problem. I attach a diagram of initial configuration in my lab and pcap file and kamailio.cfg. Regardless diagram, x.x.x.x addresses are public and y.y.y.y are private. When i test connection 1.1 is working, connection 1.2 not working connection 2.2 not working connection 1.3 not working connection 2.3 working Phones are connected(authenticate) to kamailio, but when i look at asterisk cli those phones are unreachable, but can call, but can not be called. On router where android works, Siemens dont, and where pc softphone works other UAC dont work. I think that my nat-traversal configure good, regardless of documentation. Please tell me, if someone has some ideas of solve this problem. Thanks in advance, and sorry for my bad english :) ---------------------------------------------------------------------- *Vladimir Stepić,* Engineer specialist in computer networking System Linux Administrator mob. tel. +381 69/18-68-431 email: e1.e398(a)gmail.com

2 1

kamailio's Makefile system : Enabling CFLAGS/LDFLAGS for udp_server.c
by aft 01 Dec '15

01 Dec '15

Hi, I'm doing some experimentation on kamailio's udp_server.c. Now i need libpcre for that. So i first thought if i enable the regex module, the CFLAGS/LDFLAGS would be available from udp_server.c. But it turns out it doesn't build(doesn't *link* to be more precise). Kamailio's build system is pretty unfamiliar for me. Although i absolutely hate autotools based build system(I prefer Scons), but i'm pretty familiar with these. But Kamailio's build system is pretty much cryptic for me(Is it based on CMake?). There is a page on devel wiki for kamailio's build system. it turns out i can do stuff like : make cfg prefix=/opt/kamailio \ SCTP=1 \ CC_EXTRA_OPTS=-I/usr/gnu/include \ group_include="standard postgres presence" \ include_modules="snmpstats lcr tls" So there might be a LD_EXTRA_OPTS or stuff like that. Which than would be defined (I guess) globally so the core c files(like udp_server.c) would be able to use that. One way i guess would be to sharpen my egrep and find that variable name. But it all feels a bit hacky. As i have already included regex module, there should be a way to reuse the CFLAGS/LDFLAGS for that module for core files like udp_server.c. Cheers Arif

2 1

Re: [SR-Users] Nathelper module
by Daniel-Constantin Mierla 01 Dec '15

01 Dec '15

Hello, On 30/11/15 14:52, Marino Mileti wrote: > > I would like to use nathelper module in order to "ping" all registered > client and "delete" (from locations table) clients that are not > reachable (maybe there's some network issue, or some other problem) > > > > I've seen that it's possible to have this feature for natted contact > using nathelper (but i haven't tried this functionaliy) > > > > Is there a way to remove records from location table without wait that > its expire is 0? > > see: http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.p.… Also, look at the parameter of usrloc module for handling contacts that have the tcp connection cut. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Help Required
by anakweze charles 30 Nov '15

30 Nov '15

Hi All. Does anyone know where to find training guide for Kamailio HA? I want to have 2 Kamailio in HA with four (4) Freeswitch servers behind them. I will also like to customer cgrate to manage the accounting side in a multi-tenant environment. I am just researching for this on my own and I will appreciate a guide to realize this objective. Thanks

1 0

SNMPstats dialog 0
by Igor Potjevlesch 30 Nov '15

30 Nov '15

Hello, I have rebooted my kamailio machine and now the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 And also the status is: KAMAILIO-MIB::kamailioDialogUsageState.0 = INTEGER: idle(0) How can I change it to active(1)? I have dialog.so loaded as well as modparam("dialog", "dlg_flag", 4) and dlg_manage(). Thank you. Regards,

3 5

Reverted Error Message
by Sam Ware 30 Nov '15

30 Nov '15

Daniel, I got the following error message (currently running version 4.3): Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR: dispatcher [dispatcher.c:517]: mod_init(): call load dispatching DST ID_AVP set but no size for hash table Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing modul e dispatcher (/usr/local/lib64/kamailio/modules/dispatcher.so) I noticed in the following DEV GIT (Version 3.1) message that the error message was changed to a message that more clearly defined the error and correction for it. But the my current message would indicate that it got changed back to the older error message. http://lists.sip-router.org/pipermail/sr-dev/2010-November/009799.html Just thought I would point it out because the newer message from the GIT message seems clearer to me. -- *Sam D Ware* CIO *Vera Networks LLC* Email: sam.ware(a)veranetworks.com

2 1

Seremis Manual
by Omnes Leroy 30 Nov '15

30 Nov '15

Hi, Is there a manual I can get so Seremis. I am a new user. Thanks, Omnes

2 1

Kamailio geolocation module
by Mititelu Stefan 28 Nov '15

28 Nov '15

Hi everyone, Is there such a thing like "geolocation" module in Kamailio? (or something related to it i.e. [1] [2]). Thanks, Stefan Mititelu [1] https://tools.ietf.org/html/rfc6442 [2] https://tools.ietf.org/html/rfc3693#page-6

2 4

crash on using the $expires variable
by Jayesh Nambiar 28 Nov '15

28 Nov '15

Hi, I am trying to extract out the expires value from the incoming REGISTER request. It might be present in the Contact header or as a separate Expires header and hence I planned to use the $expires(max) PV to get this. But kamailio just crashes as soon as I use the variable. Here's when it crashes: if(is_method("REGISTER")) { xlog("L_INFO", "Expires is $expires(max)"); if(!save("location", "0x01")) { t_reply("500", "Error while saving AOR"); } } It crashes at the xlog point. Here's the backtrace for reference: Reading symbols from /usr/local/kamailio/sbin/kamailio...done. [New LWP 16609] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid -f /usr/local/kamail'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f34b658b311 in pv_get_expires (msg=0x7f34bd879e58, param=0x7f34bd7e9220, res=0x7ffcf67af530) at pv_core.c:3060 3060 if(c->expires && c->expires->body.len) { (gdb) backtrace #0 0x00007f34b658b311 in pv_get_expires (msg=0x7f34bd879e58, param=0x7f34bd7e9220, res=0x7ffcf67af530) at pv_core.c:3060 #1 0x00000000004fb6e1 in pv_get_spec_value (msg=0x7f34bd879e58, sp=0x7f34bd7e9208, value=0x7ffcf67af530) at pvapi.c:1304 #2 0x00000000004fbb46 in pv_printf (msg=0x7f34bd879e58, list=0x7f34bd80ef38, buf=0x7f34bd8735a0 "Expires is ", len=0x7ffcf67af618) at pvapi.c:1363 #3 0x00007f34b67ea745 in xl_print_log (msg=0x7f34bd879e58, list=0x7f34bd80ef38, buf=0x7f34bd8735a0 "Expires is ", len=0x7ffcf67af618) at xl_lib.c:37 #4 0x00007f34b67eb0f9 in xlog_helper (msg=0x7f34bd879e58, xm=0x7f34bd808648, level=2, line=0, facility=-1) at xlog.c:215 #5 0x00007f34b67ec160 in xlog_2_helper (msg=0x7f34bd879e58, lev=0x7f34bd7e9130 "", frm=0x7f34bd808648 "84\177", mode=0, facility=-1) at xlog.c:284 #6 0x00007f34b67ec19c in xlog_2 (msg=0x7f34bd879e58, lev=0x7f34bd7e9130 "", frm=0x7f34bd808648 "84\177") at xlog.c:292 #7 0x000000000041e23b in do_action (h=0x7ffcf67b0b40, a=0x7f34bd817cd0, msg=0x7f34bd879e58) at action.c:1065 #8 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd813e18, msg=0x7f34bd879e58) at action.c:1548 #9 0x000000000041ab16 in do_action (h=0x7ffcf67b0b40, a=0x7f34bd7e8738, msg=0x7f34bd879e58) at action.c:677 #10 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd7e8738, msg=0x7f34bd879e58) at action.c:1548 #11 0x000000000041e094 in do_action (h=0x7ffcf67b0b40, a=0x7f34bd7e89c8, msg=0x7f34bd879e58) at action.c:1044 #12 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd7e19d8, msg=0x7f34bd879e58) at action.c:1548 #13 0x000000000042b04d in run_top_route (a=0x7f34bd7e19d8, msg=0x7f34bd879e58, c=0x0) at action.c:1634 #14 0x000000000050f3d4 in receive_msg ( buf=0xa80c00 <buf> "REGISTER sip:devsip.tlkn.com SIP/2.0\r\nVia: SIP/2.0/UDP 203.153.53.145:8300;branch=z9hG4bK119a.e23ed10b856f38b41c93d1d259d8869f.0;i=b1\r\nVia: SIP/2.0/TCP 172.16.16.32:52358;rport=52359;received=203.153."..., len=712, rcv_info=0x7ffcf67b0e60) at receive.c:196 #15 0x0000000000613b9e in udp_rcv_loop () at udp_server.c:495 #16 0x00000000004aa119 in main_loop () at main.c:1577 #17 0x00000000004b0e1e in main (argc=5, argv=0x7ffcf67b1318) at main.c:2553 Thanks, - Jayesh

3 4

restore_uri_reply() called twice in 200OK to BYE
by Efelin Novak 27 Nov '15

27 Nov '15

Hi Folks, I have a problem with To and From headers in 200 OK to BYE request. Their URI is doubled when leaving Kamailio. Incoming 200 OK: From : <sip:00421111@IP_ADDRESS>;tag=as017a2986 To : <sip:00421222@IP_ADDRESS0>;tag=K7ryDgH2g7gFm Outgoing 200 OK: From : <sip:+421111@IP_ADDRESS><sip:+421111@IP_ADDRESS>;tag=as017a2986 To : <sip:+421222@IP_ADDRESS><sip:+421222@IP_ADDRESS>;tag=K7ryDgH2g7gFm I'm using uac module to replace these headers in auto and dialog mode: modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) append_fromtag is also set: modparam("rr", "append_fromtag", 1) When I run kamailio in debug mode I can see the function restore_uri_reply() is called twice: . . . 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 234, <received> = <IP_ADDRESS>; state=6 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK0bb8c5b2>; state=6 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <5060>; state=16 0(3133) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 0(3133) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=62 0(3133) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=teHD72X67aQUm 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 0(3133) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [60]; uri=[sip:421222@IP_ADDRESS:5060] 0(3133) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:421222@IP_ADDRESS:5060>] 0(3133) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <103> <BYE> 0(3133) DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 10806 label 0 branch 0 0(3133) DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0x220071b4)! 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x220071b4, callback type 2, id 0 entered 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=as46bccb83 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421222@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421222@IP_ADDRESS>> 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x220071b4, callback type 2, id 0 entered 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421222@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421222@IP_ADDRESS>> . . . I think the problem is in registering a function restore_uris_reply() in callbacks in functions (both in replace.c of uac module) * void rr_checker(struct sip_msg *msg, str *r_param, void *cb_param) * static void replace_callback(struct dlg_cell *dlg, int type, struct dlg_cb_params *_params) Both of these callbacks are registered for BYE and ACK requests (INVITE request is using different callback), therefore running restore_uri_reply() twice. For testing purposes after I have removed the callback in rr_checker and recompile the uac module, To and From headers were OK. In my configuration I'm calling uac_replace_to in RELAY route: route[RELAY] { if ( !has_totag() ) { dlg_manage(); uac_replace_to("$ru"); } if (!t_relay()) { sl_reply_error(); } } So I think there is a bug in uac module, when one function is registered twice in callbacks for NON-INVITE requests. Or is there a problem with my configuration script? Can anyone point me to the direction how to solve this issue? Thank you Efelin Kamailio version: kamailio 4.4.0-dev6 (i386/linux) a66e22-dirty

2 2

Kamailio as UAC
by Alexandru Covalschi 27 Nov '15

27 Nov '15

Hello everyone! Subj is an ethernal theme, I know, but odds are so that I just need to do that. I've configured UAC auth to successfully register and my route[PSTN] looks like that # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } if !ds_is_from_list() return; # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; if (strempty($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) + ":" + $sel(cfg_get.pstn.gw_port); } remove_hf("To"); insert_hf("To: <sip:$rU@sipprovider.com>\r\n", "Call-ID"); uac_replace_from("","sip:user@sipprovider.com"); route(RELAY); exit; On INVITE's I get 407 PROXY-AUTH, which are transfered to backend FS. If I try to put if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "Remote asked for authentication"); uac_auth(); } to MANAGE_FAILURE or MANAGE_REPLY route Kamailio can't start. Is that even possible? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 4

how to log sql query
by John Smolka 27 Nov '15

27 Nov '15

Hi guys, I use kamailio with mysql. Some queries are huge and when I need to debug something it takes a time, so the debbuging would be easier. Is there any way how could I log all sql queries into a log file ? Thanks, John

2 1

Media address family
by Sergey Okhapkin 27 Nov '15

27 Nov '15

What is the best way to find out address family (IPV4 or IPV6) of media in SDP? My goal is to reject INVITEs which have different address families in headers and SDP, i.e. IPv4 INVITE with IPv6 media address in SDP and vice versa.

3 2

Correct start up of RTPProxy and Kamailio
by Michael Nielsen 27 Nov '15

27 Nov '15

I've installed RTPProxy and Kamailio and would like to have them start up correct on boot. First RTPProxy and then Kamailio. How to accomplish this?

2 1

2 Kamailio servers on same domain
by Michael Nielsen 26 Nov '15

26 Nov '15

I've installed Kamailio on 2 different servers. They share the same PostgreSQL database for users etc. I've setup a single domain, sip.my-domain.com, with latency responds to the 2 servers. So a user will always land on the server with the fastest responds in a respective area of the world. However, it seems as if 2 users are registered on different Kamailio servers - even though they share the same database - they cannot call each other? Have I misunderstood something? I've enabled WITH_NAT.

2 5

kamailio very slow or stuck during period from time to time
by Thibault Gueslin 26 Nov '15

26 Nov '15

I am facing an issue with kamailio 4.3.3 that is very similar to http://lists.sip-router.org/pipermail/sr-users/2015-May/088195.html Had someone found the solution or faced similar issue ? Sometimes kamailio is receiving the message (check with wireshark, logs in tcp) but the message is not handled nor relayed. After a certain amount of time everything works fine. I will do what as has been advised : enabling pike & check with gdb when it appears. thibault

3 13

issue using sipcapture to pg database
by david escartin 26 Nov '15

26 Nov '15

hello how are you? i have a problem using sipcapture module i'm writing in postgres database some sip messages but the msg field is stored ina strange format, \x5349502f322e30203438372052657175657374205465726d696e61746 5640d0a5669613a205349502f322e302f5544502037392e3137302e36342e3136383a353038303b6272616e63683d7a39684734624b30583665375a70394e446533513b72656365697665643d37392e3137302e36342e3136383b72706f72743d353038300d0a5265636f72642d526f7574653a 203c7369703a3139302e3136362e3233322e3139333a393236323b7472616e73706f72743d7564703b6c723e2c3c7369703a37392e3137302e36382e3138363b6c723b667461673d394448397648374435444653653b6469643d3639612e616437393e0d0a43616c6c2d49443a2033323666333 037632d373138642d346164392d393234362d3333336432323466613439380d0a46726f6d3a2022313931373838393437363922203c7369703a31393137383839343736394037392e3137302e36342e3136383e3b7461673d394448397648374435444653650d0a546f3a203c7369703a313830 39323935373838374037392e3137302e36382e3138363a353036303b757365723d70686f6e653e3b7461673d736263303830367974646c746f6c302d43432d33300d0a435365713a20383338353333383820494e564954450d0a5761726e696e673a203339392031302e31382e352e363420225 353313230303031463135364c3934345b30303030305d2043616e63656c207265636569766564206f6e20696e697469616c20696e76697465220d0a436f6e74656e742d4c656e6774683a20300d0a0d0a do you know how to configure it to see it in ascii? thanks a lot and regards david escartin

2 1

[CFP] reminder! FOSDEM RTC dev-room talks: deadline Friday
by Daniel Pocock 25 Nov '15

25 Nov '15

Reminder: speaker's deadline this Friday, 27 November at 23:59 UTC We have already received several really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the call for papers: http://danielpocock.com/fosdem-2016-free-rtc-dev-room-and-lounge We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

1 0

Kamailio v4.3.4 Released
by Daniel-Constantin Mierla 25 Nov '15

25 Nov '15

Hello, Kamailio SIP Server v4.3.4 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.4. For more details about version 4.3.4 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/11/kamailio-v4-3-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

TLS not enough memory issue with git master
by Anthony Messina 25 Nov '15

25 Nov '15

I have noticed the following issue which began with builds somewhere between git master commits bff0a08 and 6173ef7. I did not see this issue with my previous builds and haven't been able to pin down the problem, which is why I haven't formally filed a bug. Any help or guidance is appreciated, because this has crippled my use of Kamailio. Only a restart enables it to work again until the issue recurs. ERROR: tls [tls_server.c:189]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: 8870536 ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading I currently build against and run openssl-1.0.1k-12.fc22.x86_64. I have a very small operation and the only change on the operational side is that all 5 of my mobile UACs (yes, that's all) have switched from CSipSimple/Android to Zoiper/Android, which doesn't yet have support for client-side certificates so verify_certificate and require_certificate are off for both the server and client config. The server is started with: /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -m 64 -M 8 I have tried modifying the shared mem to 128 but the issue still occurs. Even right now, I am seeing the error when only one UAC has established a TLS connection: # kamcmd tls.list { id: 572 timeout: 3475 src_ip: 10.77.79.156 src_port: 58688 dst_ip: 10.77.79.3 dst_port: 5061 cipher: ECDHE-RSA-RC4-SHA SSLv3 Kx=ECDH Au=RSA Enc=RC4(128) Mac=SHA1 ct_wq_size: 0 enc_rd_buf: 0 flags: 2 state: established } # kamailio.cfg enable_tls=yes loadmodule "tls.so" modparam("tls", "connection_timeout", 60) #modparam("tls", "tls_log", 1) #modparam("tls", "tls_debug", 1) #modparam("tls", "low_mem_threshold1", -1) #modparam("tls", "low_mem_threshold2", 0) modparam("tls", "session_cache", 1) # tls.cfg [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/example.org.key.pem certificate = /etc/kamailio/example.org.crt.pem server_name = example.org cipher_list = ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-SHA384:ECDHE-RSA- AES128-GCM-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-RSA-RC4-SHA:ECDHE-RSA-AES256- SHA:DHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES256-SHA256:DHE-RSA-AES128-GCM- SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA:RC4- SHA:AES256-GCM-SHA384:AES256-SHA256:CAMELLIA256-SHA:ECDHE-RSA-AES128- SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:CAMELLIA128- SHA:!aNULL:!eNULL:!EXPORT:!DES:!3DES:!MD5:!PSK [client:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/example.org.key.pem certificate = /etc/kamailio/example.org.crt.pem cipher_list = ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-SHA384:ECDHE-RSA- AES128-GCM-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-RSA-RC4-SHA:ECDHE-RSA-AES256- SHA:DHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES256-SHA256:DHE-RSA-AES128-GCM- SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA:RC4- SHA:AES256-GCM-SHA384:AES256-SHA256:CAMELLIA256-SHA:ECDHE-RSA-AES128- SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:CAMELLIA128- SHA:!aNULL:!eNULL:!EXPORT:!DES:!3DES:!MD5:!PSK Thanks. -Anthony -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

3 16

Kamailio/RTPProxy incorrect port 0
by Igor Potjevlesch 25 Nov '15

25 Nov '15

Hello, I can't figure out why Kamailio reports hundreds of calls with the following error: "/usr/local/sbin/kamailio[2997]: ERROR: rtpproxy [rtpproxy.c:2727]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" It seems that each time this error occurs, we got a one-way audio call. I already check and increase the limit for opening files that the process can open with ulimit -n 500000. But, anyway, if I look to /proc/$PID/fd, I never see more than 1000 files open. I also check the socket with netstat: same; I can see the error while there's only less than 800 sockets open. Is someone know what can be the trigger of this? Regards, Igor.

5 10

Kamailio:Contact validation during INVITE
by Prashant Desai 25 Nov '15

25 Nov '15

Hi, We observe that the following kamailio API is trying to find out user registration by comparing source IP and port with stored contact, if it fails, then it compares based on incoming SIP msg contact uri. is_registered(struct sip_msg* _m, udomain_t* _d){ getContactP(_m,_d); } pcontact_t * getContactP(struct sip_msg* _m, udomain_t* _d) {); //Getting pcontact from source IP and port if (ul.get_pcontact_by_src(_d, &received_host, _m->rcv.src_port, _m->rcv.proto, &c) == 1) { LM_DBG("No entry in usrloc for %.*s:%i (Proto %i) found!\n", received_host.len, received_host.s, _m->rcv.src_port, _m->rcv.proto); } if (c == NULL) { //Getting pcontact from contact uri. if (ul.get_pcontact(_d, &ct->uri, &c) == 0) { } } We have use case where INVITE message is from same source IP and port but having unregistered contact.With this scenario,is_registered() will return success. We would like to know reason behind comparing the source IP and port to fetch the pcontact? Regards, Prashant

3 5

Releasing Kamailio v4.3.4
by Daniel-Constantin Mierla 25 Nov '15

25 Nov '15

Hello, I am considering to release Kamailio v4.3.4 next week on Wed, Nov 25 or the following day. As usual, announce any issues that you are aware and not reported to sr-dev or bug tracker. Also, if some patch was not backported and you think it has to be, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 1

Build error on Solaris 11 with Solaris Studio
by Spencer Thomason 25 Nov '15

25 Nov '15

Hello, I realize its probably uncharted waters but I was hoping someone could point me in the right direction… I attempting to build Kamailio on a UltraSPARC T2 box running Solaris 11.3 with Sun cc and I’m not getting very far: $ CC=cc gmake all CC (cc) [kamailio] timer_proc.o cc: Warning: -xarch=v9 is deprecated, use -m64 to create 64-bit programs "timer_funcs.h", line 111: warning: argument mismatch "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "cfg/../atomic/atomic_unknown.h", line 59: #warning: no native memory barrier implementations, falling back to slow lock based workarround "cfg/../lock_alloc.h", line 72: warning: argument mismatch "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "ut.h", line 236: warning: argument mismatch "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 240: warning: argument mismatch Any help would be greatly appreciated! The contents of config.mak are below: # this file is autogenerated by make cfg MAIN_NAME= kamailio CFG_NAME= kamailio SCR_NAME= sip-router FLAVOUR= kamailio INSTALL_FLAVOUR= kamailio SRC_NAME= sip-router RELEASE= 4.3.3 OS= solaris ARCH= sparc64 C_DEFS= -DNAME='"kamailio"' -DVERSION='"4.3.3"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"cc 5.1"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=4003003 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DDBG_QM_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DUSE_PTHREAD_MUTEX -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H DEFS_RM= PROFILE= CC= cc LD= cc MKDEP= cc -xM1 MKTAGS= ctags LDFLAGS= -xO5 C_INCLUDES= CC_MKDEP_OPTS= MOD_LDFLAGS= -G -xO5 LIB_LDFLAGS= -G -xO5 UTILS_LDFLAGS= -xO5 LIB_SONAME= "-h " LD_RPATH= -"-R " LIB_SUFFIX= .so LIB_PREFIX= lib LIBS= -lfast -ldl -lresolv -L/usr/local/lib -lxnet -lnsl -lrt LEX= flex YACC= yacc YACC_FLAGS= -d -b cfg PREFIX= /usr/local LOCALBASE= /usr/local TAR= gtar INSTALL= ginstall INSTALL_CFG= ginstall -m 644 INSTALL_BIN= ginstall -m 755 INSTALL_MODULES= ginstall -m 755 INSTALL_DOC= ginstall -m 644 INSTALL_MAN= ginstall -m 644 INSTALL_LIB= ginstall -m 755 INSTALL_TOUCH= touch INSTALL_SHARE= ginstall -m 644 Q= 1 CC_EXTRA_OPTS= CPU= CFLAGS_RM= CFLAGS= -m64 -g -xO5 -fast -native -xarch=v9 -xCC -xc99 $(CC_EXTRA_OPTS) MOD_CFLAGS= -xcode=pic32 $(CFLAGS) LIB_CFLAGS= -xcode=pic32 $(CFLAGS) UTILS_CFLAGS= $(CFLAGS) BASEDIR= $(DESTDIR) basedir= $(BASEDIR) DESTDIR= LIBDIR= lib64 RUNBASEDIR= $(DESTDIR) runbasedir= $(RUNBASEDIR) PREFIX= $(LOCALBASE) prefix= $(PREFIX) cfg_prefix= $(basedir)$(prefix) cfg_dir= etc/$(MAIN_NAME)/ bin_prefix= $(basedir)$(prefix) bin_dir= sbin/ modules_prefix= $(basedir)$(prefix) modules_dir= $(LIBDIR)/$(MAIN_NAME)/ doc_prefix= $(basedir)$(prefix) doc_dir= doc/$(MAIN_NAME)/ man_prefix= $(basedir)$(prefix) man_dir= man/ ut_prefix= $(basedir)$(prefix) ut_dir= share_prefix= $(basedir)$(prefix) share_dir= share/$(MAIN_NAME)/ lib_prefix= $(basedir)$(prefix) lib_dir= $(LIBDIR)/$(MAIN_NAME)/ data_prefix= $(basedir)$(prefix) data_dir= $(MAIN_NAME)/ cfg_target= $(prefix)/$(cfg_dir) lib_target= $(prefix)/$(lib_dir) data_target= $(prefix)/$(data_dir) run_prefix= $(runbasedir) run_dir= var/run/$(MAIN_NAME)/ run_target= $(run_prefix)/$(run_dir) override makefile_defs:=1 C_DEFS:=$(filter-out $(DEFS_RM) $(extra_defs),$(C_DEFS)) $(extra_defs) CFLAGS:=$(filter-out $(CFLAGS_RM) $(CC_EXTRA_OPTS), $(CFLAGS)) $(CC_EXTRA_OPTS) Thanks! Spencer

1 0

Grab users password from WWW-Auth header
by Alexandru Covalschi 24 Nov '15

24 Nov '15

Hello! My problem is I need to do users authentication through API. So I need to replace if (!www_authenticate("$td", "subscriber")) { www_challenge("$td", "1"); } With if (!my_auth_script()) { www_challenge("$td", "1"); } The main problem is - how can I grab or compare users password? I know nonce, which I understand is MD5 salt. Can I, for example, grab users password from API, then grab the MD5 string and the nonce user sent me, calculate MD5 on base of API password and nonce - and then compare MD5 strings sent by user and calculated? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

3 11

Issue with SNMPstats
by Igor Potjevlesch 24 Nov '15

24 Nov '15

Hello, I have configured the SNMPstats module fine in my machine because I get the SNMP information with the new KAMAILIO MIBS but I don't understand why the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 However I can see the transactions: KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32: 54797 Should I configure an specific parametr of the SNMPstats module? Regards,

3 6

MESSAGE is not successfull in Kamailio IMS
by Thierry Luo 24 Nov '15

24 Nov '15

Hi All, I installed kamailio on Ubuntu as IMS pCSCF/iCSCF/sCSCF. REGISTER is fine, but when I tried to send a MESSAGE (SMS) from one user to another, it was always not successful. I captured IP packets on the server, and it seemed that the error was caused on pCSCF. User1 and User2 are both behind their NAT network. Suppose User1's private IP is Private_IP1, and User1's public IP (on router) is Public_IP1, and User2's private and public IPs are Private_IP2 and Public_IP2: User1 pCSCF iCSCF sCSCF user2 MESSAGE ----------------> MESSAGE ---------------------> MESSAGE -------------------> MESSAGE <----------------------------------------------------- MESSAGE ---------------------------------------------------------------------------- ----> Unfortunately, from the captured data on the server, in the last step, the MESSAGE was sent to the Private_IP2 of User2. It should be sent to the Public_IP2. Of course, the datagram with the destination of Private_IP2 cannot reach User2 successfully. Please help analyze why the pCSCF sent the MESSAGE packet to User2's private IP instead of public IP. The configuration file was attached for reference. BR. Thierry Luo

2 6

Simple User Agent change
by Steven Maddox 24 Nov '15

24 Nov '15

Hi, Sorry to ask something probably very simple but I'm difficulty finding the right kinds of resources to learn from which are succinct enough to not overload my head. We have a PBX which we don't control (and hate) which expects only certain makes/models phones to register (it checks for user agents). I have a phone which unfortunately doesn't support tweaking its user agent, so I'd like to pass it through an extremely SIP proxy to have it's user agent tweaked and all other data to/from the PBX transparently let through. I've managed to knock up something (that isn't a proxy) using Asterisk where I made a trunk and a peer... but obviously since this isn't a proxy... things like BLF and transferring calls will never work. If anyone can point me in the direction of a really simple set up to achieve this, please let me know :) -- Steven Maddox

2 2

kazoo module documentation
by Vik Killa 24 Nov '15

24 Nov '15

Hello, I'm testing kazoo module for use with RabbitMQ I noticed there is a required parameter that is not listed in the documentation: http://kamailio.org/docs/modules/devel/modules/kazoo modparam("kazoo", "pua_mode", 0) The above parameter is required but not documented. Can someone please update the documentation? Thank you. /V

2 1

Problem with resuming suspended transactions.
by Dmytro Bogovych 23 Nov '15

23 Nov '15

Greetings. We build specialized softphone for iOS. Push notifications are used to notify user and wake up softphone app on missed calls and messages. We used Daniel's example config script as base for own configuration file. It has the same parts and does not differ too much. I talk about this example: http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… Everything was ok for few monthes - with kamailio 4.1.2. Few weeks ago we upgraded to 4.3.3. Now paused calls cannot be resumed properly. Softphone receives 500 Server error (No error 0/SL) in response This is problem part of script: route[RELAYRESUMED] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if ($var(rr_target) != "") { xlog("L_INFO","Relay to $var(rr_ip):$var(rr_port)\n"); $ru = "sip:" + $rU + "@" + $var(rr_ip) + ":" + $var(rr_port) + ";transport=" + $var(rr_proto); xlog("L_INFO","Current request URI is $ru\n"); if (!t_relay()) xlog("L_INFO","Cannot relay PUSHED invite."); sl_reply_error(); } else { xlog("L_WARN","Target proto is not set\n"); sl_reply_error(); } exit; } It produces log ----- Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: resuming trasaction [23927:345343979] 1985987629 (23927:345343979) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Relay to 95.133.110.17:11791 Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Current request URI is sip:1985987629@95.133.110.17:11791;transport=tls Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (0/SL) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Checking FAILED messages for 1985987629: 0 ----- sl_reply_error() sends 500 with No error (0/SL) description. Later 180 & 200 are sent. But client softphone terminates transaction already (as it handles 500 response code) . There is no "Cannot relay PUSHED invite." message in the log however. I will be happy to heary any idea what can be cause... Thank you!

2 3

Kamailio-SEMS Integration
by Prashant Desai 23 Nov '15

23 Nov '15

Hi, We are working on kamailio-4.2.3 and we require B2BUA functionality to be achieved.From kamailio official website we got to know,we can integrate kamailio with SEMS to achieve B2BUA functionality. We have few queries regarding this integration. 1.SEMS-1.6.0(latest version available) is been used,is this version suits kamailio-4.2.3? 2.SEMS-1.6.0 does not support unix based communication with kamailio-4.2.3,as SEMS-1.6.0 doesn't have unix socket interface plugin. Since we need unix socket based communication with SEMS.can we port the unix socket interface plugin to SEMS-1.6.0?If no,please suggest us better approaches. Regards, Prashant.

2 2

Kamailio, rtpproxy, websocket and 488 error
by Alfredo Pastor 23 Nov '15

23 Nov '15

Hello, I was doing some tests with Elastix MT that use version 4.1.6 of Kamailio and by default uses rtpproxy, I had the problem of error 488 and unable to make calls from websockets (Incompatible SDP). I could see that Peter Dunkley in a video explains that the problem is solved by rewriting the SDP, but he uses the mediaproxy not rtpproxy. What are your recommendations? I hope you can help me. Chreers, Alfredo.

1 0

rtpengine via-branch=extra question
by Juha Heinanen 22 Nov '15

22 Nov '15

rtpengine readme tells: via-branch=... - Include the “branch” value of one of the “Via” headers in the request to the RTP proxy. Possible values are: “1” - use the first “Via” header; “2” - use the second “Via” header; “auto” - use the first “Via” header if this is a request, or the second one if this is a reply; “extra” - don't take the value from a header, but instead use the value of the “extra_id_pv” variable. This can be used to create one media session per branch on the RTP proxy. When sending a subsequent “delete” command to the RTP proxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “rtpengine_delete”, or stop all sessions for a call when not passing one of those two flags there. if via-branch=extra is used in offers of branched, should it also be used in answers and deletes? -- juha

2 6

Functionality of rtpproxy
by kai.ohnacker＠cbc.de 20 Nov '15

20 Nov '15

Hello, I have installed a kamailio server with rtpproxy. The server is located between 2 networks and bridge these networks. My question: Does handle rtpproxy only 1 call at the same time? Because my configuration works, but only for one call from the first to the second network. The second call is not working. But if I make internal or external call, I can do more than one call. Do I need the multiple rtpproxy usage configuration? I thought not, because I don't use different IPs for the second call... Cheers, Kai

2 1

403 Forbidden for INVITE in kamailio for ims call
by Priyaranjan Nayak 20 Nov '15

20 Nov '15

Hi All, I have installed kamailio and using as a i-cscf,s-cscf and p-cscf server. I have taken FHoSS(OpenImscore) for hss server Now I am able to do register and unregister. When I am sending INVITE for call, the calling-party and called-party were registerd but I am getting "403 Forbidden - Dialog not found on S-CSCF or Terminating user not suitable for unregistered services". Could you please tell me what is the root cause of above error message ? Best regards, Priyaranjan

1 0

RFC: rtpproxy and a=nortpproxy:yes
by Daniel-Constantin Mierla 19 Nov '15

19 Nov '15

Hello, I am looking to get some feedback on some issues I noticed more and more lately. Apparently some SIP devices (media servers, phones, ...) are keeping the "a=nortpproxy:yes" line in SDP when replying to an INVITE that contains such line. [Alice] ------> [Kamailio+RTPProxy] ------> [Bob] The 200ok response from Bob has "a=nortpproxy:yes" in SDP. By default, that line in SDP makes the rtpproxy not to engage itself anymore in rtp relaying, and as a result things like no audio or one way audio happens. Anyone else encountering such situations? If yes, what are the devices with such behaviour? So far I noticed with some FreeSwitch and Snom -- none of them I can control, so there might be a specific configuration of those devices, not something by default there. The solution is to set: modparam("rtpproxy", "nortpproxy_str", "") and use flag 'r' for rtpproxy_manage() if the IP in SDP is not a private address. I already updated the default config for master to use flag 'r' if the SDP media IP is not private, wondering if nortpproxy_str should be set to empty in kamailio.cfg (or made empty as default in config). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

rtpproxy database structure for kamailio 4.2.3
by Jan Hazenberg 19 Nov '15

19 Nov '15

Hi all, I'm trying to load the rtpproxy url's from database but get a error when starting kamailio (4.2.3): ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'set_name' in 'field list' Looks like the field set_name is missing in the database structure, but i can't find it in the docs. If i add the field kamailio starts, but what is the field set_name used for. Seems it needs to be a int value: ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name 'test' - must be integer Thanks, Jan

2 2

Fw: Kamailio with Siremis
by Ryan Holbein 19 Nov '15

19 Nov '15

________________________________ From: Ryan Holbein <rtholbein(a)hotmail.com> Sent: Wednesday, November 18, 2015 8:07 AM To: miconda(a)gmail.com Subject: Re: [SR-Users] Kamailio with Siremis Hello, This is what I have setup so far I know it is not much but looking for help this is my very first Proxy. If you look at my Attachments I am trying to do IP auth. I have 2 Asterisk Boxs with 192.168.89.2 and .3.. The Proxy has 2 nics the outside and the 192.168.89.0/24 ... How do I setup the LCR rule if you look at the LCR Attachment? Also is my trusted permissions setup correct(Those IPs you see are correct)? If you are willing to help Thank you in Adv. ________________________________ From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Thursday, November 12, 2015 12:05 PM To: Ryan Holbein; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis If you have a single Asterisk box, then you can simply forward using kamailio.cfg only. Look at route[PSTN] in default configuration file for an example. If you have more Asterisk boxes, then depends on what you need/want to do: - for load balancing look at dispatcher module - it has a complete sample config in the docs - for least cost routing, then lcr module is indeed an option Cheers, Daniel On 12/11/15 17:56, Ryan Holbein wrote: Now I assume in LCR I will add the Asterisk boxs so it can route? Does that sound correct? Anyone able to provide an example? Thank you ________________________________ From: sr-users <sr-users-bounces(a)lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org> on behalf of Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com> Sent: Thursday, November 12, 2015 3:01 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis Hello, look at permissions module. Default kamailio can do IP auth by enabling WITH_AUTH and WITH_IPAUTH and adding the trusted IP in database table 'address' with group id 1. Siremis has a panel for managing the records in address table. Cheers, Daniel On 11/11/15 14:10, Ryan Holbein wrote: Hello, I have setup with my phone provider an IP endpoint to do IP Auth. How would I go about setting that up in Kamailio or even better can some tell me how to do that in Siremis? _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda<http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

2 2

ndb_mongodb, db_mongodb and replicasets
by Kelvin Chua 19 Nov '15

19 Nov '15

anybody ever tried using a mongodb url with replicasets and parameters to work with kamailio? Kelvin Chua

2 11

Content-Type
by Safdar Khan 19 Nov '15

19 Nov '15

How to check content-type of ongoing various type of media streams in kamailio?

2 1

siptrace segfault - attack?
by Marian Piater 18 Nov '15

18 Nov '15

Hello, we had segfault today. Kamailio has been running for few months without problems. Nov 18 10:51:20 sbc kernel: [11326028.926502] kamailio[14452]: segfault at 30 ip 00007f20f7f3838a sp 00007ffef3ab7b10 error 4 in siptrace.so[7f20f7f23000+27000] GDB: Reading symbols from kamailio...done. [New LWP 14452] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kam'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 1041 sto.totag = get_to(msg)->tag_value; (gdb) bt #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 #1 0x0000000000457ca9 in do_action (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1053 #2 0x0000000000463cb9 in run_actions (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1548 #3 0x00000000004643bf in run_top_route (a=0x7f2100b8a928, msg=0x7f2100f96750, c=0x0) at action.c:1634 #4 0x0000000000573ea7 in receive_msg ( buf=0x9c9400 <buf> "OPTIONS sip:100@XXX.XXX.XX.XX SIP/2.0\r\nv: SIP/2.0/UDP 69.64.39.119:5060;branch=z9hG4bK-82135822;rport\r\nContent-Length: 0\r\nf: \"MisterX\"<sip:100@1.1.1.1>;tag=61326665333131663133633401333733313630343335"..., len=357, rcv_info=0x7ffef3ab85b0) at receive.c:196 #5 0x0000000000493ff5 in udp_rcv_loop () at udp_server.c:495 #6 0x000000000051fdd7 in main_loop () at main.c:1573 #7 0x0000000000525b6b in main (argc=13, argv=0x7ffef3ab8998) at main.c:2533 (gdb) quit Our IP is hidden, but there is 69.64.39.119, which is foreign address and I think, it was attack. Unfortunatelly I don't have SIP packet details, but you can see 1.1.1.1 or strange tag or content-length=0 in received message. I looked into siptrace.c and there is function sip_trace_prepare where is get_from(msg) checked, but no get_to(msg). This function is run from main sip_trace function. So I think, we need also check get_to(msg) function. I just disabled siptrace module, but we need it. Thank you. Marian

1 1

Stop msrp relay from kamailio to freeswitch
by Safdar Khan 18 Nov '15

18 Nov '15

Hi folks I have integrated kamailio and freeswitch. The basic register and authentication is happening through kamailio and other media services through freeswitch like file sharing (using msrp). My question is how can I stop file sharing through msrp via freeswitch since everything going from kamailio to freeswitch to client. I just wanted to share file only through kamailio. I am attaching my cfg file below... #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_PRESENCE #!define WITH_TLS #!define WITH_MSRP #!define WITH_XHTTP #!define WITH_MSILO #!define WITH_NAT #!substdef "!MY_IP_ADDR!192.168.5.99!g" #!substdef "!MY_DOMAIN!<SERVER_IP>!g" #!substdef "!MY_WS_PORT!9080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!define WITH_WEBSOCKETS # Kamailio (OpenSER) SIP Server v4.3 - 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="sip.mydomain.com" /* 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:127.0.0.1:5060 /* 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 listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 ####### 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 #!ifdef WITH_FREESWITCH freeswitch.bindip = "192.168.5.99" desc "FreeSWITCH IP Address" freeswitch.bindport = "5090" desc "FreeSWITCH Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/kamailio-4.3/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" #!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 #!ifdef WITH_MSRP loadmodule "msrp.so" #!endif loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "cfgutils.so" #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" ##loadmodule "nathelper.so" #!endif #!ifdef WITH_MSILO loadmodule "msilo.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", 1) # 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- #modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.5.99:5060") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "NULL") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-4.3/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/usr/local/kamailio-4.3/etc/kamailio/privkey.pem") modparam("tls", "certificate", "/usr/local/kamailio-4.3/etc/kamailio/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/kamailio-4.3/etc/kamailio/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) modparam("tls", "crl", "/usr/local/kamailio-4.3/etc/kamailio/my_crl.pem") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;") #!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 #!ifdef WITH_MSRP # ----- msrp params ----- modparam("msrp", "sipmsg", 1) modparam("msrp", "cmap_size", 8) modparam("msrp", "use_path_addr", "msrp://192.168.5.99:5060") #!endif modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=8") modparam("sqlops","sqlcon","ca=>mysql://kamailio:kamailiorw@localhost/kamailio") #!ifdef WITH_AUTH # ----- auth params ----- #modparam("auth", "nonce_count", 1) #modparam("auth", "qop", "") #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif #!ifdef WITH_MSILO # ------ msilo params ------ modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@192.68.5.98") modparam("msilo","contact_hdr","Contact: <sip:registrar@192.168.5.99>\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","") modparam("msilo", "check_time", 200000) modparam("msilo", "expire_time", 259200) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # 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); #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE_MESSAGE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); ## #!ifdef WITH_MSILO ### m_dump(); ### #!endif exit; } } # USER location service route[LOCATION] { xlog("going inside LOCATION\n"); #!ifdef WITH_SPEEDIAL # 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 #!ifdef WITH_MSILO xlog("going inside ifdefMSILO\n"); #if(is_method("MESSAGE")) # { xlog("going inside MSILO\n"); route(MSILO); # exit; #} #!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; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif #!ifdef WITH_MSILO if(is_method("MESSAGE")){ return; } #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) 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) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); route(RELAY); exit; } #!endif # manage outgoing branches # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_MSILO #manage failure message failure_route[MANAGE_FAILURE_MESSAGE] { if (!method=="MESSAGE") { exit; } log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { t_reply("202", "Accepted"); }else{ t_reply("503", "Service Unavailable"); } } #!endif # 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 } #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); route(FSVBOX); } } #!endif route[8] { sql_query("ca", "insert into statistics (time_stamp,random,shm_used_size,shm_real_used_size, shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts, $RANDOM,$stat(used_size),$stat(real_used_size),$stat(max_used_size), $stat(free_size),$stat(location-users),$stat(location-contacts))", "ra"); } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0", "$msrp(method)")) { if(auth_get_www_authenticate("myrealm", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } msrp_cmap_save(); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } if (msrp_cmap_lookup()) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_relay(); } else { msrp_reply("481", "Session-does-not-exist"); } } else { msrp_reply("501", "Request-method-not-understood"); } } #--------------websockets starts here------------- #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; exit; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif #--------------websockets ends here------------- #---------------msilo starts here------------------ route[MSILO]{ #!ifdef WITH_MSILO if(!is_method("MESSAGE")) return; ##..........added...... if(lookup("location")) { if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; }; if(!lookup("location")) { $var(rc)=$rc; if (! t_newtran()) { sl_reply_error(); exit; }; switch ($var(rc)) { case -1: case -3: if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; case -2: t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; } }; t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; #!endif return; } #------------------msilo ends here----------------

1 0

Kamailo crashes with auth_radius module
by Volkan Oransoy 18 Nov '15

18 Nov '15

Hi, I try to enable auth_radius module with my kamailo box. But it crashes on the first REGISTER request. I have enabled it with this two lines. loadmodule "auth_radius.so" modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") This is the authentication part of my script. if (is_method("REGISTER")) { if (!radius_www_authorize("$td")) { switch ($rc) { case -7: send_reply("500", "Server Internal Error"); exit; case -1: send_reply("400", "Bad Request"); exit; default: }; if (defined($avp(digest_challenge)) && ($avp(digest_challenge) != "")) { append_to_reply("$avp(digest_challenge)"); }; send_reply("401", "Unauthorized"); exit; }; } And in the /etc/radiusclient/dictionary file, I included sip and kamailio dictionaries. $INCLUDE dictionary.sip $INCLUDE dictionary.kamailio Service succesfully starts but it crashes on the first REGISTER request. Nov 18 11:30:27 sip kernel: [159645.360105] traps: kamailio[6941] general protection ip:7f89b1660920 sp:7ffc8d3bac78 error:0 in libc-2.19.so [7f89b15cb000+19f000] Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:728]: handle_sigs(): child process 6941 exited by a signal 11 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:731]: handle_sigs(): core was generated Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6959]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6951]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6936]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6948]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6944]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6937]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6953]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6945]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6938]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6955]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6943]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6942]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6957]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6939]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6940]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Thanks,

2 3

Authorization selects
by Alexandru Covalschi 17 Nov '15

17 Nov '15

Hello everyone! I need to extract values from authentication header, but 408. @authorization["string"] 409. @authorization["string"].username 410. @authorization["string"].username.user 411. @authorization["string"].username.domain 412. @authorization["string"].realm 413. @authorization["string"].nonce 414. @authorization["string"].uri 415. @authorization["string"].cnonce 416. @authorization["string"].nc 417. @authorization["string"].response 418. @authorization["string"].opaque 419. @authorization["string"].algorithm 420. @authorization["string"].qop these values doesn't seem to work for me. Used https://github.com/kamailio/kamailio/blob/master/doc/select_list/select_cor… doc. $var(z) = @authorization["string"].realm; xlog("Realm $var(z)"); Using that simple check I get nothing in $var(z) version: kamailio 4.3.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, 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 with gcc 4.9.2 Can anyone check that please? Or maybe syntax is wrong and I need newer selects doc? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 3

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

4 8

General question about mohqueue
by David Villasmil Govea 16 Nov '15

16 Nov '15

Hello list, I've been given a task to setup a predictive dialer with for 8000+ agents taking accepted calls in a queue. I've done this for a much smaller setup in freeSWITCH but i recently saw that kamailio has a mohqueue (which i already setup as a PoC) that might work for this. My question is whether you think it's a good idea to go with this on kamailio or is freeSWITCH better suited for this kind of application? The setup would be to generate outgoing calls based on a list (predictive dialer) and send answered calls to a queue to be answered by an agent. Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Kamailio 4.3 and Debian 8.2 systemctl
by Grant Bagdasarian 16 Nov '15

16 Nov '15

Hello, I've just installed Kamailio 4.3 on Debian 8.2, but starting kamailio does not work since Debian requires systemctl to be setup. I've tried the below config but it didn't work (https://github.com/kamailio/kamailio/issues/294#issuecomment-138590939) usr/local/src/kamailio# systemctl status kamailio.service â kamailio.service - Kamailio SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled) Active: failed (Result: resources) systemd[1]: [/etc/systemd/system/kamailio.service:14] Not an absolute path, ignoring: $PIDFILE systemd[1]: Failed to load environment files: No such file or directory systemd[1]: kamailio.service failed to run 'start' task: No such file or directory systemd[1]: Failed to start Kamailio SIP Server. systemd[1]: Unit kamailio.service entered failed state. [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=syslog.target network.target [Service] Type=forking Environment='PIDFILE=/var/run/kamailio/kamailio.pid' Environment='CFGFILE=/etc/kamailio/kamailio.cfg' Environment='SHM_MEMORY=64' Environment='PKG_MEMORY=8' Environment='USER=kamailio' Environment='GROUP=kamailio' EnvironmentFile=/etc/default/kamailio PIDFile=$PIDFILE # ExecStart requires a full absolute path ExecStart=/usr/sbin/kamailio -P $PIDFILE -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP ExecStopPost=/bin/rm -f $PIDFILE Restart=on-abort [Install] WantedBy=multi-user.target Does anyone have a working kamailio.service file? Regards, Grant

4 3

TLS module not initialized in 4.3.3, worked in 4.1.5
by Sebastian Damm 16 Nov '15

16 Nov '15

Hello, we just updated one kamailio server from 4.1.5 to 4.3.3, and although the config file is correct and kamailio starts up, it doesn't initialize TLS and says " tls support enabled, but no tls engine available (forgot to load the tls module?)" In the log I see: Old shutdown (last lines): Nov 13 11:44:38 lasola /usr/sbin/kamailio[15890]: DEBUG: <core> [mem/shm_mem.c:235]: shm_mem_destroy(): destroying the shared memory lock Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:271]: tcp_read_data(): error reading: Connection reset by peer (104) Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:1296]: tcp_read_req(): ERROR: tcp_read_req: error reading New startup (first lines): Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [daemonize.c:583]: set_core_dump(): core dump limits set to 18446744073709551615 Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2475]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2476]: main(): disabling tls... Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [async_task.c:88]: async_task_init(): start initializing asynk task framework Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): xmlrpc Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules//sl.so] Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): sl In our config file we have the following lines for TLS (pretty late, after all other module loading and after most parameters): #!ifdef ENABLETLS loadmodule "tls.so" modparam("tls", "private_key", "/etc/ssl/private/my.kamailio-key.pem") modparam("tls", "certificate", "/etc/ssl/certs/my.kamailio.crt") #!ifdef TLS_CA_CHAIN # Maybe we want to use a chain to the CA modparam("tls", "ca_list", "/etc/ssl/certs/my.ca-bundle.crt") #!endif enable_tls=1 listen=tls:1.2.3.4:5061 #!endif After starting up, kamailio listens on port 5060, but not on port 5061. In version 4.1.1, this config worked without a problem. Has anybody seen this before? the tls module is there and available, it doesn't say anything about "cannot load module", and it is only a warning message. I'm also wondering, why this message is the first after starting the server. From config I would expect that sl, tm and all the other modules should be initialized before tls. Best Regards, Sebastian

2 9

Detecting problems of kamailio under very high load
by aft 16 Nov '15

16 Nov '15

We have multiple kamailio servers with 4 cpu cores and 16G RAM. We use kamailio+rtpproxy as a outbound sip proxy. Usually there are many thousands of concurrent sip sessions of occurring there. Periodically sometimes it just stops serving request and spits out 5** replies. At that point we usually reload the kamailio daemon and things return back to normal. As we have a lot of servers, manually doing this is a pain in the neck. So We installed a homer/sipcapture on a separate server . And from there, we periodically scan for 500 msgs for a given time interval. If there are any, we then proceed to reload kamailio. I know its hell of an inefficient system to monitor. So I'm wondering, if there can be any more rational way to detect problems pre-emptively. any fifo/mi command which we could run to find out if the kamailio instance is hung up? Cheers Arif

3 2

Loading mohqueue stops kamailio from starting up
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello all, I installed kamailio on debian from kamailio's apt repository. Configured it with kamdbctl, started it up properly, no issues. I configured mysql also, worked properly. Next i loaded up "mohqueue" and everything went to hell :) This is my config: loadmodule "mohqueue.so" # ----- moh-queue ----- modparam ("mohqueue", "db_url", "mysql://user:password@localhost/kamailio") modparam ("mohqueue", "db_qtable", "mohqueue") modparam ("mohqueue", "db_ctable", "mohqcalls") modparam ("mohqueue", "mohdir", "/var/kamailio/MOH") and my db table for mohqueue is: +----+------------+-------------------------+--------------------+---------+-------+ | id | name | uri | mohdir | mohfile | debug | +----+------------+-------------------------+--------------------+---------+-------+ | 1 | test_queue | sip:1234@172.16.163.130 | /var/kamailio/MOH/ | test | 9 | +----+------------+-------------------------+--------------------+---------+-------+ when starting, i get: ... 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(5069) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(5069) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(5069) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Anyone can help? Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 2

(no subject)
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello guys, I'm trying to startup 4.3 and i'm getting the following: 0(4420) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 192 bytes for row values at 0x7fe6bf6b5a48 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [1] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [ sip:1234@172.16.163.130] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [/var/kamailio/MOH/] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test.8] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test_queue] 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [9] 0(4420) ERROR: mohqueue [mohq_db.c:504]: update_mohq_lst(): Queue,Field (test_queue,mohdir): Unable to find MOH files (/var/kamailio/MOH//test.8.*)! 0(4420) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules/sl.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_tm> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_newtran> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_tcp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_udp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(4420) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(4420) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(4420) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Any help is appreciated, Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Kamailio queues
by David Villasmil Govea 13 Nov '15

13 Nov '15

Hello Guys, I'm new to this list. I'm trying to setup a queue with 4.3.3 (x86_64/linux) and I see a couple of things i'm wondering about: - kamdbctl created the table "mohqueues" and the associated value in the "version" tables. Whereas kamailio uses "mohqueue" (no extra "s" at the end). - after fixing that, it finds the table, but when searching for the moh file, which i set as "/var/kamailio/MOH/test.8" on the mohqueue table, but kamailio can't find it: 0(61540) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(61540) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules ls -l /var/kamailio/MOH/test.8 -rw-rw-rw- 1 root root 166038 Jun 10 2009 /var/kamailio/MOH/test.8 Can anyone lend a hand? Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Re: [SR-Users] Flaky & unstable RTP connections
by Infinicalls Infinicalls 13 Nov '15

13 Nov '15

Hi, I have started RTP with # ./rtpproxy -F -l localhost -s udp:*:7890 and when I issue #netstat -pln | grep rtpp I get, udp 0 0 0.0.0.0:7890 0.0.0.0:* 8759/rtpproxy I am using another similar instance of RTP also. Still my connection seems to be flaky. This doesn't occur always. I am not using dispatcher here. Simple rtpproxy with udp:rtp1 and udp:rtp2 in kamailio.cfg file. Is there anyway to tell if RTP is healthy other than making calls and finding out? Can somebody test it from your end? Or is this problem client specific? I am using microlite and sipper on windows. phone.infinicalls.com u:test;p:test123 u:trial;p:trial123 5060 regards Ganesh Kumar

1 0

Support-for-AKAv2-MD5-authentication-in-kamailio
by Anil Kumar 13 Nov '15

13 Nov '15

Hi, I want to enable AKAv2-MD5 type authentication using a plain kamailio. Is there any way I can achieve it ? Please do needful. Thanks Anil

2 1

Kamailio with Siremis
by Ryan Holbein 12 Nov '15

12 Nov '15

Hello, I have setup with my phone provider an IP endpoint to do IP Auth. How would I go about setting that up in Kamailio or even better can some tell me how to do that in Siremis?

2 5

issue with uac_req_send to websocket peer
by Vasiliy Ganchev 12 Nov '15

12 Nov '15

Hi folk! Have a scenario: UA-A -> REGISTER -> kamailio kamailio make auth, and save(location), after this, before exit, I want to send to UA-A a custom NOTIFY with specific event (this should be done without SUBSCRIBE/PUBLISH/NOTIFY logic) .... $uac_req(method) = "NOTIFY"; $uac_req(ruri) = $sel(contact.uri); $uac_req(furi) = "sip:service@" + $rd; $uac_req(turi) = $sel(contact.uri); $uac_req(hdrs) = ......... here custom header is added xlog("L_ERR", "debuginfo, NOTIFY to ru $uac_req(ruri), fu $uac_req(furi); uac_req_send(); ............ when UA-A is connected via UDP/TCP/TLS - works OK when UA-A connected via WS/WSS", have errors: Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: <script>: debuginfo, NOTIFY to ru sip:251@10.100.1.218:47114;transport=ws, fu sip:service@10.100.1.85, Nov 10 14:59:29 /usr/sbin/kamailio[6260]: WARNING: <core> [msg_translator.c:2799]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message I see no problem with ws connect (calls to/from this ws peer works OK), immediately after registration: root@local:/home/admin# kamctl fifo ws.dump 2: wss:127.0.0.1:40239 -> wss:127.0.0.1:5061 (state: OPEN, last used 5s ago, sub-protocol: sip) The difference in UDP/TCP/TLS from WS/WSS - that first connects directly to Kamailio to port 5060/5061, but second connects to custom ports, where listen special daemon and proxy sip to kamailio 5060/5061 via localhost I tried also to add in script $uac_req(sock) = "tls:127.0.0.1:5061"; but the result is the same version: kamailio 4.2.5 (i386/linux) any suggestion? cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/issue-with-uac-req-send-to-websocke… Sent from the Users mailing list archive at Nabble.com.

2 2

Unable to find dialog
by Ryan Brindley 12 Nov '15

12 Nov '15

Hey community, Our current switch is being overloaded and when it does we're getting these 2 messages spammed in our logs: WARNING: dialog [dlg_handlers.c:1127]: dlg_onroute(): unable to find dialog for ACK with route param '35d.d2f1' [3411:7981] ERROR: tm [t_reply.c:1341]: t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 503, received: 100 We're obviously working on not having the switch get so overloaded, but I'm also curious to understand why it can't find the dialogs. Is it a memory issue? Networking/dropping packets issue? I also imagine it's not as simple as a single overall answer either, but nevertheless I figured I'd reach out to the community and see if anyone has an understanding of what is happening and would be willing to take a few mins to explain. Thanks in advance, Ryan Brindley Software Development Officer Stratics Networks, Inc. 1.866.635.6918 x108

2 1

Active-Passive Auto deployment of Kamailio
by Gholamreza Sabery 11 Nov '15

11 Nov '15

Hello. Recently I started a very simple open source project for auto deploying an Active-Passive Kamailio cluster using Corosync and Pacemaker. For auto deployment I used Ansible. I thought it can be useful for some users. Link of project's repository: https://github.com/ghrst/Kamailio-HA

2 2

Kamailio can not read to_tag at BYE request
by Yuriy Gorlichenko 11 Nov '15

11 Nov '15

Hello. I have some one provider that sends me BYE request with to_tag that kamailio can not parse IP my.pro.vi.der.5060 > my.ser.vi.ce.5060: UDP, length 406 E.......x...S.0X_..L......8.BYE sip:0987654321@my.ser.vi.ce SIP/2.0 Via: SIP/2.0/UDP my.pro.vi.der:5060;branch=z9hG4bKsj7bv820c0s18k8a15q1sd000ag33.1 Call-ID: 00137fb21f65f19a3094c38b30d80f5c CSeq: 1479 BYE From: <sip:1234567890@my.pro.vi.der>;tag=SDi9r0899-5ci36vojn0 To: <sip:0987654321@my.pro.vi.der>;tag=as6635bbe1 Reason: Q.850;cause=16;text="Normal call clearing" Max-Forwards: 69 Content-Length: 0 So I guess that problem in provider (may be ";" with another code) But may be we have some instruments to get tag (without parsing strings in a config file or something like that )

2 1

Removing transport=TLS from the contact header
by Rizwan Khan 10 Nov '15

10 Nov '15

I want to remove the transport=TLS from the contact header. How can I do that inside Kamailio? The flow is an under: Client --------> Kamailio -------------> FS TLS UDP Client <-------- Kamailio <------------- FS TLS UDP FS gets a request on udp but it complains about TLS profile as the contact header has the transport=TLS. Since its receing traffic on UDP, if I only get rid of the value from the Contact Header, it would be great. Thanks, Rizwan Khan

4 15

Header field Contact not changed on kamailio 4.2
by Зуев Михаил 09 Nov '15

09 Nov '15

Hi all, I need to modify the host part of a contact header. I'm trying something like: route { ..... if (remove_hf("Contact")){ if (insert_hf("Contact: rn", "Contact")) xlog("Contact modified"); } if (!t_relay("YYY.YYY.YYY.YYY","5060")) { sl_reply_error(); } ...... } When i look SIP DUMP on YYY.YYY.YYY.YYY side i see older Contact field. I try subst_hf with regexp, subst and simply remove Contact, all this does not work. What i am doing wrong? -- С уважением Зуев Михаил

3 2

Examples
by Ryan Holbein 09 Nov '15

09 Nov '15

Hello, I have everything setup and installed... Does anyone have a good link or could tell me the steps of how to connect my trunks to phone provider and then another one would be how to route the calls to the internal PBX system. Thank you

3 4

kamailio enabling reply to options
by amar Smart Telecom 09 Nov '15

09 Nov '15

Dear All, Could you pls advise: How can we enable 'options' in Kamailio. Other server is sending "options" but our Kamailio is not responding. Thanks.

2 1

Re: [SR-Users] Branch & Forking
by Daniel-Constantin Mierla 09 Nov '15

09 Nov '15

Hello, the aggregation of replies for a parallel fork is done automatically and negative responses are absorbed. Can you send a ngrep trace of a call where you found that the negative response of a branch is not absorbed in case a branch answered? Cheers, Daniel On 05/11/15 16:41, Marino Mileti wrote: > I'm playing with branch and i'm using this simple route (copied from kamailio website) > > route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com"); > append_branch("sip:c@example.com"); > t_relay(); > break; > } > > A B & C rings at the same time. > > If B or C aren't registered, and the caller send a CANCEL, it receives also a 404 messsage because for example B wasn't online. > > Is it possible that Kamailio "absorb" this 404? I don't want that my caller receive it :) > > Marino Maria Mileti > marino.mileti(a)alice.it > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

Outgoing SIP messages
by Varghese Paul 09 Nov '15

09 Nov '15

Hi all, Is it possible to log the outgoing SIP messages relayed from kamailio ? Regards Varghese Paul

3 3

Question about siremis
by Bruno Salzano 09 Nov '15

09 Nov '15

Hello, i hope not to be OT. However I noticed that latest release is 4.2.0 and that this version is compatible with kamailio 4.2.x. Is it compatible also with 4.3.x? Another question is related to Siremis development: this great webapplication is developed using PHPOpenBiz Framework. From several days the PHPOpenBiz website is down. So will Siremis development continue on PHPOpenBiz? Is there some release schedule? Best Regards Bruno

2 1

Preprocessor define puzzle
by Alex Balashov 09 Nov '15

09 Nov '15

Hi, I've got: # define WITH_VENDOR_ROUTING_LNPOIP at the top of my config. This is intended to mean that this constant is commented out and disabled ('# define' vs. '#!define'), and tests correctly in a given request route. Later: route[STAGE_3] { ... #!ifdef WITH_VENDOR_ROUTING_LNPOIP xlog("L_INFO", "LNPOIP ENABLED STAGE 3\n"); #!else xlog("L_INFO", "LNPOIP NOT ENABLED STAGE 3\n"); #!endif route(OUTBOUND_VENDOR_SELECT); } This tests correctly: Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP NOT ENABLED STAGE 3 However, in route[OUTBOUND_VENDOR_SELECT]: #!ifdef WITH_VENDOR_ROUTING_LNPOIP xlog("L_INFO", "LNPOIP ENABLED\n"); #!else xlog("L_INFO", "LNPOIP NOT ENABLED\n"); #!endif This prints: Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP ENABLED Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP NOT ENABLED ... which I must admit is quite puzzling. Looks like it's both defined and not defined. Running 4.3:16be09c here. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Kamailio 4.2.3 generates core dump when using sql_xquery in event_route htable:mod-init
by Jan Hazenberg 06 Nov '15

06 Nov '15

Daniel, I'm currently running a kamailio 4.2.3 with htable enabled and i'm trying to load data into a hash table using the "event_route[htable:mod-init]". When i try to execute a sql_xquery kamailio generates a core dump on startup. Example event_route: event_route[htable:mod-init] { sql_xquery("kamailio", "SELECT id,number FROM blacklist_number", "blacklist_number"); } Log output when starting kamailio: 0(16905) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(16905) INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available 0(16905) INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 0(16905) INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled 0(16905) INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: //etc/kamailio/permissions.allow 0(16905) INFO: permissions [permissions.c:608]: mod_init(): default allow file (//etc/kamailio/permissions.allow) not found => empty rule set 0(16905) INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: //etc/kamailio/permissions.deny 0(16905) INFO: permissions [permissions.c:617]: mod_init(): default deny file (//etc/kamailio/permissions.deny) not found => empty rule set 0(16905) INFO: pike [pike.c:115]: pike_init(): PIKE - initializing 0(16905) INFO: pike [ip_tree.c:88]: init_lock_set(): probing 256 set size 0(16905) INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 124928 0(16905) INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 249856 0(16905) INFO: rtpproxy [rtpproxy.c:1593]: rtpp_test(): rtp proxy <udp:10.1.8.20:22000> found, support for it enabled 0(16905) NOTICE: <script>: Startup event route! Segmentation fault (core dumped) Could this be a bug? Or is there a better way to load data from a custom table into a hashtable on startup. Thanks, Jan Hazenberg

2 2

running kamilio on mac
by Stefan Mancunianet 06 Nov '15

06 Nov '15

Hi, I’ve installed kamilio on mac, but when trying to run it i get the following error message: [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Can sctp flag be disabled ? It’s not required for my needs, and from what I found on the web, it’s very tricky to use sctp on Mac. Thanks

3 4

Webpage Display
by Ryan Holbein 06 Nov '15

06 Nov '15

Hello, I have installed: CentOS 6 Kamailio<http://www.kamailio.org/> v4.2 Siremis v4.2 I went thought the siremis install just fine but when you get to the last page and click launch siremis the very next screen is displayed wrong. Unable to login and display any images.

3 6

segmentation fault when mongodb & tls enabled. ( kamailio 4.3.2)
by Jijo 06 Nov '15

06 Nov '15

I'm observing a segmentation fault when mongodb & tls enabled. It doesn't happen when one of the module is disabled. OS: centos 6.7 kamailio 4.3.2 mongo-c-driver version 1.1.10. It happens immediately after i register a subscriber. The location table is updated with the new data and in parallel core is generated as well. Loaded symbols for /lib64/libnss_dns-2.12.so Core was generated by `/usr/local/kamailio_proxy/sbin/kamailio -f /usr/local/kamailio_proxy/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 0x00007fd3adf65e69 in atomic_cmpxchg_int (var=0x56444e3055445030, old=0, new_v=1) at ../../atomic/atomic_x86.h:233 233 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) Missing separate debuginfos, use: debuginfo-install cyrus-sasl-lib-2.1.23-15.el6_6.2.x86_64 cyrus-sasl-plain-2.1.23-15.el6_6.2.x86_64 db4-4.7.25-19.el6_6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-42.el6.x86_64 libcom_err-1.41.12-22.el6.x86_64 libcurl-7.19.7-46.el6.x86_64 libidn-1.18-2.el6.x86_64 libselinux-2.0.94-5.8.el6.x86_64 libssh2-1.4.2-1.el6_6.1.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-20.el6.x86_64 mysql-libs-5.1.73-5.el6_6.x86_64 nspr-4.10.8-1.el6_6.x86_64 nss-3.18.0-5.3.el6_6.x86_64 nss-util-3.18.0-1.el6_6.x86_64 openldap-2.4.40-5.el6.x86_64 openssl-1.0.1e-42.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x00007fd3adf65e69 in atomic_cmpxchg_int (var=0x56444e3055445030, old=0, new_v=1) at ../../atomic/atomic_x86.h:233 #1 0x00007fd3adf65eb6 in futex_get (lock=0x56444e3055445030) at ../../futexlock.h:99 #2 0x00007fd3adf66468 in prv_lock_tree_branch (b=192 '\300') at ip_tree.c:47 #3 0x00007fd3adf664e5 in lock_tree_branch (b=192 '\300') at ip_tree.c:66 #4 0x00007fd3adf6dda6 in swap_routine (ticks=123206032, param=0x0) at pike_funcs.c:293 #5 0x00000000005fef8e in compat_old_handler (ti=1971296512, tl=0x7fd32ccea430, data=0x7fd32ccea430) at timer.c:996 #6 0x00000000005ff94c in slow_timer_main () at timer.c:1130 #7 0x00000000004a8676 in main_loop () at main.c:1628 #8 0x00000000004ae38e in main (argc=11, argv=0x7fff1d97ec28) at main.c:2533 (gdb) On Clean up after the core, mongo db does segmentation fault as well Loaded symbols for /lib64/libnss_files-2.12.so Reading symbols from /lib64/libnss_dns-2.12.so...Reading symbols from /usr/lib/debug/lib64/libnss_dns-2.12.so.debug...done. done. Loaded symbols for /lib64/libnss_dns-2.12.so Core was generated by `/usr/local/kamailio_proxy/sbin/kamailio -f /usr/local/kamailio_proxy/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 __pthread_mutex_destroy (mutex=0x7fd32cce7be8) at pthread_mutex_destroy.c:28 28 if ((mutex->__data.__kind & PTHREAD_MUTEX_ROBUST_NORMAL_NP) == 0 Missing separate debuginfos, use: debuginfo-install cyrus-sasl-lib-2.1.23-15.el6_6.2.x86_64 cyrus-sasl-plain-2.1.23-15.el6_6.2.x86_64 db4-4.7.25-19.el6_6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-42.el6.x86_64 libcom_err-1.41.12-22.el6.x86_64 libcurl-7.19.7-46.el6.x86_64 libidn-1.18-2.el6.x86_64 libselinux-2.0.94-5.8.el6.x86_64 libssh2-1.4.2-1.el6_6.1.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-20.el6.x86_64 mysql-libs-5.1.73-5.el6_6.x86_64 nspr-4.10.8-1.el6_6.x86_64 nss-3.18.0-5.3.el6_6.x86_64 nss-util-3.18.0-1.el6_6.x86_64 openldap-2.4.40-5.el6.x86_64 openssl-1.0.1e-42.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 __pthread_mutex_destroy (mutex=0x7fd32cce7be8) at pthread_mutex_destroy.c:28 #1 0x00007fd32c86f9e6 in _mongoc_ssl_thread_cleanup () at src/mongoc/mongoc-ssl.c:555 #2 _mongoc_ssl_cleanup () at src/mongoc/mongoc-ssl.c:106 #3 0x00007fd32c8606a9 in _mongoc_do_cleanup () at src/mongoc/mongoc-init.c:127 #4 0x000000344480cdb3 in pthread_once () at ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_once.S:104 #5 0x00007fd32c8523df in __do_global_dtors_aux () from /usr/lib64/libmongoc-1.0.so.0.0.0 #6 0x0000000000000000 in ?? () (gdb) mongo-c-driver version 1.1.10 version: kamailio 4.3.2 (x86_64/linux) b5980b-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, 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: b5980b -dirty compiled on 13:35:24 Sep 28 2015 with gcc 4.

3 11

UacRegInfo flags
by DanB 06 Nov '15

06 Nov '15

Hey Guys, I have been digging around but could not find some proper documentation on flags which are coming back in the UacRegInfo rpc query. Any chance I can find out from the flags what is the status of that registration(whether password failed or user online)? Thanks in advance for any kind of tip! DanB

2 2

Webpage Display
by Ryan Holbein 05 Nov '15

05 Nov '15

Can any help me with this .. please go to 65.61.9.54 and look at webpage how do I fix it?

1 0

q-value on unsupported clients
by Daniel W. Graham 05 Nov '15

05 Nov '15

I have a need to use q-value for serial forking but my clients don't support it. I need to specify unique q-value on all ports on the client device. I explored using the User-Agent header and appending q-value, but this wouldn't help on multi-port clients. Some things I have been thinking about are 'abusing' the authorization user or from user to somehow append q-value but this opens the door to other issues, and a database trigger however there would be no way to match this up to a record in another table as the usernames would be the same. Any thoughts/ideas or methods others have used in situations like this? -Dan

4 7

Branch & Forking
by Marino Mileti 05 Nov '15

05 Nov '15

I'm playing with branch and i'm using this simple route (copied from kamailio website) route { seturi("sip:a@example.com"); append_branch("sip:b@example.com"); append_branch("sip:c@example.com"); t_relay(); break; } A B & C rings at the same time. If B or C aren't registered, and the caller send a CANCEL, it receives also a 404 messsage because for example B wasn't online. Is it possible that Kamailio "absorb" this 404? I don't want that my caller receive it :) Marino Maria Mileti marino.mileti(a)alice.it --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Adding proprietary event in kamailio
by Priyaranjan Nayak 05 Nov '15

05 Nov '15

Hi All, If I send a SUBSCRIBE message with "Event: presence" then I am getting 200 OK. When I am sending some proprietary event(i.e. Event: as-feature-event ) then I am getting 489 Bad event. Could you please help me how can I add proprietary event in the kamailio-4.2.5 to handle the SUBSCRIBE message with proprietary event ? Thanks Priyaranjan

2 6

pv_printf buffer for spec value
by Sven Neuhaus 05 Nov '15

05 Nov '15

Hello, I'm getting these errors since the last couple of changes I made to the Kamailio 4.3.x configuration: ERROR: <core> [pvapi.c:1361]: pv_printf(): no more space for spec value ERROR: <core> [pvapi.c:1370]: pv_printf(): buffer overflow -- increase the buffer size... I'm not even using pv_printf() so this error is a bit surprising. The first thing I tried was increasing the buffer from its default size of 1024 to 4096 (pv_buffer_size=4096) but the error messages persist. What other functions use this buffer? Regards, -Sven Neuhaus

3 10

RTP Proxy to integrate with kamailio
by Safdar Khan 05 Nov '15

05 Nov '15

Hi friends, Its been two days and still finding a way to install rtp proxy and integrate it with kamailio. I have checked running status of rtp proxy by *netstat* command in this way... *#netstat -pln | grep rttpproxy* and i got the output like that(i assume its working): *udp 0 0 127.0.0.1:7722 <http://127.0.0.1:7722> 0.0.0.0.*:* Well here is my setup of kamailio and rtp proxy... 1.I am using two computers one for running kamailio and another one for rtp proxy. 2.Both are running seperately. 3.Two android mobile phones to test the service(App using imsdroid). I am little bit confused with ip settings in kamailio configuration file in the part where *rtpproxy_sock* is given.For now i have given the *ip:port of *the computer on which rtp proxy is running separately. And i am getting the error message in *syslog* like this *can't send command to a RTP proxy* *proxy <udp:192.168.5.97:7722 <http://192.168.5.97:7722>> does not respond, > disable it* *can't get version of the RTP proxy* Actually i want to record the calls. Any kind of help to configure kamailio with rtp proxy is appreciated.

4 4

listening on multiple port
by Thibault Gueslin 04 Nov '15

04 Nov '15

I would like to set up kamailio in order to listen to multiple port. I have set : port=5060 port=5228 But it always listen on the port indicated on last line: Listening on udp: 127.0.0.1:5228 udp: x.x.x.x:5228 tcp: 127.0.0.1:5228 tcp: x.x.x.x:5228 Aliases: I don't have any "listen" in my config file. Is it supposed to work like that ?

4 7

[CFP] FOSDEM 2016, RTC devroom, speakers, volunteers neeeded
by Daniel Pocock 04 Nov '15

04 Nov '15

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2016 takes place 30-31 January 2016 in Brussels, Belgium. http://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (including the Saturday night dinner), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 30 January 2016. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Speaking opportunities ---------------------- Note: if you used Pentabarf before, please use the same account/username Main track: the deadline for main track presentations is midnight today, 30 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit Real-Time Communications dev-room: deadline 27 November Please also use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM16/ Other dev-rooms: some speakers may find their topic is in the scope of more than one dev-room. It is permitted to apply to more than one dev-room but please be kind enough to tell us if you do this. You can find the full list of dev-rooms at https://www.fosdem.org/2016/schedule/tracks/ Lightning talks: deadline 27 November The lightning talks are an excellent opportunity to introduce a wider audience to your project. Given that dev-rooms are becoming increasingly busy, all speakers are encouraged to consider applying for a lightning talk as well as a slot in the dev-room. On the "General" tab, please look for the "Track" option and choose "Lightning Talks". https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. For any questions, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 30 January - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists FOSDEM is made possible by volunteers and if you have time to contribute, please feel free to get involved through https://volunteers.fosdem.org/ Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 28 and 29 January 2016 http://wiki.xmpp.org/web/Summit_19 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially on 29 January. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 29 January On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: http://planet.jabber.org ralphm(a)ik.nu http://planet.sip5060.net daniel(a)pocock.pro http://planet.opentelecoms.org daniel(a)pocock.pro Please also link to the Planet sites from your own blog or web site. Contact ======= For discussion and queries, please join the free-rtc mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Daniel Pocock <daniel(a)pocock.pro> Ralph Meijer <ralphm(a)ik.nu> Saúl Ibarra Corretgé <saul(a)ag-projects.com> Iain R. Learmonth <irl(a)debian.org>

3 3

Re: [SR-Users] Avoid multiple registration
by Daniel-Constantin Mierla 04 Nov '15

04 Nov '15

Ok, good to know it was solved. For the records, if you want to have different number of max contact per user, then you can rely on the xavp configured to registrar xavp_cfg parameter, e.g.,: modparam("registrar", "xavp_cfg", "reg") $xavp(reg=>max_contacts) = 2; save("location"); Cheers, Daniel On 04/11/15 12:23, Marino Mileti wrote: > > Thank you Daniel!! > > I've removed the 0x04 flag and now works great!! > > > > Cheers, > > Marino > > > > On 03/11/15 17:30, Daniel-Constantin Mierla wrote: > > > > Hello, > > On 03/11/15 11:29, Marino Mileti wrote: > > I would like to avoid multiple registration with the same AOR... > > I know that there's a parameter for this (max_contacts) but if I > set it to 1, all REGISTER overwrite location info. > > > max_contacts will reject new registrations/contacts if there is > already a valid contact in the location table. > > A flag as second parameter to save(...) is the option to overwrite > existing contact with the one from current registration request. > > Cheers, > Daniel > > > > Instead i would like to reply with error if a REGISTER arrive with the > same AOR just registered on location server. > > > > Is that possible? Any help is appreciated J > > > > /Marino Maria Mileti/ > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > > Avast logo <https://www.avast.com/antivirus> > > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat > > > ------------------------------------------------------------------------ > Avast logo <https://www.avast.com/antivirus> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

Re: [SR-Users] Avoid multiple registration
by Daniel-Constantin Mierla 04 Nov '15

04 Nov '15

Hello, On 03/11/15 11:29, Marino Mileti wrote: > > I would like to avoid multiple registration with the same AOR... > > I know that there's a parameter for this (max_contacts) but if I set > it to 1, all REGISTER overwrite location info. > max_contacts will reject new registrations/contacts if there is already a valid contact in the location table. A flag as second parameter to save(...) is the option to overwrite existing contact with the one from current registration request. Cheers, Daniel > > > > Instead i would like to reply with error if a REGISTER arrive with the > same AOR just registered on location server. > > > > Is that possible? Any help is appreciated J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo <https://www.avast.com/antivirus> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

2 1

Need to increase performance to handle more SUBSCRIBE messages
by Priyaranjan Nayak 04 Nov '15

04 Nov '15

Hi All, I am sending 300 REGISTER per second and getting response properly.When I am sending 50 SUBSCRIBE per second the Kamailio is not responding properly. Now We are using kamailio-4.2.5 version. Is there any procedure to increase the performance to handle more SUBSCRIBE in a second ? Thanks, Priyaranjan

2 3

Re: [SR-Users] Avoid multiple registration
by Olle E. Johansson 04 Nov '15

04 Nov '15

> On 03 Nov 2015, at 11:29, Marino Mileti <marino.mileti(a)alice.it> wrote: > > I would like to avoid multiple registration with the same AOR... > I know that there's a parameter for this (max_contacts) but if I set it to 1, all REGISTER overwrite location info. > > Instead i would like to reply with error if a REGISTER arrive with the same AOR just registered on location server. > > Is that possible? Any help is appreciated J > A simple glance at the documentation page for the REGISTRAR shows me a function called “registred”. http://kamailio.org/docs/modules/4.3.x/modules/registrar.html#registrar.f.r… "The function returns true if the AOR in the URI is registered, false otherwise. “ If you use that one and refuse a registration if true, you will get exactly what you want. Reading available documentation is a good thing (TM) :-) Regards, /Olle

1 0

Serial Forking and Lookup Function
by Daniel W. Graham 04 Nov '15

04 Nov '15

Looking for some clarification / direction on setting up forking. I have configured clients to use q-value and is stored in location table. I see several examples as described in TM module documentation but does not mention use of lookup() function. Per registrar documentation- lookup() The functions extracts username from Request-URI and tries to find all contacts for the username in usrloc. If there are no such contacts, -1 will be returned. If there are such contacts, Request-URI will be overwritten with the contact that has the highest q value and optionally the rest will be appended to the message (depending on append_branches parameter value). Append_branch parameter The parameter controls how lookup function processes multiple contacts. If there are multiple contacts for the given username in usrloc and this parameter is set to 1, Request-URI will be overwritten with the highest-q rated contact and the rest will be appended to sip_msg structure and can be later used by tm for forking. Does lookup() build the destination set and then I can proceed by doing t_load_contacts(), t_next_contacts() and setup failure route? Any direction is appreciated! Thanks, -dan

2 2

Huge TLS performance regression migrating from 3.3.7 to 4.2.6
by Tristan Mahé 03 Nov '15

03 Nov '15

Hi List, I've been reaching a strange thing lately, trying to upgrade to kamailio 4.2.6 ( identical config ): - 35% loss of performance on TLS connections ( Sipp REGISTER scenario, easily reproduced ). - more TCP workers needed to ease the TCP queues ( 32 workers on 3.3.7, at least 64 are needed on 4.2.6 to avoid "queue full" messages in logs ). - kamailio 4.2.6 intermittent processing of requests ( Kamailio stops accepting new tcp connections, then resume a bit later, then stops again,... ) => starts happening when Kamailio is treating more than 20k tcp connections, with between 500-1k req/s. During those periods, trying to connect to kamailio with an openssl s_client -connect will end up in a timeout or with a long delay before the certificate is received ( 03 CONNECTED ). - kamailio 4.2.6 dropping connections, kamailio 3.3.7 keeping those connections: after a while, the number of active connections starts to decrease on kam4, on kam3 it stays constant. I was wondering if I was alone in this situation or if someone else is impacted, has some advices ? It does not seem to be a config issue, as it is the same that was deployed with both versions.

2 8

Testing RTP Proxy
by Safdar Khan 03 Nov '15

03 Nov '15

Hi friends, I have configured kamailio to work with rtp proxy and i have followed this link http://nil.uniza.sk/sip/kamailio/rtp-proxy-kamailio-32-configuration-debian… However i am not sure that it is working perfectly or not. My Question is.. 1.Is above link ensures that rtp proxy will work with kamailio or there is another way to deal with rtp proxy and kamailio? 2.If the above link is successful integration of rtp proxy and kamailio then how to test it. Thank you

2 1

Re: [SR-Users] Domain register_myself
by DanB 03 Nov '15

03 Nov '15

Hey Daniel, That would make sense also why it worked for me in the past! Many thanks for so fast answer! DanB

1 0

Avoid multiple registration
by Marino Mileti 03 Nov '15

03 Nov '15

I would like to avoid multiple registration with the same AOR... I know that there's a parameter for this (max_contacts) but if I set it to 1, all REGISTER overwrite location info. Instead i would like to reply with error if a REGISTER arrive with the same AOR just registered on location server. Is that possible? Any help is appreciated J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

host for pstn.gw
by Max 03 Nov '15

03 Nov '15

Hi. I'd like to use Kamailio as a SIP proxy routing calls to upstream SIP provider. As far as I've understood the parameters I should set are: pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" The problem is that my upstream provider uses dynamic IP for its server. How can I use hostname instead of ip. >From what I see in route[PSTN] example it seems like it should just work if I use pstn.gw_ip = "sip.myprovider.lol" desc "My upstream SIP host" But if so - why the confusing name? And if not - how do I make it work? Is there pstn.gw_host or smth like that? regards, Max.

3 2

DB Problems after upgrade
by Nahum Nir 02 Nov '15

02 Nov '15

Hi All, I had a working Kamailio with Siremis. I upgraded wanted a new cfg file (big mistake, I know). I defined the DBURL at kamailio.cfg and the server start without errors, users can register but I can not make calls and Siremis missed calls table is not updated. What files and where should I update? Should I run kamdbctl? Thanks, Nir

2 1

Kamailio configuration file rtpproxy_manage not found when starting
by amar Smart Telecom 02 Nov '15

02 Nov '15

Dear All, I was trying to make such that rtpproxy is used for all calls in kamailio, Part of Kamailio.cfg looks like following: ============================= loadmodule "rtpproxy.so" # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") 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"); rtpproxy_manage("co") ; # added amar 20151015 } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } But when I run Kamailio start it gives error: Error looks like following: =================== Oct 15 16:13:42 smart kamailio[18472]: * Not starting Kamailio SIP server: invalid configuration file! Oct 15 16:13:42 smart kamailio[18472]: * Oct 15 16:13:42 smart kamailio[18472]: * 0(18480) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command rtpproxy_manage (params 1) What could be the problem?

3 3

Kamailio E164 Normalisation
by Daniel Bryars 02 Nov '15

02 Nov '15

Hello, I have a requirement to normalise a number to an international E164 inside Kamailio. In my scenario I'm receiving a call from "a platform", and then making a call out to the same platform. I'm then adding some services to that call (e.g. recording it) on the way through. The problem is that sometimes the number I receive in the FROM field in the initial invite from "A platform" has 44 at the front, but no +. "A Platform" is then rejecting the call back in from myself, because the FROM field isn't formatted correctly. Scenario 1 - WORKS FINE - sometimes we see this. PSTN -> FROM:07700900000 -> A PLATFORM -> FROM:07700900000 -> ME Call Centre <- A PLATFORM <- FROM:07700900000 <- ME Scenario 2 DOES NOT WORK - sometimes we see this. PSTN -> FROM:447700900000 -> A PLATFORM -> FROM:447700900000 -> Aeriandi Call Centre <- A PLATFORM <- FROM:447700900000 <- Aeriandi Scenario 3 WORKS by fixing up the number by adding a + PSTN -> FROM:447700900000 -> A PLATFORM -> FROM: 447700900000 -> Aeriandi Call Centre <- A PLATFORM <- FROM:+447700900000 <- Aeriandi (ADD a PLUS) So I'm investigating if it's possible to fix up the number and add the + in Kamailio. Clearly I could detect if the number starts with 44 and change this to +44, but this seems very brittle (what about other international numbers) - it there a proper way to do this, or a standard module? Unfortunately asking "A platform" to send me the "FROM" in a consistent format is not an option. Many Thanks, Daniel

3 2

kamctl fifo t_uac_dlg TLS
by Ahmed Salem 02 Nov '15

02 Nov '15

Hi, I have a kamailio tls configuration which is working fine for calls & messages In the tls configuration i set these parameters to yes for the server and client verify_certificate = yes require_certificate = yes But when I use "kamctl fifo t_uac_dlg MESSAGE" to send a message I get the following errors, when verify_certificate & require_certificate are set to no the message is sent correctly. Can somebody please help me resolve this issue. tls [tls_server.c:1193]: tls_read_f(): TLS write:error:14090086:SSL routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading tls [tls_server.c:1193]: tls_read_f(): TLS accept:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Thanks Ahmed

2 5

Domain register_myself
by DanB 02 Nov '15

02 Nov '15

Hey Guys, I was recently playing with "register_myself" setting of the domain module (which I was the impression that I have used it successfully in the past) and I have the feeling it does not longer behave. I have the following config: """ modparam("domain", "db_url", DBURL) modparam("domain", "register_myself", 1) .... # block foreign domains here if !is_uri_host_local() { sl_send_reply("603","Only local domains accepted"); exit; } """ Based on the register_myself setting I would expect that is_uri_host_local passes the request but it does not although the request is towards the IP address of my server. Is this the intended behavior or am I just unlucky and hit a bug? Ta, DanB

2 1

dmq_usrloc for writing to DB on one machine
by Sebastian Damm 02 Nov '15

02 Nov '15

Hi, I evaluated the dmq_usrloc module, and it works pretty good if there are two identical machines which replicate their location with each other. But what I actually want to do is to send registrations to an extra machine which is just there for writing the location into a central DB (in one table for all the registrars), from where it can be accessed (by a web page for example). So for example, I have three machines, registrar 1 and registrar 2, and receiver. receiver will never interact with customers. It is just there for writing the location into the database. I configured both registrars to send their packets to receiver, and I configured the receiver to send its packets (which it actually never does) to a dns name containing both registrars. But after reading the content of some PING messages and reading something about "autodiscovery" of dmq on the mailing list, I'm not sure whether my scenario will work the way I want it to. Can somebody explain whether I can use dmq_usrloc for this purpose? Or is it only intended to be used for bidirectional replication? Best Regards, Sebastian

2 5

issues with RTP on different hosts & dispatcher mod
by Infinicalls Infinicalls 02 Nov '15

02 Nov '15

Hi, I whenever I try to add rtpproxy to my config file modparam("nathelper", "rtpproxy_sock", "udp:firsthost:12221 udp:secondhost:12221") , I get this error. 0(45810) ERROR: <core> [modparam.c:139]: set_mod_param_regex(): parameter <rtpproxy_sock> of type <1> not found in module <nathelper> 0(45810) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 423, column 82: Can't set module parameter ERROR: bad config file (1 errors) 0(45810) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized I am having with_nat. Secondly, I tried with dispatcher and it works for one way calls. ie. User1 is able to call User2, but vice versa is not possible. Both RTP1 and RTP2 are in different machines and Kamailio sits on a different machine. All have identical network setups. And no NAT. Any idea. Can somebody guide me? thanks. regards Ganesh Kumar

2 1

100% CPU usage after NTP date update
by Stefan Kohlhauser 02 Nov '15

02 Nov '15

Hello everyone! Setup: I am currently using a Kamailio 4.2.3 on an embedded device (low CPU and RAM). The device is not able to store the current date, so after a reboot its 1. Jan 1970 until NTP updates the date. Problem: If the date is updated after the Kamailio has started (a leap of 45 years) the Kamailio uses up all the CPU and renders the device useless for a very long time. Notes: There are no incomming requests at the time. (But the Kamailio is not responsive during that time anyway.) Adjusting the internal Kamailio-time seems to take longer when the date update leap is bigger. My questions: 1) Can I prevent the Kamailio from using up the entire CPU after a date update? Or maybe do the internal ticks-update more gracefully? 2) For my better understanding: What happens within the Kamailio after adjust_ticks has updated the internal date? Thanks in advance for your help! Best regards, Stefan

2 4

Multi-Array with AVPs
by Jonathan Hunter 02 Nov '15

02 Nov '15

Hi Guys, Hope all are well? sorry to post but Im just trying to get something working where Im trying to convert a php function script into the kamailio.cfg. Im getting there but wonder how I can group results of sets of avps with another avp. Say I have these variables; $(avp(NTgrp)) = 'SB100_000361@1.2.3.4|SB100_000213@1.2.3.5|SB100_000225@1.2.3.6|SB100_000361@1.2.3.7|SB100_000213@1.2.3.8';$(avp(Prio)) = '1|1|1|2|2';$(avp(Wgt)) = '10000|10000|80000|30000|70000';$(avp(CInc)) = '0|0|1|0|1';$(avp(WTot))= '100000|100000|100000|100000|100000'; I then run a while script to split up their variables via the delimiter | in user; $var(j)=0;while($var(j)<$var(count)) {$(avp(NTgrpP))=$(avp(NTgrp){s.select,$var(j),|});$(avp(PrioP))=$(avp(Prio){s.select,$var(j),|});$(avp(WgtP))=$(avp(Wgt){s.select,$var(j),|});$(avp(CIncP))=$(avp(CInc){s.select,$var(j),|});$(avp(WTotP))=$(avp(WTot){s.select,$var(j),|});xlog("L_INFO"," Now Each time we have $(avp(NTgrpP)) $(avp(PrioP)) $(avp(WgtP)) $(avp(CIncP)) $(avp(WTotP))\n");$var(j)=$var(j)+1;} This then results in;INFO: <script>: Now Each time we have SB100_000223(a)1.2.3.4 1 10000 0 100000INFO: <script>: Now Each time we have SB100_000225(a)1.2.3.5 1 10000 0 100000INFO: <script>: Now Each time we have SB100_000386(a)1.2.3.6 1 80000 0 100000INFO: <script>: Now Each time we have SB100_000213(a)1.2.3.7 2 30000 0 100000INFO: <script>: Now Each time we have SB100_000361(a)1.2.3.8 2 70000 0 100000 Now what Id like to do is group them via priorities so I can do some further functional checks on them when in same priority group, as you can see there are 3 results at priority 1 and 2 at priority 2. Is there away I can create something of a multi-array using the AVP $(avp(PrioP)) I just created so I can group by priority? As in php I would do something like; $multi_arr[$prio_arr[$p]][] = array( "ntgrp" => $ntgrp_arr[$p], "prio" => $prio_arr[$p], "wght" => $wght_arr[$p], "cinc" => $cinc_arr[$p], "wtot" => $wtot_arr[$p], ); } Any help would be great. Many thanks Jon

3 6

Q: about CRL list (TLS)
by Vladimer Gabunia 02 Nov '15

02 Nov '15

hello all. we compiled kamailio with TLS Support. but have next problem when using CRL Lits. Our Certificate issuing scheme is follow: Offline Root CA -> Enterprise SubCA -> Server and Phone Certificate CRL list is signed by SubCA. option "require client certificate is enables (1) " When we enable CRL list, phones are not registered. CA file is offline RootCA certificate in pem format. We think that the reason is that СRL was signed by Subca or incorrect CRL format. CRL is converted from MS CRL to PEM. (What is the format for the CRL) maybe someone have experiance with similar scenarios? thanks in advance!! ________________________________ [gh.ge] ვლადიმერ გაბუნია IT სამსახურის უფროსი ტელ: (+995) 32 2505222 +8183 მობ: (995) 577 095333 შპს "ჯეო ჰოსპიტალს" სათავო ოფისი თბილისი 0160, ვაჟა-ფშაველას გამზ. № 16; http://www.gh.ge <http://gh.ge>

2 5

Problems with dispatcher module | feature disabled
by Alessio Casco 02 Nov '15

02 Nov '15

Hello! We noticed today an issue with the kamailio loadbalancing. Basically kamailio is not considering if an host is up or not, backends are set always as "FLAGS: AP" even if the host is down. I see this when the proxy starts : ERROR: dispatcher [dispatcher.c:768]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled These the parameters I have set: # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;code=404") modparam("dispatcher", "ds_ping_from", "sip:proxy@sip.nexmo.com") modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_hash_size", 10) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 10) modparam("dispatcher", "use_default", 0) modparam("dispatcher", "attrs_pvname", "$avp(carrierattr)") modparam("dispatcher", "setid_pvname", "$avp(carriergroup)") modparam("dispatcher", "flags", 2) I use 4.3.3, I think this issue started after upgrading from 4.3.1 but I can't say for sure. Can someone please give us some advices on this? Thanks Alessio

2 5

Which option uses for check certificate revocation in kamailio 1.5.x ?
by Evgeniy Ivanov 02 Nov '15

02 Nov '15

Hello! I use kamailio 1.5.4-tls (i386/linux). I configured the tls options: disable_tls = 0 listen=tls:192.168.50.60:5061 tls_certificate="/etc/kamailio/tls/crt.pem" tls_private_key="/etc/kamailio/tls/key.pem" tls_ca_list="/etc/kamailio/tls/ca.pem" tls_verify_client=on tls_require_client_certificate=on I need to check certificate revocation, but the option tls_crl does not work. Which option uses for check certificate revocation in kamailio 1.5.x ?

3 6

Display name
by Loïc Boissy 30 Oct '15

30 Oct '15

Hello, I would like to know if kamailio is able to store contact "display name" in its database? Thank you for your help. Best regards. Loïc

2 2

Routing outgoing call from asterisk to registered SIP provider
by Bruno Salzano 29 Oct '15

29 Oct '15

Hello, i've DID provider that require to register to it's server. I've been able to configure this provider using kamailio and uac and to route all inbound calls from DID provider to internal asterisk media server. All is working great. Now i need help to configure call routing from internal asterisk to the external DID provider. I've done something like: route[LOCATION] { if(is_method("INVITE")) { route(FROMASTERISK); route(TOASTERISK); exit; } [...] } route[FROMASTERISK] { if (ds_is_from_list()) { xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server Cluster\n"); rtpengine_manage("direction=internal direction=external replace-origin replace-session-connection trust-address"); $ru = "sip:TESTING_NUMBER@EXTERNAL_PROVIDER_IP"; route(RELAY); exit; } } It seems that when i route the call outbound, i get a 407. So i would like to react to this using uac_auth() as described here: http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth() Is this the right way? Can someone help me, giving hints and some working example? Using kamailio 4.3.3 with rtpengine 4.1.0 Best regards, Bruno -- The life is short.. live at your best!

4 6

gcc warning for t_cancel
by Yasin CANER 27 Oct '15

27 Oct '15

3 3

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