- sr-users - kamailio.org

Kamailio and RTPProxy in-front of freeswitch
by Achintha 04 Aug '16

04 Aug '16

hi all, i have configured kamailio and RTPProxy on one ubuntu server then freeswitch on another server. all servers installed on digital ocean. i was tried to build following setup, SIP_Call------> kamailio+RTPPROXY-------> freeswitch put following configurations on kamailio. modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") modparam("rtpproxy", "rtpproxy_retr", 2) modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)") mhomed=1 route[FROM_TRUNK] #force_send_socket("pu.bl.ic.ip "); rtpproxy_manage("","pu.bl.ic.ip "); $du="sip:ip.of.free.switch:5080"; route(RELAY); call was connected but rtp is not going through the kamailio server like folows, SIP FLOOR SIP_Call------> kamailio-------> freeswitch SIP_Call<------ kamailio<------- freeswitch RTP FLOOR SIP_Call ------> freeswitch SIP_Call <------ freeswitch please help me to configure kamailio and RTPProxy to manage RTP -- thanking you.. Achintha

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Re: [SR-Users] Path & Loose Route
by Luke Milbourne 04 Aug '16

04 Aug '16

Hi Daniel, Thanks again for your reply again. I remembered about the debug setting and done it yesterday, not at my pc right now but I remember one of the log lines was saying "the topmost route header is me". Still trying to understand it, but will get a copy of the log lines for you, if you wouldn't mind taking a look. Thanks again, Luke On 4 Aug 2016 06:57, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: Hello, set debug=3 in kamailio.cfg and look at log messages in syslog. If you find it too verbose, use dispatcher module and set debug level only for core and rr modules to 3. Cheers, Daniel On 03/08/16 13:22, Luke Milbourne wrote: Hi Daniel, I've added some more logging and can see loose_route() is returning -1 - route calculation has been unsuccessful. Is there a way of finding out why this is the case? Kind regards, Luke On 2 August 2016 at 09:49, Luke Milbourne <luke.milbourne(a)gmail.com> wrote: > I added the below xlog lines into my script: > > xlog("L_NOTICE", "Before loose_route: $si -> $ru - $du\n"); > xlog("L_NOTICE", "Before record_route: $si -> $ru - $du\n"); > xlog("L_NOTICE", "After both: $si -> $ru - $du\n"); > > Which generated the below output in syslog: > > Aug 2 09:46:11 ip-10-10-2-5 /sbin/kamailio[529]: NOTICE: <script>: Before > loose_route: 10.10.0.25 -> sip:201@192.168.1.114:5062 - <null> > Aug 2 09:46:11 ip-10-10-2-5 /sbin/kamailio[529]: NOTICE: <script>: Before > record_route: 10.10.0.25 -> sip:201@192.168.1.114:5062 - <null> > Aug 2 09:46:11 ip-10-10-2-5 /sbin/kamailio[529]: NOTICE: <script>: After > both: 10.10.0.25 -> sip:201@192.168.1.114:5062 - <null> > > Seems that the destination URI is never set, the Route header is below: > > Route: <sip:201@10.10.2.5:5072>;lr;received=sip:[PUBLIC_IP]:45126 > > The use_received modparam for path is also set to 1. > > Any idea's/suggestions are greatly appreciated :) > > Kind regards, > Luke > > On 2 August 2016 at 09:40, Luke Milbourne <luke.milbourne(a)gmail.com> > wrote: > >> Morning Gents, >> >> Firstly, thanks for taking the time to reply :) >> >> Kamailio is doing the NAT detection and simply acting as a passthrough to >> FreeSWITCH (which handles all SIP registrations etc). >> >> Will setup the log lines shortly and share the results. >> >> kamailio -V outputs; >> >> version: kamailio 4.3.4 (x86_64/linux) 83e842 >> 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: 83e842 >> compiled on 18:22:39 Jan 26 2016 with gcc 4.9.2 >> >> Kind regards, >> Luke - Chicane Support >> >> On 2 August 2016 at 07:52, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> yes, loose_route() should be used there. Can you add a log message and >>> print $ru and $du before and after loose_route() in config and then send >>> the result to this mailing list? >>> >>> Btw, what version of Kamailio are you using? >>> >>> Cheers, >>> Daniel >>> >>> On 01/08/16 14:46, Luke Milbourne wrote: >>> >>> Is it correct to use loose_route() along side the path module? >>> >>> On 1 August 2016 at 13:33, Luke Milbourne <luke.milbourne(a)gmail.com> >>> wrote: >>> >>>> Hi Daniel, >>>> >>>> Yep, I've got the below in my kamailio.cfg >>>> >>>> modparam("path", "use_received", 1) >>>> >>>> Kind regards, >>>> Luke >>>> >>>> On 1 August 2016 at 13:25, Daniel-Constantin Mierla <miconda(a)gmail.com> >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> On 01/08/16 14:14, Luke Milbourne wrote: >>>>> >>>>> Hi all, >>>>> >>>>> Currently trying to use Kamailio as a passthrough to some FreeSWITCH >>>>> boxes. >>>>> >>>>> When receiving an invite from FreeSWITCH which contains a Route header >>>>> including received information, example below; >>>>> >>>>> Route: <sip:201@10.10.2.5:5072>;lr;received=sip:[PUBLIC_IP]:45126 >>>>> >>>>> When using loose_route() Kamailio still send the invite to the rfc1918 >>>>> address instead of the received IP:PORT in the Route header. >>>>> >>>>> Am I misunderstanding how this should work? >>>>> >>>>> >>>>> if you use path module, have you enabled the option to use the >>>>> received parameter? >>>>> >>>>> >>>>> https://www.kamailio.org/docs/modules/stable/modules/path.html#idp38033932 >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> -- >>>>> Daniel-Constantin Mierlahttp://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Luke Milbourne >>>> >>>> Tel: 07857154817 >>>> Google Talk/Email: luke.milbourne(a)gmail.com >>>> >>> >>> >>> >>> -- >>> Luke Milbourne >>> >>> Tel: 07857154817 >>> Google Talk/Email: luke.milbourne(a)gmail.com >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> >> -- >> Luke Milbourne >> >> Tel: 07857154817 >> Google Talk/Email: luke.milbourne(a)gmail.com >> > > > > -- > Luke Milbourne > > Tel: 07857154817 > Google Talk/Email: luke.milbourne(a)gmail.com > -- Luke Milbourne Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlahttp://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ 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

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Kamailio on Mac
by Carlos Ruiz Díaz 04 Aug '16

04 Aug '16

Hi all, I'm switching my development environment from Linux to Mac. Is there any tutorial on how to setup Kamailio on this OS? It may be an Unix flavor, but it's still quite different from Linux :). Thanks, -- Carlos http://caruizdiaz.com http://ngvoice.com +52 55 3048 3303

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Kamailio + rtpengine + freeswitch
by Kotb, Amir 04 Aug '16

04 Aug '16

[03:50] <meamo> I have a small query, can anybody assist please? [03:51] <meamo> I am running kamailio + rtpengine on a google compute cloud instance [03:51] <meamo> have setup them to work with devices behind nat, and everything works [03:52] <meamo> no i have added freeswitch. but I don't know how to configure rtpengine to work with both [03:52] <meamo> when I use #!define with_nat and #!define with_freeswitch, I get no voice in the calls. When I remove, #!define with_freeswitch, everything works normally Best regards, Amir -- Amir KOTB, Msc., Bsc. Postgraduate Researcher Department of Electrical Engineering & Electronics, The University of Liverpool, Brownlow Hill, Liverpool L69 3GJ, UK Mobile: +44-(0) 7428844234 Email: A.Kotb(a)liverpool.ac.uk Skype: A.kotb1 Web: Http://uk.linkedin.com/in/AOKotb Please consider the environment before printing this email

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Path & Loose Route
by Luke Milbourne 04 Aug '16

04 Aug '16

Hi all, Currently trying to use Kamailio as a passthrough to some FreeSWITCH boxes. When receiving an invite from FreeSWITCH which contains a Route header including received information, example below; Route: <sip:201@10.10.2.5:5072>;lr;received=sip:[PUBLIC_IP]:45126 When using loose_route() Kamailio still send the invite to the rfc1918 address instead of the received IP:PORT in the Route header. Am I misunderstanding how this should work? -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

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Help with dispatcher ping: Kamailo 4.1
by Andy Chen 03 Aug '16

03 Aug '16

Hi, I'm currently running Kamailio 4.1 and was wondering if there is a way to control ds_ping_interval on a dispatcher setid. Thanks. -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) achen(a)thinkingphones.com

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Fail to register at trunk through UAC
by Yuriy Gorlichenko 03 Aug '16

03 Aug '16

Hi. All trunks works fine ony one not works Kamailio sends REGISTER with proxy auth provider answers with 407 Kamailio not send any REGISTER Just answers in log uac_reg_tm_callback(): authentication failed for <trunkname> At attachement kamialio debug 3 log and sip log

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SSL Certificate (Second Try)
by JCOT 02 Aug '16

02 Aug '16

I'm not sure where to send this information so I hope it's okay to share it with the mailing list, as well as the whois registrants: I went to register for a new account at https://serweb.iptel.org/user/reg. Firefox told me that the SSL certificate was bad. Specifically: -- serweb.iptel.org uses an invalid security certificate. The certificate is not trusted because it was signed using a signature algorithm that was disabled because that algorithm is not secure. The certificate expired on 10/22/2014 12:39 AM. The current time is 08/01/2016 09:55 PM. -- Thank you.

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SSL Certificate
by JCOT 02 Aug '16

02 Aug '16

I'm not sure where to send this information so I hope it's okay to share it here: I went to register for a new account at https://serweb.iptel.org/user/reg. Firefox told me that the SSL certificate was bad. Specifically:

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A: Upgrading kamailio.org os
by Daniel-Constantin Mierla 02 Aug '16

02 Aug '16

Hello, a short note to announce that the os on kamailio.org server will be upgraded in the near future, likely next week, exact date still to be decided. Hopefully there will be only very short downtime intervals. Main affected services: website (including the wiki) and mailing lists. I will send a notification once the procedure starts, but I wanted to allow enough time for this information to propagate properly within the community. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Saving and extracting contact header
by Nahum Nir 01 Aug '16

01 Aug '16

Hello, I am quite a newbe in Kamailio scripts and further to my effort to implement push notification, I am adding to the contact header device ID. I have two problems: 1. When the register of the callee expires Kamailio frees the record so the information on the calle is not available. 2. Even when the callee is registered I do not how to access the contact field. Tried reg_fetch_contacts(...,"callee") and there is a lot of info but not the contact header. Thanks and regards, Nir

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CDP -Could not open module error
by Andrew Okri 01 Aug '16

01 Aug '16

Hi All, I have just installed KamailioIMS 4.4.2 on a Ubuntu 16.04 vm and configured it according to the kamailio tutorial. When l attempt to start the icscf or scscf kamailio, it fails to start due to its inability to open the CDP module referring to the log_db_config function in particular.I have looked at the archive to see if anyone else have encountered this problem, but could not find anything. The log_db_config function takes a config object as a parameter, hence l have inspected the CDiameterPeer config file carefully and it seems both xml and diameter compliant. I am at loss where else to look. Please would anyone be kind enough to assist with getting a scscf and icscf started please. Regards Andrew

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Kamailio static comilation
by Pratik V Patel 01 Aug '16

01 Aug '16

Hello all, I want to compile kamailio open source statically. I want to remove dependency for dynamic shared libraries. can anyone help me to get out of it? I have tried by chang in Makefiles but it does not work. What the meaning of "static_modules" during compilation. It is still generating dynamic shared libraries. Please do needful. -- Thanks And Regards, Pratik V. Patel Ext: 558

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cnxcc-web error
by Infinicalls Infinicalls 01 Aug '16

01 Aug '16

Hi, I am really struggling to get the cnxcc-web installed. the /var/log/apache2/error.log says [Mon Aug 01 02:52:00.177725 2016] [:error] [pid 17534] [client 115.241.40.163:1851] PHP Deprecated: ServiceManagerAwareInterface is deprecated and will be removed in version 3.0, along with the ServiceManagerAwareInitializer. Please update your class ZfcUser\\Authentication\\Adapter\\Db to remove the implementation, and start injecting your dependencies via factory instead. in /usr/local/cnxcc/cnxcc-web/vendor/zendframework/zend-mvc/src/Service/ServiceManagerConfig.php on line 129, referer: http://xxxxxxxx.com/user/login I am getting the login screen and it freezes there. Can somebody guide me how to go about? Or is there a simple php (without Zend framework) interface available for cnxcc? thanks Ganesh Kumar

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manage rtp with or without RTP Proxy
by Achintha 01 Aug '16

01 Aug '16

hi I have configured kamailio and freeswitch on two Ubuntu servers as follows. Sip_TRUNK------> kamailio-------> freeswitch i wan to manage rtp through the kamailio server, first i want to know, 1. can i do it without RTP Proxy ? anyway i tried it with RTP Proxy as follows, I installed rtpproxy on kamailio server and start it using “rtpproxy –F -l pu.bl.ic.ip -s udp:localhost :7722 -u root” it was working Then I put modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") modparam("rtpproxy", "rtpproxy_retr", 2) modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)") mhomed=0 route[FROM_TRUNK] force_send_socket("pu.bl.ic.ip "); rtpproxy_manage("","pu.bl.ic.ip "); $du="sip:ip.of.free.switch:5080"; route(RELAY); then call is received to user and disconnected on freeswitch console get some errors [ERR] switch_core_media.c:7179 AUDIO RTP REPORTS ERROR: [Remote Address Error!] [ERR] mod_sofia.c:2290 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 3679035755 3679035755 IN IP4 pu.bl.ic.ippu.bl.ic.ip s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4167841678 RTP/AVP 123 8 0 101 c=IN IP4 pu.bl.ic.ippu.bl.ic.ip b=TIAS:64000 a=rtpmap:123 opus/48000/2 a=fmtp:123 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=16000;useinbandfec=1 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:4167941679 a=nortpproxy:yes a=nortpproxy:yes please advice me thank you

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function to check if string is an ipv4 address
by Juha Heinanen 29 Jul '16

29 Jul '16

I did not find in textops or utils modules a function to check if a string is an ipv4 address. Does it exist somewhere else? -- Juha

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RTPEngine Kamailio crashed because someone tried to do something
by Andrew Tan 29 Jul '16

29 Jul '16

I'm getting below every 2-3 weeks someone tried to connect through the CLi of RTPengine and crashed Rtpengine. Anyone knows what to make out of this ? Thanks Andrew 82440 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: INFO: <script>: HTTP Request Received 182441 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 16) 182442 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 404 Not Found#015#012Sia: SIP/2.0/TCP 89.163.135.117:63413#015#012Server: kamailio (4.3.3 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> 182443 Jul 28 16:24:21 sipr rtpengine[1224]: New cli connection from 89.163.135.117:63587 182444 Jul 28 16:24:21 sipr rtpengine[1224]: Got CLI command: 182445 Jul 28 16:24:21 sipr rtpengine[1224]: New cli connection from 89.163.135.117:63838 182446 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13028]: WARNING: <script>: HTTP request received on 5080 182447 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182448 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182449 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182450 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182451 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182452 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13021]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182453 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13028]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182454 Jul 28 16:24:21 sipr /usr/local/sbin/kamailio[13028]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182455 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13020]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182456 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13020]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading 182457 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13021]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1407609C:SSL routines:SSL23_GET_CLIENT_HELLO:http request 182458 Jul 28 16:24:22 sipr /usr/local/sbin/kamailio[13021]: ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading

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Kamailio LB: how to get Asterisk to do the RTP
by Tickling Contest 28 Jul '16

28 Jul '16

With the help of members from this mailing list (many thanks!), I finally got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make a call using the setup that looks like this: [Kamailio 4.4.2]<->[Asterisk 13.7.2] Kamailio manages REGISTERs, but also forwarding them to Asterisk. I am able to make a call, but I get only one way audio or no audio depending on which client made the call (SipDroid->Zoiper I hear one way audio on Zoiper, but no audio if the call is made the other way). I noticed that Kamailio forced direct media between the endpoints in this situation, but my application really needs Asterisk to handle it. How do I do this? Should I start by forwarding INVITEs to Asterisk? How do I do that? Any help is appreciated. Thanks!

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undefined symbol: EVP_sha1 when using outbound module
by Francisco Burgos 28 Jul '16

28 Jul '16

Hi, I am trying to use kamailio as an edge proxy to forward requests and media to an sbc. when compiling the cfg file included on the examples. i get: undefined symbol: EVP_sha1 I believe this problem is being caused by the outbound module and after some research i managed to read that it is caused by the linking between the gcc libcrypto. I would like some help since this project requires urgency. Thanks in advance.

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TM: t_relay([host, port]) DNS Queries
by Louis-Olivier Roof 27 Jul '16

27 Jul '16

Dear Kamailio users, Ref: http://kamailio.org/docs/modules/4.2.x/modules/tm.html#tm.f.t_relay Using t_relay([host, port]) to relay messages to a specific host does not seem to generate DNS queries to the said host after Kamailio has initially processed the configuration file. It resolves the host once at startup then continues using the same IP after. Is that expected behavior? Anyone has had this issue before and found a way to work around it? Note that for this specific application, messages need to be relayed to a specified host (different from current R-URI) and this host needs to be resolved by DNS during normal operation, not only at startup. Thanks! Louis-Olivier

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Checking for 200ok Response to a REGISTER request kamailio-Asterisk
by Jonathan Hunter 27 Jul '16

27 Jul '16

Hi Guys, So currently on our network we have a kamailio server which users register against, we then replicate the register messages to 2 Asterisk boxes sat behind it so that all entities are aware of the registration state of the users. REGISTER--->KAMAILIO---->ASTERISK A ---->ASTERISK B With a REGISTER---200OK exchange between Kamailio and Asterisk. We have an issue where at some points the Asterisk servers when under load dont respond with a 200 ok(something being investigated) to the register messages sent to kamailio, so I am just working on some logic for the register message to be resent using the t_replicate and t_set_fr functions. This works well should both Asterisk servers not respond, however, as I am using replicate and it is parallel forking, if say Asterisk A answers first and is available with a 200ok then that in turn cancels the register message branch being sent to Asterisk B(which I know is fine), however there could be a scenario where Asterisk B doesnt respond, and we wont know about it to try and resend the Register message, as the branch is cancelled. Hope that makes sense? I am looking at checking that both the Asterisk servers have responded and sent a 200ok, which I can grab in an onreply route but Im just wondering if someone has done something similar or has any suggestions as it is tricky to achieve currently. I have also thought about stateless working but I really need kamailio to keep retransmitting the register until it gets a response. Many thanks Jon

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Segfault in dialog.so with 4.4.2 stable
by Dirk Teurlings - Signet B.V. 26 Jul '16

26 Jul '16

Hi, Running Kamailio on Debian from the Kamailio repository with 4.4.2 stable (unpatched). Getting some random segfaults with it now, here's the relevant backtrace from the generated core. Core was generated by `/usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.'. Program terminated with signal 11, Segmentation fault. #0 run_dlg_callbacks (type=type@entry=64, dlg=dlg@entry=0x7fceb400e2f0, req=req@entry=0x7fced4f093c8, rpl=rpl@entry=0x0, dir=<optimized out>, dlg_data=dlg_data@entry=0x0) at dlg_cb.c:253 253 if ( (cb->types)&type ) { (gdb) bt #0 run_dlg_callbacks (type=type@entry=64, dlg=dlg@entry=0x7fceb400e2f0, req=req@entry=0x7fced4f093c8, rpl=rpl@entry=0x0, dir=<optimized out>, dlg_data=dlg_data@entry=0x0) at dlg_cb.c:253 #1 0x00007fcead3648f9 in dlg_terminated (dir=<optimized out>, dlg=0x7fceb400e2f0, req=0x7fced4f093c8) at dlg_handlers.c:368 #2 dlg_onroute (req=0x7fced4f093c8, route_params=<optimized out>, param=<optimized out>) at dlg_handlers.c:1354 #3 0x00007fceb10ab0e2 in run_rr_callbacks (req=req@entry=0x7fced4f093c8, rr_param=rr_param@entry=0x7fceb12b77a0) at rr_cb.c:96 #4 0x00007fceb10952c5 in after_loose (_m=0x7fced4f093c8, preloaded=0) at loose.c:919 #5 0x000000000042b618 in do_action (h=h@entry=0x7ffeb0b3ed80, a=a@entry=0x7fced4cb4338, msg=msg@entry=0x7fced4f093c8) at action.c:1060 #6 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b3ed80, a=0x7fced4cb4338, msg=0x7fced4f093c8) at action.c:1549 #7 0x0000000000437544 in run_actions_safe (h=h@entry=0x7ffeb0b402b0, a=<optimized out>, msg=<optimized out>) at action.c:1614 #8 0x000000000053b2e8 in rval_get_int (h=0x7ffeb0b402b0, msg=<optimized out>, i=0x7ffeb0b3f1e0, rv=rv@entry=0x7fced4cb4d58, cache=cache@entry=0x0) at rvalue.c:912 #9 0x000000000054261c in rval_expr_eval_int (h=h@entry=0x7ffeb0b402b0, msg=msg@entry=0x7fced4f093c8, res=res@entry=0x7ffeb0b3f1e0, rve=rve@entry=0x7fced4cb4d50) at rvalue.c:1910 #10 0x000000000042bc91 in do_action (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4cb8f88, msg=msg@entry=0x7fced4f093c8) at action.c:1030 #11 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b402b0, a=0x7fced4cb8f88, msg=msg@entry=0x7fced4f093c8) at action.c:1549 #12 0x000000000042bcf2 in do_action (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4cb91e8, msg=msg@entry=0x7fced4f093c8) at action.c:1049 #13 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b402b0, a=0x7fced4cb3f48, msg=msg@entry=0x7fced4f093c8) at action.c:1549 #14 0x000000000042bde0 in do_action (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4ac3d70, msg=msg@entry=0x7fced4f093c8) at action.c:678 #15 0x000000000042a10a in run_actions (h=h@entry=0x7ffeb0b402b0, a=a@entry=0x7fced4ac1698, msg=msg@entry=0x7fced4f093c8) at action.c:1549 #16 0x00000000004375d0 in run_top_route (a=0x7fced4ac1698, msg=msg@entry=0x7fced4f093c8, c=c@entry=0x0) at action.c:1635 #17 0x0000000000504386 in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:240 #18 0x00000000005f5bd4 in udp_rcv_loop () at udp_server.c:495 #19 0x00000000004b2625 in main_loop () at main.c:1600 #20 0x0000000000427e2b in main (argc=<optimized out>, argv=<optimized out>) at main.c:2616 And from syslog the relevant messages before this dump: Jul 15 08:55:03 server /usr/sbin/kamailio[16470]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param 'd4c.26d1' [3149:7522] Jul 15 08:56:01 server /usr/sbin/kamailio[16481]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param 'fc.99f1' [207:8089] Jul 15 08:56:27 server /usr/sbin/kamailio[16470]: CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7fceb3f7d920 tl->next=(nil) tl->prev=(nil) Jul 15 08:56:27 server /usr/sbin/kamailio[16470]: ERROR: dialog [dlg_handlers.c:1377]: dlg_onroute(): failed to update dialog lifetime Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Duplicate entry '9584-3854-435' for key 'hash_index' (1062) Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [INSERT INTO `dialog_extra` (`h_i] Jul 15 08:57:01 server /usr/sbin/kamailio[16482]: ERROR: auth [api.c:119]: auth_check_hdr_md5(): auth:pre_auth: Credentials are not filled properly Jul 15 08:57:01 server /usr/sbin/kamailio[16483]: ERROR: auth [api.c:119]: auth_check_hdr_md5(): auth:pre_auth: Credentials are not filled properly Jul 15 08:57:54 server /usr/sbin/kamailio[16506]: NOTICE: dialog [dlg_hash.c:245]: dlg_clean_run(): dialog in delete state is too old (0x7fceb3f64470 ref 4) Jul 15 08:57:54 server /usr/sbin/kamailio[16473]: WARNING: dialog [dlg_handlers.c:1348]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7fceb3f64470 [1182:5803] with clid '09ad128753e2535d24bde58e3d7eda04@192.168.10.232:5060' and tags 'as1b497b34' '5788890C-EC6F55F-3E86ED0C' Jul 15 08:57:54 server /usr/sbin/kamailio[16469]: ERROR: dialog [dlg_handlers.c:334]: dlg_terminated_confirmed(): failed to get dialog from params! Jul 15 08:58:49 server /usr/sbin/kamailio[16467]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param '6d2.2581' [726:6226] Jul 15 08:59:24 server /usr/sbin/kamailio[16506]: NOTICE: dialog [dlg_hash.c:245]: dlg_clean_run(): dialog in delete state is too old (0x7fceb400e2f0 ref 4) Jul 15 08:59:25 server /usr/sbin/kamailio[16464]: WARNING: dialog [dlg_handlers.c:1219]: dlg_onroute(): unable to find dialog for BYE with route param '3e4.b5c1' [1251:7259] Jul 15 08:59:25 server /usr/sbin/kamailio[16465]: WARNING: dialog [dlg_handlers.c:1348]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7fceb400e2f0 [1251:7259] with clid '87791a#015#012Call-ID: 25750e286a5654361ef9405d72edbc' and tags '' 'as148f41b1' Jul 15 08:59:25 server kernel: [203670.830521] kamailio[16465] general protection ip:7fcead34b3a5 sp:7ffeb0b3e220 error:0 in dialog.so[7fcead330000+88000] Jul 15 08:59:26 server /usr/sbin/kamailio[16511]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 33 Jul 15 08:59:26 server /usr/sbin/kamailio[16458]: ALERT: <core> [main.c:739]: handle_sigs(): child process 16465 exited by a signal 11 Jul 15 08:59:26 server /usr/sbin/kamailio[16458]: ALERT: <core> [main.c:742]: handle_sigs(): core was generated Jul 15 08:59:26 server /usr/sbin/kamailio[16458]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Any insight would be appreciated! Cheers, Dirk

2 12

CANCEL message within transaction not being relayed
by Ali Soltani 26 Jul '16

26 Jul '16

Hi, We tested the following on Kamailio 3.x and 4.x and observed the same behavior. INVITE and CANCEL are sent to Kamailo from the same source within the same transaction. (branch value in the first via header are the same in both INVITE and CANCEL message. Also FROM and TO header are same as well). However, Kamailio does not relay CANCEL message. UAC keep sending CANCEL till timeout. Apparently, t_check_trans() returns false. Can anyone help me on this issue on why Kamailio does not relay CANCEL message? Thanks, AS INVITE received by Kamailio INVITE sip:1111111@10.10.10.10:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bKkjhg4n100gagvkoup741.1 Max-Forwards: 69 From: <sip:2222222@20.20.20.20>;tag=SDl9eu701-1c1026774897 To: <sip:1111111@10.10.10.10;user=phone> Call-ID: SDl9eu701-36d75c874d3f463f6c83c5c94cc98b77-8q7ug02 CSeq: 1 INVITE Contact: <sip:2222222@20.20.20.20:5060;transport=udp> Supported: em,timer,replaces,path,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: CS-CustomerName/v.6.60A.274 Content-Type: application/sdp Content-Length: 237 P-Asserted-Identity: tel:2222222 P-Asserted-Identity: sip:2222222@192.168.0.1 after 70 msec CANCEL received by Kamailio in the same transaction CANCEL sip:1111111@10.10.10.10:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bKkjhg4n100gagvkoup741.1 CSeq: 1 CANCEL Max-Forwards: 69 From: <sip:2222222@20.20.20.20>;tag=SDl9eu701-1c1026774897 To: <sip:1111111@10.10.10.10.;user=phone> Call-ID: SDl9eu701-36d75c874d3f463f6c83c5c94cc98b77-8q7ug02 Content-Length: 0 Reason: Q.850 ;cause=16

2 1

music on hold configuration
by Gaurav Bmotra 25 Jul '16

25 Jul '16

hi i m a new user of kamailio i have sertup kamailio SIP to SIP call going well but now i m tring to impliment Music On Hold i m this http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#table.parms but unable to impliment plz help me Hare i attache my kamailio config file -- *Regards:* Gaurav Bmotra Smith Infotech E-46, phase 8 , Indus Area S.A.S Nagar Mohali 160071

3 18

Kamailio+tls not working with Ubuntu 16.04
by Yuriy Gorlichenko 25 Jul '16

25 Jul '16

Just asking about any progress of this staff THere is a bug descried at the ubuntu bug tracker https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992 There is bug that i wrote at the kamailio bug tracker https://github.com/kamailio/kamailio/issues/714 Closed it because It is not trouble of kamailio itself. I suppose something wrong at the ubuntu enviroment Any ideas or may be someone resolves it?

2 3

Re: [SR-Users] Tsilo Tiimers
by Federico Cabiddu 25 Jul '16

25 Jul '16

Hi, looks like in your script, in the SENDPUSH route, you set the luaret var to 0 then you check if it's value is != 1, in which case you reply 501, Unknown Destination. Since you never change the var value you are always hitting this case. Regards, Federico On 24 Jul 2016 14:25, "Nahum Nir" <hello.shalom.hi(a)gmail.com> wrote: > Hello Fredrico, > > Please see the attached cfg. I tried to add the fragments you published > and to add a suspension at SENDPUSH but I am getting Desitnation Unknow. > > Thank you so much, > Nir > > On Thu, Jul 21, 2016 at 10:03 PM, Federico Cabiddu < > federico.cabiddu(a)gmail.com> wrote: > >> Hi Nir, >> all you should do upon receiving B's REGISTER, after saving the contact >> (by calling save function) is either resume the transaction, perform a >> lookup and relay or call ts_append (if you already relayed the INVITE). If >> you get "unavailable" (I guess that this means that you are receiving a >> 404) probably the user is not being found in the location. Check if you are >> doing these steps in your config and if it doesn't work try to send the >> relevant part of the config and the kamailio's logs. I'm currently away and >> almost without network but I'll try to have a look as soon as possible. >> >> Regards, >> >> Federico >> >> On 21 Jul 2016 16:09, "Nahum Nir" <hello.shalom.hi(a)gmail.com> wrote: >> >> Thanks, >> >> I was able to suspend the transaction and wait for the register but the >> client that sends the INVITE get unavailable right after the other side >> joins. >> Here is my debug flow: >> Client A sends the INVITES. >> >> Use case 1: Client B register one time. Then I turn off the WiFi in >> client B so it will not recieve the push notification. Client A sends the >> INVITE, server sends the PUSH and Client A is waiting for 5 seconds. >> >> Use case 3: Same as before but now client B get the push. This time >> client A that sends the INVITE gets from the server unavailable as soon as >> client B register. >> >> I did as you suggested and suspend the trunsaction before sending the >> push. Probably a second suspend is needed when client B sends the REGISTER? >> >> Thanks and regards, >> Nir >> >> On Wed, Jul 20, 2016 at 4:06 PM, Federico Cabiddu < >> federico.cabiddu(a)gmail.com> wrote: >> >>> Hi, >>> to set the timeout for a specific transaction you can use tm module >>> function t_set_fr ( >>> http://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_set_fr >>> ). >>> For example you could call it before suspending the transaction setting >>> a short timeout and then call it again in the reply route when receiving >>> the first provisional response from the client. >>> >>> Regards, >>> >>> Federico >>> On 20 Jul 2016 2:50 pm, "Nahum Nir" <hello.shalom.hi(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> How can I chage the timer waiting for the client to send the register. >>>> i.e. how much time to wait for the push? >>>> >>>> Thanks, >>>> Nir >>>> >>>> >>>> _______________________________________________ >>>> 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 >>>> >>>> >>> _______________________________________________ >>> 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 >>> >>> >> >> >

1 0

How to forward REGISTERs to Asterisk in kamailio with dispatcher module with several Asterisk PBXs?
by Tickling Contest 24 Jul '16

24 Jul '16

Hello, When using Kamailio with the dispatcher module for Asterisk load balancing, I am offloading REGISTERs to Kamailio (works fine), but would like to forward the REGISTERs to Asterisk so that outgoing calls from Asterisk will be possible. I know this is not strictly necessary (e.g., just use Kamailio to forward everything), but I have a lot of application logic behind my Asterisk boxes (using ARI) and so I don't want that to break. Thereby, I do need this information in Asterisk as soon as possible from Kamailio. Let's say I have three boxes Asterisk_1, Asterisk_2 and Asterisk_3 in the dispatcher module like so: 1 sip:192.168.1.201:5060;transport=tcp 1 sip:192.168.1.202:5060;transport=tcp 1 sip:192.168.1.203:5060;transport=tcp My system has endpoints as numerical extensions, from 101 to 110. When a new endpoint registers with kamailio (and authenticates), I would like to send the REGISTER to Asterisk. But I have the following questions: (a) Which Asterisk? I could send it to a round-robinned Asterisk (say, Asterisk_2), but that Asterisk may not be the one responsible for handling calls outbound for the endpoint that just registered. Even if I use "hash over auth username" instead of round-robin (In http://www.kamailio.org/docs/modules/4.0.x/modules/dispatcher.html#idp16940…, choose 5 for "alg", for example) that wont solve this problem as an outbound request may come from another Asterisk box which won't have the requisite registration information. Should I forward REGISTERs to all Asterisk boxes? Maybe I won't have to deal with this issue as I use realtime and so the REGISTER information, sent to one Asterisk box, is available to all Asterisk boxes after the write to the realtime DB. Is my understanding correct? (b) How do I do this in kamailio.cfg? With some examples on the Internet for WITH_ASTERISK directive, I see that you do route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport) + ";transport=tcp"; $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } What is the equivalent for when you are using the dispatcher module and the kamailio.bindip and asterisk.bindip etc. are not set in the configuration, but instead should be procured from elsewhere? Any help is appreciated.

3 9

SIP Outbound / t_next_contacts()
by Colin Morelli 24 Jul '16

24 Jul '16

Hey all, I've got two layers of Kamailio proxies running. One set of edge proxies that are parking outbound connections and doing load balancing to a set of registrar/proxies. This is working well with one exception: I can't seem to get t_load_contacts/t_next_contacts working correctly for handling multiple outbound connections. It *appears* to work (though I haven't extensively tested) whenever there's more than one registration binding for an AOR. However, when there's only one, I get: 8(27) DEBUG: tm [t_serial.c:191]: t_load_contacts(): nr_branches is 0 8(27) DEBUG: tm [t_serial.c:194]: t_load_contacts(): nothing to do - only one contact! 8(27) DEBUG: tm [t_serial.c:455]: t_next_contacts(): no contacts in contacts_avp - we are done! 8(27) ERROR: <script>: t_next_contacts() failed I tried omitting the call to t_next_contacts(), thinking it should possibly only be used in failure_route. But when I do that (while leaving t_load_contacts), I get: 5(24) ERROR: tm [t_fwd.c:1730]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding 5(24) WARNING: <core> [receive.c:258]: receive_msg(): error while trying script The relevant part of the config is: if (!lookup("location")) { $var(rc) = $rc; switch ($var(rc)) { case -1: xlog("L_INFO", "$ci|log|lookup of contacts for $ru returned no results\r\n"); send_reply("404", "Not Found"); exit; case -3: xlog("L_INFO", "$ci|log|lookup of contacts for $ru failed with code $var(rc)\r\n"); send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } if (!t_load_contacts() || !t_next_contacts()) { xlog("L_ERR", "t_load_contacts() failed\n"); sl_reply_error(); t_release(); exit; } Am I missing something else here? Note: it works fine if I remove t_load_contacts/t_next_contacts, but then I don't get proper handling of multiple sip.instance bindings/differing q values. Best, Colin

2 4

business case
by anfecora 24 Jul '16

24 Jul '16

I apologize since this is a technical mail list and this question may not belong here but i need to fill a business requirement and need more information of big companies using kamailio, i am searching over the internet but if anyone can provide me with names of real carriers using kamailio will be very useful. Thanks again for any help in this case.

2 1

Dispatcher list not loading from dispatcher.list, but loading from DB
by Tickling Contest 23 Jul '16

23 Jul '16

Hello, I have a Kamailio 4.0.4 configured with the cfg file shown in https://gist.github.com/ticklingcontest/f96aa4d08b08dc4643056d73da24f0a5 The modparam for dispatcher sets the dispatcher file to be in /var/run/kamailio/dispatcher.list. However, I recently discovered that $ kamctl dispatcher show picks up the information from the kamailio DB instead of the dispatcher.list. What's going on? Any insight is appreciated. Thanks!

1 0

kamailio starts looping invites(some) back to itself.
by Aqs Younas 22 Jul '16

22 Jul '16

Hi, I am using http_async_client and dispatcher modules of Kamailio. Here is scenario! http_async_client makes calls to an API to get groupid upon which dispatcher sends calls. I am using SIPp to generate calls with 100 cps. I see some calls being failed with an error cause, 483 Too Many Hops. Taking the trace shows Kamailio routing some calls to itself. I think this is not a configuration issue since all calls are same(follow same logic) and some calls get failed. Any pointer is much appreciated. Best Regards.

1 0

sip options not relaying
by gerry kernan 22 Jul '16

22 Jul '16

Hi I have a kamailio proxy in front of asterisk ver 13 servers. I'm using path support to route to the relevant asterisk server depending on the domain in the sip header. The incoming registers & invites from sip endpoints get routed to the correct asterisk server , but when asterisk is sending sip options to the end points my kamailio config is not routing the option on to the endpoint . Kamailio debug shows error below ERROR: <script>: 192.168.200.131 OPTIONS sip:1333@192.168.200.112:5062 -- domain "192.168.200.112" is not found in domain table 192.168.200.131 Am I correct in thinking that this SIP option should be getting routed via my on-reply route. the sip option looks like it is failing here at the start of my config if (src_ip != BACKEND_NET4) { xlog("L_NOTICE", "FROM CLIENT IP: $si $rm $ru $td \n"); # SIP request packet client->backend if( !loose_route() ) { if (!lookup_domain("$td", "dattr_")) { xlog("L_NOTOCE" , "Domain lookup, $rm "); xlog("L_ERR", "$si $rm $ru -- domain \"$td\" is not " + "found in domain table $si\n"); drop(); } Sample of SIP option captured on the kamailio server OPTIONS sip:1333@192.168.200.112:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.131:5060;branch=z9hG4bK4f910445 Max-Forwards: 70 Route: <sip:192.168.200.148;lr;received=sip:87.198.223.83:16070> From: "Unknown" <sip:Unknown@192.168.200.131>;tag=as54cfb9df To: <sip:1333@192.168.200.112:5062> Contact: <sip:Unknown@192.168.200.131:5060> Call-ID: 257dea110f5255275aa38d9c45e0e235@192.168.200.131:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(13.10.0) Date: Fri, 22 Jul 2016 15:31:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer, path Content-Length: 0 OPTIONS sip:1333@192.168.200.112:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.131:5060;branch=z9hG4bK20611e3b Max-Forwards: 70 Route: <sip:192.168.200.148;lr;received=sip:87.198.223.83:16070> From: "Unknown" <sip:Unknown@192.168.200.131>;tag=as1b6a0faf To: <sip:1333@192.168.200.112:5062> Contact: <sip:Unknown@192.168.200.131:5060> Call-ID: 7f8ed5ea7a147f2b61269df862af7fae@192.168.200.131:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(13.10.0) Date: Fri, 22 Jul 2016 15:31:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer, path Content-Length: 0 Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

1 1

Kamailio + Asterisk MWI Issues
by TEG AMJG 21 Jul '16

21 Jul '16

Dear list I am quite new to Kamailio and i know that probably this thread has been answered quite few times. But i have few questions I was able to configure Kamailio + Asterisk MWI with Kamailio handling the registration of users. The problem is that i am quite sure that is not the right way. What i did was just putting "subscribemwi=no" on every user declared on Asterisk and he will send NOTIFY's when a voicemail has been received for a particular user to Kamailio and Kamailio will just forward them to the endpoints (because he knows the location of the users). I know about the modules presence presence_xml and presence_mwi of Kamailio which will handle the subscription of users and send NOTIFY's but it is useless since I am not verifying any NOTIFY that comes from Asterisk, just forwarding. Now, some others solution are just a bunch of scripts that, for me, it doesnt make much sense since is the same idea like how i did it just more hacky and unnecesary, i guess?. This solution can be found here: http://saevolgo.blogspot.com/2012/07/asterisk-behind-kamailio-voicemail-mwi… There is another solution that i found in few threads here which i dont understand really well, is to send PUBLISH messages from Asterisk to Kamailio so Kamailio sends NOTIFY's with the event and content about the amount of voice messages in a particular mailbox. Some say that you need to create the message PUBLISH and send it with sipsak or another solution in Asterisk. For what i see in SIP documentation, the last solution seems a bit less of a hack lets say, more resilient. With all said, i have a few questions Is it the solution I made really a bad idea?. I havent take a lot of time understanding SUBSCRIBE/NOTIFY messages, just the basics, i dont know if there are some security issues i should think about in this particular flow. Should i check the status of the authorization of the subscription in Kamailio before forwarding the mesage (if there is a way, how)? or it really doesnt make sense at all? because after all, my endpoints receives the NOTIFY and "you have a new voicemail" message appears. Now, about the last solution, is there any guide of someone that has done that in a clean way?, may someone guide me how to construct the PUBLISH message and when should i send it?, should i create a cron or something?, this PUBLISH message needs to be send for every user?. may someone guide me (if this is considered a cleaner way of doing it) how to make this? Many of the answers in other threads about this are not enough to understand the idea as a whole, at least for me. Thank you very much in advance Alejandro -- - Saludos a todos

1 0

Kamailio stops accepting calls for few seconds after 80 CPS.
by Aqs Younas 21 Jul '16

21 Jul '16

Greeting list. I am doing the load test of Kamailio with SIPp using most basic script(kamailio.cfg) with logging level set to minimum to dispatch requests to SIPp UAS. This is my dispatcher code snippet. $avp(quality) = "1"; ds_select_dst("$avp(quality)", "4"); route(RELAY); At 70 CPS calls work fine. But if I increase CPS to 80, Kamailio stops accepting calls for few seconds and comes to life after being idle for some time. 100 CPS for sip proxy is not hot cake. Also, I see in http://www.kamailio.org/docs/openser-performance-tests/ tests were made with -l27 and -m7000 which looks wronged to me. ./sipp -sf uac_inv.xml <http://www.kamailio.org/docs/openser-performance-tests/data/uac_inv.xml> -rsa 192.168.2.102:5060 192.168.2.102:5070 -m 200000 -r 7000 -d 1 -l 27 There must be something wrong in my configuration since this cps is very low. kamailio -version version: kamailio 4.4.1 (x86_64/linux) 90be8b 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, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 90be8b SHM_MEMORY=768 PKG_MEMORY=64 Any suggestion is much appreciated. Best Regards,

2 2

Issue with ftag parameter in Record-Route header
by Valentin Christoph 21 Jul '16

21 Jul '16

Hi community, I'm using an MGCF that creates From tags like **%%!!MYOFFICE++00!!%%**A0F1F1BA Now the special characters "%%" are allowed in tags (i.e. "token" according to RFC 3261), token = 1*(alphanum / "-" / "." / "!" / "%" / "*" / "_" / "+" / "`" / "'" / "~" ) but they are NOT allowed in URI parameters according to RFC 3261 uri-parameter = transport-param / user-param / method-param / ttl-param / maddr-param / lr-param / other-param other-param = pname [ "=" pvalue ] pname = 1*paramchar pvalue = 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = "[" / "]" / "/" / ":" / "&" / "+" / "$" unreserved = alphanum / mark mark = "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")" escaped = "%" HEXDIG HEXDIG which makes the MGCF ignore the whole RR header and sending subsequent requests directly to the peer SIP UA, instead of sending it to the CSCF. Hence it's not OK to just copy the From tag into the ftag parameter, imho. We did a workaround in the MGCF, now sending **~~!!MYOFFICE++00!!~~**A0F1F1BA, but strictly spoken the kamailio S-CSCF is not compliant to RFC 3261. What's your opinion? Is there an configuration option to make it compliant? All the best Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

Tsilo Tiimers
by Nahum Nir 21 Jul '16

21 Jul '16

Hello, How can I chage the timer waiting for the client to send the register. i.e. how much time to wait for the push? Thanks, Nir

2 2

Trouble with Asterisk behind Kamailio
by Tickling Contest 21 Jul '16

21 Jul '16

Hello, I am following the instructions in http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb to configure my Kamailio and Asterisk (using PJSIP realtime with MySQL). I am able to get Kamailio register the clients (tested with Zoiper softphone), but having trouble getting any further to load balance a couple of Asterisk servers. Here is a range of issues I have: (a) The Kamailio server keeps sending UDP SIP messages to the Asterisk server and it is not clear where to control what protocol to use to send SIP messages Asterisk server(s). (b) I followed the instructions for the dispatcher module in the wiki, but it is not clear why I should use the #!define WITH_ASTERISK directive to enumerate the bind IPs and ports (how does this reconcile with the list of Asterisk servers in the dispatcher.list file?): !ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.102" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.1.101" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" Also, where do I specify the protocol to use when sending to the Asterisk servers? Is it auto detected? If so, why is Kamailio not detecting that the Asterisk servers are only listening on TCP? Any help is appreciated. My entire /etc/kamailio/kamailio.cfg file is in here: https://gist.github.com/ticklingcontest/c7437dbc2d8e838834ee4d852753f254. I am using Kamailio 4.0.4 with Asterisk 13.7.2 using TCP for PJSIP. Thanks!

3 3

Kamailio Crashes when dispatcher failover is being applied
by TEG AMJG 21 Jul '16

21 Jul '16

Dear list Im quite new to Kamailio and i found something while trying to see the failover behaviour in Kamailio. I have a kamailio v4.3.5 load balancing two Asterisk boxes. This happens just when i stop one of the asterisk and make a call that is being load balanced to the same asterisk box that i stopped before, when trying to go to failure_route and dispatch it to the other box then Kamailio just crashes. I have debug=4 in the config file and it really doesnt say much about the error. This is the relative logging i could found: *Jul 18 16:30:13 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Forwarding non-registrar requests to AsteriskJul 18 16:30:13 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Dispatching to Asterisk Box sip:10.0.1.201:5060 <http://10.0.1.201:5060>Jul 18 16:30:13 kam2 /usr/sbin/kamailio[69550]: INFO: <script>: UBICAR: requestm=INVITE reply=<null> SIP packet comes from the Asterisk Box duid=PBX2;my=pbx2;maxload=10000 callid=4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060 <http://4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060>Jul 18 16:30:13 kam2 /usr/sbin/kamailio[69550]: INFO: <script>: UBICAR: requestm=INVITE reply=<null> SIP packet comes from the Asterisk Box duid=PBX2;my=pbx2;maxload=10000 callid=4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060 <http://4d52095c6f2ed1ab2de2d6c70340a126@10.0.1.201:5060>Jul 18 16:30:18 kam2 /usr/sbin/kamailio[69547]: INFO: <script>: UBICAR: setting call load requesturi=<null> callid=896175650(a)192.168.14.76 <896175650(a)192.168.14.76>Jul 18 16:30:20 kam2 /usr/sbin/kamailio[69550]: INFO: <script>: UBICAR: delete call load requesturi=sip:10005@10.0.1.201:5060 <http://sip:10005@10.0.1.201:5060> callid=896175650(a)192.168.14.76 <896175650(a)192.168.14.76>Jul 18 16:30:38 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Forwarding non-registrar requests to AsteriskJul 18 16:30:38 kam2 /usr/sbin/kamailio[69548]: INFO: <script>: UBICAR: Dispatching to Asterisk Box sip:10.0.1.201:5060 <http://10.0.1.201:5060>Jul 18 16:30:42 kam2 /usr/sbin/kamailio[69559]: ALERT: <script>: UBICAR: Asterisk sip:10.0.1.201:5060 <http://10.0.1.201:5060> failed to answer, selecting other one!Jul 18 16:30:42 kam2 /usr/sbin/kamailio[69559]: INFO: <script>: UBICAR: Dispatching to Asterisk Box sip:10.0.1.200:5060 <http://10.0.1.200:5060>Jul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: ALERT: <core> [main.c:728]: handle_sigs(): child process 69559 exited by a signal 11Jul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: ALERT: <core> [main.c:731]: handle_sigs(): core was generatedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLDJul 18 16:30:43 kam2 /usr/sbin/kamailio[69563]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69562]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69561]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69558]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69557]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69556]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69555]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69554]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69553]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69552]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69551]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69548]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69547]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69549]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69550]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69560]: INFO: <core> [main.c:798]: sig_usr(): signal 15 receivedJul 18 16:30:43 kam2 /usr/sbin/kamailio[69546]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized* My configuration when applying load balancing and failover is detailed here: *# Send to Asteriskroute[TOASTERISK] {#!ifdef WITH_LOADBALANCE xlog("L_INFO","UBICAR: Forwarding non-registrar requests to Asterisk \n"); if(!ds_select_dst("0", "10")) { sl_send_reply("500", "Service Unavailable"); xlog("L_ERROR","UBICAR: No destinations available for $rd \n"); exit; } xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n"); t_set_fr(0,4000); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit;#!endif* *# manage failure routing casesfailure_route[MANAGE_FAILURE] { #xlog("L_ALERT","UBICAR: failure requestm=$rm requesturi=$ru callid=$ci\n"); if (t_is_canceled()) { exit; }#!ifdef WITH_LOADBALANCE xlog("L_ALERT","UBICAR: Asterisk $du failed to answer, selecting other one!"); # next DST - only for 500 reply or local timeout (set by t_set_fr()) if (t_check_status("500") || t_branch_timeout() || !t_branch_replied()) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); #select the new destination if(ds_next_dst()) { t_set_fr(0,4000); t_on_failure("MANAGE_FAILURE"); xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n"); route(RELAY); exit; }else{ #last available node failed to reply, no other destinations available send_reply("404", "No destination"); exit; }#!endif }}* It may be a bug in that Kamailio version because i used Kamailio v4.2 before and i didnt have that inconvenient, still i had to upgrade to v4.3 because i needed the default_socket modparam that wasnt implemented in Kamailio v4.2 Any hint about this issue would be appreciate it Best Regards Alejandro

2 1

Managing concurrent ports
by Grant Bagdasarian 21 Jul '16

21 Jul '16

Hello, Are there any recommended modules for managing concurrent ports for different endpoints/customers/accounts/etc? I believe the dialog and dialog_ng modules do exactly this, but how is the performance of these modules under high loads? What is the release time of the ports? Are there perhaps other modules? Regards, Grant

3 6

Kamailio + rtpproxy + SRTP
by Konstantin M. 20 Jul '16

20 Jul '16

I have some problem with SSL+rtpproxy, kamailio successfully decodes SIP, rtpproxy decodes SRTP frames, all is going to freeswitch unencrypted. FS transfers the call back, the call is going unencrypted though kamailio, kamailio encodes to SSL, but rtpproxy DOESN'T encodes RTP frames to SRTP. Hence, endpoint SIP equipment rejects the call with 488 Not Acceptable Here Scenario 1. Endpoint device -> SRTP -> kamailio -> RTP -> freeswitch 2. Freeswitch calls back 3. freeswitch -> RTP -> kamailio -> PROBLEM (SIP[TLS] + RTP (but not SRTP)) -> Endpoint device 4. Endpoint device -> 488 Not Acceptable Here -> Kamailio Could somebody point me what's wrong here? Thanks!

2 1

WEBRTC Confusion (2)
by Zaka 20 Jul '16

20 Jul '16

Dear List: I am still stuck. Still unable to get Kamailio to listen on WEBRTC port* (in this case 8000)* Please advise if it has something to do with disabling TLS/MSRP? Please see the log & related config file in line here: *Log:* 0(5724) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_PORT!8000!g *and the result of config file check:* root@callcntr:/usr/local/etc/kamailio# kamailio -eE -ddd -cf /usr/local/etc/kamailio/websocket.cfg 0(5902) INFO: <core> [main.c:1911]: main(): private (per process) memory: 8388608 bytes 0(5902) DEBUG: <core> [mem/f_malloc.c:323]: fm_malloc_init(): fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(5902) DEBUG: <core> [mem/f_malloc.c:325]: fm_malloc_init(): fm_malloc_init: F_HASH_SIZE=2099, fm_block size=33912 0(5902) DEBUG: <core> [mem/f_malloc.c:327]: fm_malloc_init(): fm_malloc_init(0x7f9ca6e1f010, 8388608), start=0x7f9ca6e1f010 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac2700)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac2680)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26c0)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac2720)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26a0)[0] to 0 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26e0)[0] to 0 0(5902) DEBUG: <core> [pvapi.c:1976]: pv_init_buffer(): PV print buffer initialized to [10][8192] 0(5902) DEBUG: <core> [main.c:2097]: main(): read 286773155 from /dev/urandom 0(5902) DEBUG: <core> [main.c:2103]: main(): seeding PRNG with 1755363211 0(5902) DEBUG: <core> [main.c:2107]: main(): test random numbers 726213135 321707966 991183401 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e3fc48 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: DBURL 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [DBURL] value [mysql://kamailio:kamailiorw@localhost/kamailio] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [DBURL]=[mysql://kamailio:kamailiorw@localhost/kamailio] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e3ffe8 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_IP_ADDR!192.168.1.16!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_IP_ADDR 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_IP_ADDR] value [192.168.1.16] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_IP_ADDR]=[192.168.1.16] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40320 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_DOMAIN!callcntr.com.al!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_DOMAIN 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_DOMAIN] value [callcntr.com.al] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_DOMAIN]=[callcntr.com.al] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40658 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_PORT!8000!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_WS_PORT 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_WS_PORT] value [8000] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_WS_PORT]=[8000] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40988 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WSS_PORT!443!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_WSS_PORT 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_WSS_PORT] value [443] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_WSS_PORT]=[443] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e40cb8 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_MSRP_PORT!9000!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_MSRP_PORT 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_MSRP_PORT] value [9000] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_MSRP_PORT]=[9000] (0) 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=0 0(5902) DEBUG: <core> [re.c:455]: subst_run(): matched (16, 10): [MY_IP_ADDR] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [ppcfg.c:192]: pp_subst_run(): preprocess subst applied [#1] to [!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g] - returning new string [!MY_WS_ADDR!tcp:192.168.1.16:MY_WS_PORT!g] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=0 0(5902) DEBUG: <core> [re.c:455]: subst_run(): matched (29, 10): [MY_WS_PORT] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [ppcfg.c:192]: pp_subst_run(): preprocess subst applied [#2] to [!MY_WS_ADDR!tcp:192.168.1.16:MY_WS_PORT!g] - returning new string [!MY_WS_ADDR!tcp:192.168.1.16:8000!g] 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:436]: subst_run(): running. r=1 0(5902) DEBUG: <core> [re.c:505]: subst_str(): no match 0(5902) DEBUG: <core> [re.c:313]: subst_parser(): ok, se is 0x7f9ca6e41118 0(5902) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_ADDR!tcp:192.168.1.16:8000!g 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: MY_WS_ADDR 0(5902) DEBUG: <core> [cfg.lex:1776]: pp_define_set(): ### setting define ID [MY_WS_ADDR] value [tcp:192.168.1.16:8000] 0(5902) DEBUG: <core> [ppcfg.c:160]: pp_substdef_add(): ### added substdef: [MY_WS_ADDR]=[tcp:192.168.1.16:8000] (0) 0(5902) DEBUG: <core> [cfg.lex:1690]: pp_define(): defining id: WITH_WEBSOCKETS 0(5902) DEBUG: <core> [cfg.lex:1795]: pp_define_get(): ### returning define ID [MY_IP_ADDR] value [192.168.1.16] 0(5902) DEBUG: <core> [cfg.lex:1795]: pp_define_get(): ### returning define ID [MY_IP_ADDR] value [192.168.1.16] loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(5902) DEBUG: <core> [ppcfg.c:223]: pp_ifdef_level_check(): same number of pairing preprocessor directives #!IF[N]DEF - #!ENDIF 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 0: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "REQINIT" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:421]: print_action(): force_rport( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<0> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50ff0> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_ERR" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "400" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "WITHINDLG" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42f78> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e43950> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "AUTH" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4c350> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:543]: print_action(): , type<22> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44968> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "REGISTRAR" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "484" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "LOCATION" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "RELAY" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 1: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44460> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 2: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "483" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d7d0> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "Malformed SIP message from $si:$sp " 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4cdb0> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 3: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_ERR" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "400" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:347]: print_action(): route( 0(5902) DEBUG: <core> [route_struct.c:452]: print_action(): "RELAY" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42f78> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "404" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 4: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44460> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 5: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42a70> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e44428> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "404" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:442]: print_action(): assign( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<21> 0(5902) DEBUG: <core> [route_struct.c:517]: print_action(): , type<22> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e42a70> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:448]: print_action(): UNKNOWN( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:517]: print_action(): , type<24> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): routing table 6: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4e8b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "$fd" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4e810> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e443b8> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): onreply routing table 0: 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 0 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 9 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50e68> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 0 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): event routing table 1: 0(5902) DEBUG: <core> [route_struct.c:448]: print_action(): UNKNOWN( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<0> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:448]: print_action(): UNKNOWN( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<0> 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50660> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 4 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_DBG" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_WARN" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50660> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 4 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "403" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:401]: print_action(): if ( 0(5902) DEBUG: <core> [route_struct.c:483]: print_action(): type<22> 0(5902) DEBUG: <core> [route_struct.c:485]: print_action(): ) { 0(5902) DEBUG: <core> [route_struct.c:338]: print_action(): drop( 0(5902) DEBUG: <core> [route_struct.c:455]: print_action(): 1 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 1 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:519]: print_action(): } else { 0(5902) DEBUG: <core> [route_struct.c:545]: print_action(): }; 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e50660> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 4 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "404" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:2132]: print_rl(): event routing table 2: 0(5902) DEBUG: <core> [route_struct.c:418]: print_action(): external_module_call( 0(5902) DEBUG: <core> [route_struct.c:467]: print_action(): f_ptr<0x7f9ca6e4d808> 0(5902) DEBUG: <core> [route_struct.c:493]: print_action(): , 2 0(5902) DEBUG: <core> [route_struct.c:524]: print_action(): , "L_INFO" 0(5902) DEBUG: <core> [route_struct.c:546]: print_action(): ); 0(5902) DEBUG: <core> [route.c:2134]: print_rl(): 0(5902) DEBUG: <core> [route.c:129]: route_add(): mapping routing block (0xac26e0)[core:receive-parse-error] to 3 0(5902) INFO: <core> [sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: 192.168.1.16:5060 tcp: 192.168.1.16:5060 Aliases: tcp: callcntr:5060 udp: callcntr:5060 config file ok, exiting... root@callcntr:/usr/local/etc/kamailio# *Config File:* #!KAMAILIO # # Simple/sample kamailio.cfg for running a proxy/registrar with TLS and # WebSockets support. #!substdef "!DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g" #!substdef "!MY_IP_ADDR!192.168.1.16!g" #!substdef "!MY_DOMAIN!callcntr.com.al!g" #!substdef "!MY_WS_PORT!8000!g" #!substdef "!MY_WSS_PORT!443!g" #!substdef "!MY_MSRP_PORT!9000!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" ##!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" ##!substdef "!MSRP_MIN_EXPIRES!1800!g" ##!substdef "!MSRP_MAX_EXPIRES!3600!g" ##!define LOCAL_TEST_RUN ##!define WITH_TLS #!define WITH_WEBSOCKETS ##!define WITH_MSRP ####### Global Parameters ######### fork=yes children=4 #!ifdef WITH_TLS enable_tls=1 #!endif listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_IP_ADDR #!ifdef WITH_TLS #listen=MY_WSS_ADDR #!endif #!endif #!ifdef WITH_MSRP listen=MY_MSRP_ADDR #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 #!ifdef LOCAL_TEST_RUN debug=2 mpath="modules" #!else debug=0 mpath="/usr/local/lib64/kamailio/modules/" #!endif loadmodule "db_mysql.so" loadmodule "tm.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 "auth.so" loadmodule "auth_db.so" loadmodule "kex.so" loadmodule "mi_rpc.so" loadmodule "corex.so" #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_MSRP loadmodule "msrp.so" loadmodule "htable.so" loadmodule "cfgutils.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" #!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 ####### 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 { * if (($Rp == MY_WS_PORT)) { * *#### I deleted/ modified this logical expression ####* xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } 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")) { record_route(); } # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484", "Address Incomplete"); exit; } # user location service route(LOCATION); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } if (!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } if (uri == myself && is_method("OPTIONS") && !(uri=~"sip:.*[@]+.*")) { options_reply(); 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(RELAY); } else { 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 t_relay(); exit; } else { # ACK without matching transaction... # ignore and discard exit; } } sl_send_reply("404", "Not Found"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!is_subscriber("$ru", "subscriber", "1")) { t_newtran(); send_reply("404", "Not Found"); exit; } if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: send_reply("480", "Temporarily Unavailable"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; case -3: send_reply("500", "Server Internal Error"); exit; } } } # Authentication route route[AUTH] { 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")) { 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", "Forbidden"); exit; } } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT) && !(proto == WS)) || $Rp == MY_MSRP_PORT) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($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 #!ifdef WITH_MSRP event_route[msrp:frame-in] { msrp_reply_flags("1"); if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) && $Rp != MY_MSRP_PORT) { xlog("L_WARN", "MSRP request received on $Rp\n"); msrp_reply("403", "Action-not-allowed"); exit; } if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } if (!www_authenticate("MY_DOMAIN", "subscriber", "$msrp(method)")) { if (auth_get_www_authenticate("MY_DOMAIN", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } if ($hdr(Expires) != $null) { $var(expires) = (int) $hdr(Expires); if ($var(expires) < MSRP_MIN_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Min-Expires: MSRP_MIN_EXPIRES\r\n"); exit; } else if ($var(expires) > MSRP_MAX_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Max-Expires: MSRP_MAX_EXPIRES\r\n"); exit; } } else { $var(expires) = MSRP_MAX_EXPIRES; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); $shtex(msrp=>$var(sessid)) = $var(expires) + 5; # - Use-Path: the MSRP address for server + session id $var(hdrs) = "Use-Path: msrps://MY_IP_ADDR:MY_MSRP_PORT/" + $var(sessid) + ";tcp\r\n" + "Expires: " + $var(expires) + "\r\n"; msrp_reply("200", "OK", "$var(hdrs)"); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if ($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "Session-does-not-exist"); exit; } else if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); } else { msrp_reply("501", "Request-method-not-understood"); } } #!endif Thanks a million! Zaka One accurate measurement is worth more than a thousand expert opinions!

2 4

reading siptrace trace_on current value in script
by ananda agrawal 20 Jul '16

20 Jul '16

Hi, Is it possible to check wether the trace_on for siptrace module is on or off from within the config script, e.g inside routing logic, thannks in advance -Anand

1 0

Crocodile-msrp
by Marcelo Pignataro 19 Jul '16

19 Jul '16

Hi, I don't really know if I'm writing to the right list, sorry if I don't. Has someone been using crocodile-msrp library? I need to develop a webchat app with Kamailio together and I found Crocodile, but I see it hasn't documentation neither guides to developers. I also realize it hasn't active community, last commit done to crocodile-msrp library by its developers was 2013. I need some information to know if I should use it or not. Thank you and best regards, Marcelo

1 0

Push Notifications
by Nahum Nir 18 Jul '16

18 Jul '16

Hello, Found the presentation about asynch push. Is there an example cfg file? Thanks, Nir

3 6

Custom header on reply is removed on t_relay
by David Wilson 18 Jul '16

18 Jul '16

Hello All, We have a Kamailio proxy (P1) and a custom client (A). Calls are working reliably with A communicating directly with P1. When A sends INVITE to P1, in some cases, we insert a custom header ‘X-Routing’ in the reply. Now, testing with a new, additional proxy (P2) between these two: A -> P2 -> P1 A sends INVITE via P2 to P1, P1 replies include X-Routing header field (confirmed by logging message buffer received from P1 by P2). However, the reply relayed from P2 to A does not contain the custom header. P2’s reply_route block only contains logging code and I’ve checked for calls to remove_hf. I can’t find documentation or code to explain why the header is omitted. Is there a module config parameter that I can set to preserve custom headers? Any suggestions welcome! Regards, Dave Wilson.

2 1

Rx Interface connected, get error code: SIP/2.0 403 Forbidden - You must register first with a S-CSCF
by Kristian Martens 18 Jul '16

18 Jul '16

Hi, I have installed Kamailio version 4.4.1. Everything works just fine. If I connect a PCRF via Rx to the P-CSCF, the registration still runs through as expected but the subsequent SUBSCRIBE is rejected with "403 Forbidden - You must register first with a S-CSCF" error cause or is simply ignored (see attached wireshark traces). And, if I disconnect Rx and restart the P-CSCF, the "forbidden" problem still persists. (But I can work around this by cleaning up the P-CSCF database.) Any suggestions how I could get Rx working? Thanks, Kristian

3 7

kamailio-dbg package
by Laura García Pérez 18 Jul '16

18 Jul '16

Is it possible to install kamailio-dbg in CentOS? Thanks, Laura García Pérez www.logslie.com <http://www.logslie.com/> +34680923532 +12602069698

2 2

Conditionally calling TOPOS
by Nathan Ward 18 Jul '16

18 Jul '16

Hi, I think it make sense to put this topic in a seperate message, though this is very much the same scenario mentioned here: http://lists.sip-router.org/pipermail/sr-users/2016-July/093864.html I would like the ability initiate topology hiding only for certain messages - i.e. on inbound calls, I would like my Kamailio ‘SBC' to retain the routing information for inbound requests, but strip it for outgoing requests - so that I can see the full path a call has taken to reach and then traverse my network all the way to an egress SBC but then this is all stripped before forwarding the call to a 3rd party (i.e. customer, or carrier). I note that this isn’t possible at present, but I’d like to take a stab at the code, perhaps designating ‘trusted’ IPs for which topology hiding doesn’t take place. Before I embark on this, is this something that would be considered useful, or is there a good reason to have full topology hiding that I’m missing? -- Nathan Ward

2 1

TOPOS - REGISTER
by Nathan Ward 18 Jul '16

18 Jul '16

Hi all, I’m doing some testing using Kamailio to do some lightweight SBC type functions. We have considered OpenSIPs as well as it has built in b2bua bits, but having chosen Kamailio for other parts of our network it makes sense to stick with one piece of software I think. Luckily, we only need (want) topology hiding, and don’t need other b2bua things. I have a server in the middle of our network sending registrations out to a 3rd party network for a number of lines, using Kamailio+uacreg. This works great. It sends these registrations through an SBC, also running Kamailio. I do this by setting the proxy column of the uacreg table to the SBC’s inside address. On the SBC, I rewrite the Contact header, so calls come in to the SBC. I am running topos on the SBC, and would like to be able to do topology hiding for the outgoing REGISTER. I note that there is code in tps_skip_msg to specifically avoid topology hiding for REGISTER and PUBLISH messages. Is there a SIP reason for this, or is this just a part of the code that hasn’t been finished yet or wasn’t needed by the author, so it’s been excluded? I note that topoh is the same, with a th_skip_msg function that has I think identical code, so no doubt the same reasoning is applied there. This is a very minor issue, but I’d like to understand the reasoning, and contribute code if the reason is code rather than SIP. -- Nathan Ward

2 1

RTPProxy Issues
by TEG AMJG 18 Jul '16

18 Jul '16

Dear list I am quite new to Kamailio and i have been able to solve some NAT Traversal issues with symmetric SIP+RTP putting kamailio+rtpproxy behind NAT, i am also load balancing some asterisk boxes for transcoding and some other services like voicemail. Also i am using SIPp for load testing Now the question is that, while everything is working great with NAT (even when i am using SIPp for testing with low load) when i am about to test it with more than 150 calls or something i am starting to get the following error: *kamailio[34549]: ERROR: rtpproxy [rtpproxy.c:2735]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy* I have been trying to look at some SIP packets when the error starts and i am not seeing anything out of the normal (while being quite difficult to debug when having so many concurrent calls at SIP level) My configuration doesnt have anything new, the only thing is that, because Kamailio is behind NAT i used some ideas from some other users and debates from this same mailing list: *# RTPProxy control* *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;* * if(!route(FROMASTERISK)){* * force_send_socket(10.0.1.207);* * rtpproxy_manage("rw","10.0.1.206");* * }else{* * force_send_socket(10.0.1.206);* * rtpproxy_manage("rw","10.0.5.203");* * }* * if (is_request()) {* * if (!has_totag()) {* * add_rr_param(";nat=yes");* * }* * }* * if (is_reply()) {* * if(isbflagset(FLB_NATB)) {* * fix_nated_contact();* * }* *#!endif* * return;* *}* This solution is based of someone's idea from this mailing list: http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html As you can see in my configuration, my "public network" is really private network so i tweak the NATDETECT route to not include RFC1918 networks as part of the detection *# Caller NAT detection route* *route[NATDETECT] {* *#!ifdef WITH_NAT* *if (nat_uac_test("1")) {* * if (nat_uac_test("18")) {* * force_rport();* * if (is_method("REGISTER")) {* * fix_nated_register();* * } else {* * fix_nated_contact();* * }* * setflag(FLT_NATS);* * return;* * }* * }* *#!endif* * return;* *}* Also i would like to know how may i use RTPProxy logs?, i tried to configure it like this: /etc/rsyslog.conf *# Kamailio logging* *local0.* -/var/log/kamailio.log* *#RTPProxy logging* *local3.* /var/log/rtpproxy.log* /etc/sysconfig/rtpproxy *OPTIONS="-l 10.0.1.206 -A 10.0.5.203 -s udp:127.0.0.1:7722 <http://127.0.0.1:7722> -r /var/lib/rtpproxy/sessions -d DBUG:LOG_LOCAL3 -m 35000 -M 55000 -F"* I cannot see anyhing in the logs Thank you very much Alejandro

3 3

Questions
by francisco rodriguez 15 Jul '16

15 Jul '16

Hello ; I have the sip account with you, but to put it to work in a team asks me some data. you can tell me what they are. my account is sip: masterlaradio(a)iptel.org I request the following information IP SERVER :? SERVER USER? USERNAME? STUN DOMAIN? DNS SERVER? thanks for your attention and collaboration Francisco Rodriguez Enviado desde Correo<https://go.microsoft.com/fwlink/?LinkId=550986> para Windows 10

1 0

Websocket Path Issues
by Colin Morelli 15 Jul '16

15 Jul '16

Hey all, I'm using Kamailio as an outbound edge proxy for websocket connections. When my registrar calls out to Kamailio to forward to a websocket connection that has since been killed, rather than just entering the failure branch, it throws a few errors: 5(24) WARNING: <core> [msg_translator.c:2760]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 5(24) ERROR: <core> [msg_translator.c:1978]: build_req_buf_from_sip_req(): could not create Via header 5(24) ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request 5(24) ERROR: tm [t_fwd.c:1737]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches 5(24) ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (2/SL) Ultimately this means the proxy gets a 500 rather than a 430 flow failed, and the registration sits around in the proxy when it should be removed. Am I missing something obvious to get this behavior? Best, Colin

2 2

Re: [SR-Users] Determining if request is from a local domain
by Shane Harrison 15 Jul '16

15 Jul '16

Hi all, Solved! As is often the case, after working all morning on the problem, after posting I discovered my error. I noticed that I hadn't included in my post the "alias=" command. It was indeed in my script but I had used aws-rcsstg.voip.systems instead of aws-rcsstg.vvoip.systems ! I don't know how many times I have stared at that but I guess you often see what you want to see...... Apologies for wasting your time. Kind regards Shane > -----Original Message----- > From: Shane Harrison > Sent: Wednesday, 13 July 2016 2:48 p.m. > To: 'sr-users(a)lists.sip-router.org' > Subject: Determining if request is from a local domain > > Hi all, > > From previous posts my understanding is that RURI's are matched against > reverse DNS lookup of listen addresses and aliases set via the "alias=" > parameter in the config file (assuming the domain module isn't being used). > However I am having difficulty in getting this to work. From looking at the > code and the log file, it appears to me as if the alias isn't making it into the list > of hosts that is checked against for some reason. > > Kamailio proxy is behind a loadbalancer (Kamailio dispatcher) and proxy is > trying to relay the request because it doesn't recognise it as a local domain. > Any thoughts on where my error is would be appreciated. > > Relevant CFG file section - note that > RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT is 52.64.xxx.221 :5061 > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > ~~~~~~~~~~~~~~~~~~~~ > auto_aliases=yes > > # aliases and listening settings for sipproxy alias="aws-rcsstg.voip.systems" > listen=tcp:10.4.1.62:5060 advertise > RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT > > > My request is a REGISTER as follows: > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > received message > REGISTER call-id b2ed2fdd(a)192.168.148.30 received from 10.4.1.62 raw msg > REGISTER sip:aws-rcsstg.vvoip.systems;transport=tls SIP/2.0 > Via: SIP/2.0/TCP > 52.64.xxx.221:5060;branch=z9hG4bKa691.5fd57c2a783d3a5d7f6f80fcd0c815d > 6.0;i=d91 > Via: SIP/2.0/TLS > 192.168.148.30:5061;received=203.171.46.54;rport=64515;branch=z9hG4bK68 > b886fd00518;keep > From: <sip:+6421608919@aws- > rcsstg.vvoip.systems;transport=tls>;tag=4af7e7d8 > To: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls> > Call-ID: b2ed2fdd(a)192.168.148.30 > CSeq: 1995005270 REGISTER > Max-Forwards: 69 > Contact: > <sip:+6421608919@203.171.46.54:64515;transport=tls>;expires=240;+sip.inst > ance="<urn:uuid:23624e3d-c98e-3b76-9acb-5499202e2f8d>";reg- > id=1;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp- > application.ims.iari.rcs.sm";+g.oma.sip-im;+g.3gpp.iari-ref="urn%3Aurn- > 7%3A3gpp-application.ims.iari.gsma-is";+g.3gpp.cs-voice;+g.3gpp.iari- > ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs";+g.3gpp.iari- > ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcse.sp";+g.3gpp.icsi- > ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";video > User-Agent: FlowTalk-WindowsCommunicator_5.x > P-Access-Network-Info: IEEE-802.3 > Allow: > INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO,MESS > AGE,SUBSCRIBE,PUBLISH > Allow-Events: refer, presence > Supported: 100rel, replaces, timer, gruu, join, histinfo, path > Date: Wed, 13 Jul 2016 00:27:39 GMT > Content-Length: 0 > Path: > <sip:52.64.xxx.221:5060;transport=tcp;lr;received=sip:203.171.46.54:64515%3 > Btransport%3Dtls> > > > Relevant section of the log where if( !uri == me ) is tested: > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: > ERROR: *** cfgtrace:request_route=[SIPOUT] > c=[//etc/kamailio/kamailio.cfg] l=1052 a=16 n=if Jul 13 03:18:06 aws-rcsstg- > sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> > [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: > 24==9 && [aws-rcsstg.vvoip.systems] == [10.4.1.62] Jul 13 03:18:06 aws- > rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> > [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 > (advertise 5061) matches port 5061 Jul 13 03:18:06 aws-rcsstg-sip-proxy-i- > b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:603]: > grep_sock_info(): grep_sock_info - checking advertise if host==us: 24==13 > && [aws-rcsstg.vvoip.systems] == [52.64.xxx.221] Jul 13 03:18:06 aws-rcsstg- > sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> > [forward.c:448]: check_self(): check_self: host != me > > > Kind regards > > Shane Harrison > Senior Software Engineer > > Imagination Technologies NZ Limited > Level 2 > 1 Market Grove > Lower Hutt, 5010 > New Zealand > > PO Box 30-449 > Lower Hutt, 5040 > New Zealand > > Phone: +64 4 890-3681 ext 3361 >

2 1

Re: [SR-Users] kamcmd & route[REMOTE_TEST]
by Daniel-Constantin Mierla 15 Jul '16

15 Jul '16

Hello, On 14/07/16 11:02, Marino Mileti wrote: > > Hi guys, > > > > I've two question regarding rpc command. > > > > Is it possible to "execute" a specific route (for example my > route[REMOTE_TEST]) using a rpc command by kamcmd? > > > > If not exist some other tricks to do this? > you can send a xmlrpc, jsonrpc or just http request to kamailio and it will result is executing a route or event_route inside kamailio.cfg, from where you can execute other sub-routes. See xmlrpc, jsonrpc-s or xhttp modules. Obviously, you can send a sip message with tools like sipsak or sipp from command line and handle it inside request_route to execute what route block you want. Sometimes it may just suit executing a route block periodically -- for that see rtimer module. Cheers, Daniel > > > Another question...regarding uac module... > > When Kamailio starts the UAC module makes a REGISTER according to > database...and periodically refresh this registration. Is it possible > to "force" a refresh with rpc command by kamcmd? > > > > Many thanks! > > > > /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/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > 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://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] kamcmd & route[REMOTE_TEST]
by SamyGo 14 Jul '16

14 Jul '16

Hi Marino, How do you mean execute a route !? like enable a route for all calls onwards after the RPC command execution ? or for a specific call engage that route ? If this is to enable/disable a particular route for ALL calls based on a RPC command I'd make use of some DB or memcache variable to trigger this route i.e if(redis_cmd("srvN", "GET toggle_route", "r")){ $avp(signal) = $redis(r=>value); if($avp(signal) =~ 'true') { route(REMOTE_TEST); } } I used Redis, you can see if any RPC command can let you modify value of an AVP directly. For your second question check RPC commands uac.reg_refresh <http://www.kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.r.uac.reg_r…> . Regards, Sammy On Thu, Jul 14, 2016 at 5:02 AM, Marino Mileti <marino.mileti(a)alice.it> wrote: > Hi guys, > > > > I've two question regarding rpc command. > > > > Is it possible to "execute" a specific route (for example my > route[REMOTE_TEST]) using a rpc command by kamcmd? > > > > If not exist some other tricks to do this? > > > > Another question...regarding uac module... > > When Kamailio starts the UAC module makes a REGISTER according to > database...and periodically refresh this registration. Is it possible to > "force" a refresh with rpc command by kamcmd? > > > > Many thanks! > > > > *Marino Maria Mileti* > > *marino.mileti(a)alice.it <marino.mileti(a)alice.it>* > > > > *[image: cid:006a01cb6b0e$67eecdae$_CDOSYS2.0]**Reduce your energy > consumption and keep polar bears on ice!* > > > > > ------------------------------ > [image: Avast logo] > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > _______________________________________________ > 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 > >

1 0

kamcmd & route[REMOTE_TEST]
by Marino Mileti 14 Jul '16

14 Jul '16

Hi guys, I've two question regarding rpc command. Is it possible to "execute" a specific route (for example my route[REMOTE_TEST]) using a rpc command by kamcmd? If not exist some other tricks to do this? Another question...regarding uac module... When Kamailio starts the UAC module makes a REGISTER according to database...and periodically refresh this registration. Is it possible to "force" a refresh with rpc command by kamcmd? Many thanks! 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

Modules How To ?
by Eng Hooda 14 Jul '16

14 Jul '16

Hello All, I just started using kamailio last month , using basic configuration I got from online tutorials I got it to work. Now I need to add functionality like offline messages delivery , searching lead me to msilo module , unfortunately I could not find a detailed tutorial of how to enable it and troubleshoot it's operation. All what I managed to find was the official technical documentation. https://www.kamailio.org/docs/modules/stable/modules/msilo.html It provides a list of parameters which has a default value , with no indication if the declaration of any of them is mandatory if the default values are kept intact. also there is a route code which I do not know if it was mandatory to copy to kamailio.cfg . Not to mention that the configuration file example contains lines that is already similar to what I already have , so which to keep and which to replace ? That was a case to apply to , but general answers (with msilo as an example) that apply to all modules is what I need just to avoid having to ask again if I needed to enable another module. Or may be there is something out there that I need to read first. system details : {(OS:Debian8 i386),(Kamailio4.4 with mySQL , UDP only)} Thank you all in advance. Best Regards, Eng Hooda

2 2

is_direction / $fti / $tti not working possible
by Nathan Angelacos 13 Jul '16

13 Jul '16

I'm missing something obvious here, throwing this out to get a couple extra sets of eyeballs on it. I can't get rr modules is_direction() or the $fti / $tti pv's to work in 4.3.x or 4.4.x Relevant parts of the config: loadmodule "rr.so" modparam ("rr", "enable_full_lr", 1) modparam ("rr", "append_fromtag", 1) ... route { xlog ("L_INFO", "$rm: $fu -> $ru"); record_route (); if (has_totag ()) { xlog ("L_INFO", "Before Loose route: fti: $fti tti: $tti ci: $ci"); if (loose_route ()) { if (is_direction ("downstream")) { xlog ("L_INFO", "in-dialog $rm request from caller to callee"); } else { xlog ("L_INFO", "in-dialog $rm request from callee to caller"); } xlog ("L_INFO", "After Loose route: fti: $fti tti: $tti ci: $ci"); } } if (! t_relay ()) { sl_reply_error (); } } ---- Bob and Alice are hard phones, with kamailio + dbaliases in the middle. Alice calls Bob, then Bob (the callee) sends the BYE ---- Syslog: 01. INVITE: sip:alice@kamailio.org -> sip:bob@kamailio.org:5060 02. Alias lookup returned <sip:bob@biloxi.com> , sending on 03. ACK: sip:alice@kamailio.org -> sip:bob@biloxi.com 04. Before Loose route: fti: 707601862 tti: 45686ADF-10AAEB28 ci: 8744171(a)atlanta.com 05. in-dialog ACK request from caller to callee 06. After Loose route: fti: 707601862 tti: 45686ADF-10AAEB28 ci: 8744171(a)atlanta.com 07. BYE: sip:bob@kamailio.org -> sip:alice@atlanta.com:5066 08. Before Loose route: fti: 45686ADF-10AAEB28 tti: 707601862 ci: 8744171(a)atlanta.com 09. in-dialog BYE request from caller to callee 10. After Loose route: fti: 45686ADF-10AAEB28 tti: 707601862 ci: 8744171(a)atlanta.com Using the above config, is_direction always returns "caller", even when bob is sending the message. Looking at a packet capture: 01. Kamailio Sent: Record-Route: <sip:kamailio.org;lr=on;ftag=707601862> <on the 200 ok from Bob> Kamailio Received: Record-Route: <sip:kamailio.org;lr=on;ftag=707601862> Kamailio Sent: Record-Route: <sip:kamailio.org;lr=on;ftag=707601862> 07. Kamailio Received: Route: <sip:kamailio.org;lr=on;ftag=707601862> Kamailio Sent: Record-Route: <sip:kamailio.org;lr=on;ftag=45686ADF-10AAEB28> I expect $fti to be wrong before the loose_route (line 08); but lines 09 and 10 are wrong. I'm obviously missing something trivial here. Any ideas what I should check next? Thanks!

2 2

Kamailio Enviroment
by Rodrigo Moreira 13 Jul '16

13 Jul '16

Hi, Right, to basic configuration of the P-I-S-CSCF services just I call the CFGS appropriate example (directory) for each entity to achieve basic funcionality. What exactly dispacher? How to config it? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Kamailio know which Interface Packet would route out
by SamyGo 13 Jul '16

13 Jul '16

Hi, Question: Is there a function in kamailio to get the interface/listen IP kamailio would use to route a packet out for a domain / ip ? Example: $avp(interface_ip) = find_interface_for_request("192.168.12.123"); And it gives me the Kamailio's Listen address that would be used for sending call out to this IP/Domain ! I've a weird situation where a multi-homed kamailio with about 7 different interfaces needs to send call out, I've so many RTProxy instances to engage but I can't decide which instance to engage since I do not know which Interface the Packet would route out from. Regards, Sammy

2 2

Re: [SR-Users] shm_malloc
by Daniel-Constantin Mierla 13 Jul '16

13 Jul '16

Hello, Can you be more explicit about what do you expect? Daniel On 13/07/16 07:27, Prakash wrote: > > Hello Daniel, > > > > Many thanks for the response. > > Could you please provide me the behind scenes of shm_malloc()?. > > It will be very helpful for me to segregate without any flaws. > > > > > > Thanks, > > Prakash > > > > *From:*sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On > Behalf Of *Daniel-Constantin Mierla > *Sent:* Tuesday, July 12, 2016 6:34 PM > *To:* Kamailio (SER) - Users Mailing List > *Subject:* Re: [SR-Users] shm_malloc > > > > Hello, > > I don't expect to be very hard to extract the shm management code from > kamailio. It is inside the subfolder mem/ . > > Then, there should be some dependecies that you need to get out from > the core, coming in my mind is locking (mutexes). Maybe atomic > operations (not remembering by heart if some counters are kept for > memory management) and few other bits. > > It should work without crash, you just need to get it initialized as > kamailio does it in it's main function. > > Cheers, > Daniel > > On 12/07/16 13:57, Prakash wrote: > > Hello, > > > > Is it possible to take out shared memory code from kamailio? > > if so ,Will it work without any crash? > > > > > _______________________________________________ > > 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://www.asipto.com - http://www.kamailio.org > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

access to tls X509v3 Subject Alternative Name
by Juha Heinanen 13 Jul '16

13 Jul '16

Is there some means to get access to local and peer ssl certificate X509v3 Subject Alternative Name value? For example, if there is this in certificate: Subject: CN=foo.bar.com X509v3 Subject Alternative Name: DNS:foo.bar.com, DNS:foo.car.com, DNS:car.com $tls_my_san_hostname produces only foo.bar.com. -- Juha

2 5

Re: [SR-Users] shm_malloc
by Daniel-Constantin Mierla 13 Jul '16

13 Jul '16

Hello, I don't expect to be very hard to extract the shm management code from kamailio. It is inside the subfolder mem/ . Then, there should be some dependecies that you need to get out from the core, coming in my mind is locking (mutexes). Maybe atomic operations (not remembering by heart if some counters are kept for memory management) and few other bits. It should work without crash, you just need to get it initialized as kamailio does it in it's main function. Cheers, Daniel On 12/07/16 13:57, Prakash wrote: > > Hello, > > > > Is it possible to take out shared memory code from kamailio? > > if so ,Will it work without any crash? > > > > _______________________________________________ > 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://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Determining if request is from a local domain
by Shane Harrison 13 Jul '16

13 Jul '16

Hi all, >From previous posts my understanding is that RURI's are matched against reverse DNS lookup of listen addresses and aliases set via the "alias=" parameter in the config file (assuming the domain module isn't being used). However I am having difficulty in getting this to work. From looking at the code and the log file, it appears to me as if the alias isn't making it into the list of hosts that is checked against for some reason. Kamailio proxy is behind a loadbalancer (Kamailio dispatcher) and proxy is trying to relay the request because it doesn't recognise it as a local domain. Any thoughts on where my error is would be appreciated. Relevant CFG file section - note that RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT is 52.64.xxx.221 :5061 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ auto_aliases=yes # aliases and listening settings for sipproxy alias="aws-rcsstg.voip.systems" listen=tcp:10.4.1.62:5060 advertise RCS_LB_PUBLIC_IP:RCS_LB_PUBLIC_PORT My request is a REGISTER as follows: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ received message REGISTER call-id b2ed2fdd(a)192.168.148.30 received from 10.4.1.62 raw msg REGISTER sip:aws-rcsstg.vvoip.systems;transport=tls SIP/2.0 Via: SIP/2.0/TCP 52.64.xxx.221:5060;branch=z9hG4bKa691.5fd57c2a783d3a5d7f6f80fcd0c815d6.0;i=d91 Via: SIP/2.0/TLS 192.168.148.30:5061;received=203.171.46.54;rport=64515;branch=z9hG4bK68b886fd00518;keep From: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls>;tag=4af7e7d8 To: <sip:+6421608919@aws-rcsstg.vvoip.systems;transport=tls> Call-ID: b2ed2fdd(a)192.168.148.30 CSeq: 1995005270 REGISTER Max-Forwards: 69 Contact: <sip:+6421608919@203.171.46.54:64515;transport=tls>;expires=240;+sip.instance="<urn:uuid:23624e3d-c98e-3b76-9acb-5499202e2f8d>";reg-id=1;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcs.sm";+g.oma.sip-im;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-is";+g.3gpp.cs-voice;+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs";+g.3gpp.iari-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.rcse.sp";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";video User-Agent: FlowTalk-WindowsCommunicator_5.x P-Access-Network-Info: IEEE-802.3 Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO,MESSAGE,SUBSCRIBE,PUBLISH Allow-Events: refer, presence Supported: 100rel, replaces, timer, gruu, join, histinfo, path Date: Wed, 13 Jul 2016 00:27:39 GMT Content-Length: 0 Path: <sip:52.64.xxx.221:5060;transport=tcp;lr;received=sip:203.171.46.54:64515%3Btransport%3Dtls> Relevant section of the log where if( !uri == me ) is tested: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: ERROR: *** cfgtrace:request_route=[SIPOUT] c=[//etc/kamailio/kamailio.cfg] l=1052 a=16 n=if Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 24==9 && [aws-rcsstg.vvoip.systems] == [10.4.1.62] Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 5061) matches port 5061 Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [socket_info.c:603]: grep_sock_info(): grep_sock_info - checking advertise if host==us: 24==13 && [aws-rcsstg.vvoip.systems] == [52.64.xxx.221] Jul 13 03:18:06 aws-rcsstg-sip-proxy-i-b879b469 /usr/sbin/kamailio[9367]: DEBUG: <core> [forward.c:448]: check_self(): check_self: host != me Kind regards Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

1 0

Detect 200 OK for ReINVITE
by Grant Bagdasarian 12 Jul '16

12 Jul '16

Hi, Is it possible in Kamailio, or SIP in general, to detect if a 200 OK is for a ReINVITE? Regards, Grant

4 5

Video conferencing with Kamailio
by Jay Li 12 Jul '16

12 Jul '16

Dear All, I'm curious if anybody has set up an infrastructure for video conferencing utilizing Kamailio as a proxy (like NAT support and so so). I found a kind of old tutorial "Run you own Skype-like service in less than one hour" kamailio:skype-like-service-in-less-than-one-hour [Asipto - SIP and VoIP Knowledge Base Site] written in 2013 on Kamailio 3.1. I wonder if this feature has been further developed in later releases. I know Kamailio is more often used as a SIP proxy, but somehow I got the impression that Kamailio supports WebRTC, so I wonder if it's possible to implement a many to many video conferencing infrastructure using Kamailio. If the answer is yes, how's the performance or the bottleneck I should pay more attention to (maybe bandwidth for video traffic, especially with NAT? ) Anyway I'm a newbie to both Kamailio and video conference, so any suggestion/discussion is appreciated. Thanks. Regards,Jay

2 3

destination ip address
by Péter Barabás 12 Jul '16

12 Jul '16

Hi All, i need to store some information in routing_logic script into database with useing SQLOPS module. I made a custom table and store such data like source and destination username, domain, ip address and port. It works fine for almost every parameter except destination IP address and port. I want to ask what is the correct way to determine the destination IP address and port of a user for a MESSAGE, INVITE or BYE message? The $Ri and $Rp is not the correct ones. Thanks. Dr. Péter Barabás

3 3

getting the local 408 response
by Jayesh Nambiar 12 Jul '16

12 Jul '16

Hi, I'm looking to catch the local 408 response generated by kamailio after the fr_timer strikes in. I was assuming to get it in the sl:local-request, but I dont see it in my log messages. I can catch responses like a 200 for register or a 487 for a message who's transaction is not yet created. Any proper way on how do I get to know when kamailio generates a local 408? I'm on the master branch (commit: 54562b873c2c3d0a04f0587e7706188a3eb19a74). Thanks, - Jayesh

4 6

Calculating time differences
by Sebastian Damm 11 Jul '16

11 Jul '16

Hi, we are suspending a transaction at one place in our route and resume it later. I want to log the time difference between suspending and resuming. I thought I could do it with saving $TV(Sn), but apparently I can't. Kamailio says, it can't convert the saved value into integer. This true, of course. I was searching for a transformation function converting this string into float, but there is none. Is there any way to calculate a time difference other than to live with seconds-only precision? Best Regards, Sebastian

2 1

Kamailio WEBRTC questions/ confusions
by Zaka 10 Jul '16

10 Jul '16

Dear List: I wonder if this has been asked before. Please ignore or point to the appropriate source, if needed. Simply, I have little experience with SIP servlets implementing WEBRTC/ Click2Call etc... Considering a Call Centre scenario offering a WEBRTC solution. Following are the confusions regarding requirements: 1: A Kamailio Box (running WEBSOCKET.config) 2: Customer will use web browser (websocket compatible of course) and enter address of the KAMAILIO Box or we shall need another KAMAILIO to statelessly forward the call to next available operator using Hunt Group or similar scheme? 3: If we don't plan to implement TLS/ MSRP (disabling these in the config file should be suffice?) 4: Last, but not the least, how will the users send request for login from browser? (assuming they have accounts created in the DB on Kamailio Box running WEBSOCKET supported KAMAILIO) With anticipatory thanks & regards, Zaka

3 6

Kamailio Enviroment
by Rodrigo Moreira 09 Jul '16

09 Jul '16

Hello, I would of your help installing Kamailio with the ability to authenticate to an HSS. Mainly want each IMS service to run on a different machine, ie, P / S / I-CSCF. I've tried several ways to set up, but without success. Does anyone have a step by step? I count on help. Hug. Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

1 0

re-INVITE on IP Change with Outbound
by Colin Morelli 08 Jul '16

08 Jul '16

Hey list, So I'm using the outbound module to ensure subsequent requests are delivered over an active TCP connection established by the mobile client. However, now I'm trying to add support for automatically responding to network changes (WiFi <-> LTE), and it's creating problems. Primarily, the issue is that when the device sends a re-INVITE for the call, its Route: header includes a flow-token corresponding to its *old* socket connection. Because the new request doesn't have the same source IP/port as the old connection did Kamailio thinks that the next route hop it needs to take is to pass the request over to the connection identified by the flow token. So it actually tries to re-open a TCP connection back out to the client even though I'm making the re-invite because that old transport is no longer valid. Does anyone have a recommended way of handling this case? I can add another header/param to the route header and manually remove it. Before loose_routing. Essentially, on in-dialog INVITE requests I want to ignore the outbound token on the originating side but I can't seem to find a way to do that without re-writing the header before loose_routing. Best, Colin

1 1

shtinc throws readonly error
by Sebastian Damm 08 Jul '16

08 Jul '16

Hi, I'm trying to use this line in one route: $shtinc(pushedUsers=>$avp(s:authuser)); However, when checking the syntax of this file, Kamailio says: 0(609) : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio_sip_proxy.cfg, line 3304, column 2-39: read only pvar in assignment left side According to the pseudo vars documentation, I should be able to use it like that. Can someone tell me what I am missing? Best Regards, Sebastian

2 3

tsilo ts_append problem
by gmele 08 Jul '16

08 Jul '16

Hello, we are using tsilo to achieve mobile push notification in kamailio. In tsilo module documentation, ts_append() function is used with the pseudo variable “$tU” holding the username of the to-uri. Our config looks like : route[PUSHJOIN] { $var(hjoin) = 0; lock("$tu"); $var(hjoin) = $sht(vtp=>join::$tu); $var(hstored) = $sht(vtp=>stored::$tu); $sht(vtp=>join::$tu) = $null; unlock("$tu"); if ($var(hjoin)==0) { if ($var(hstored)) { ts_append("location", "$tU"); } return; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } In ts_append() function code, ts_check_uri() is called on a variable holding a username and not a URI causing the failure of the function ts_check_uri() and resulting in ts_append() to exit. The following line is displayed in the log: [tsilo.c:205]: ts_check_uri(): bad uri [giovanni.mele]. The function ts_store() stores in the R-URI field the username and not the request URI. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay(PROXY_IP, PROXY_PORT)) { sl_reply_error(); } if (is_method("INVITE")) { ts_store(); $sht(vtp=>stored::$tu) = 1; route(SENDPUSH); } exit; } route[INVRESUME] { statsd_gauge("SIPRegistrar.InviteResumed", "+1"); lookup("location"); t_relay(PROXY_IP, PROXY_PORT); ts_store(); $sht(vtp=>stored::$tu) = 1; } With the command “kamcmd ts.dump” we can see the following lines: R-URIs: { R-URI: giovanni.mele Hash: 74014921 Transactions: { Transaction: { Tindex: 63704 Tlabel: 2105932514 } } } If we try to use the ts_append() function with the pseudo variable “$tu” holding the full to-uri address, there is no error returned by ts_check_uri() (as this is called on a correct URI) but the transaction corresponding to the to-uri cannot be found (maybe because the R-URI stored by ts_store() is the username and not the full R-URI). How can we make things work ? By storing the full R-URI with ts_store() ? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/tsilo-ts-append-problem-tp150184.ht… Sent from the Users mailing list archive at Nabble.com.

3 10

4.4.1: Only first UDP Listener is producing log messages
by Sebastian Damm 08 Jul '16

08 Jul '16

Hi, I upgraded one of our systems today to 4.4.1. And when checking whether everything is okay, I noticed, that only the first UDP listener (from kamcmd core.ps) is actually sending lines to the log. The only other logs I see are from usrloc timer worker. kamcmd> core.ps 10737 main process - attendant 10740 udp receiver child=0 sock=172.20.40.5:5060 10741 udp receiver child=1 sock=172.20.40.5:5060 10742 udp receiver child=2 sock=172.20.40.5:5060 10743 udp receiver child=3 sock=172.20.40.5:5060 We are sending quite a lot of information via xlog("L_NOTICE", "foo"); to syslog, but the only lines I see are from PID 10740. Kamailio works fine otherwise, handling all requests, the usrloc entries didn't drop, and customers don't complain. So I think it's only a logging issue. There are about 50k registrations on the machine and more than 10 Mbit of traffic. When I downgrade to 4.3.5, I get output from all listeners again. Has there been any change between 4.3 and 4.4 regarding logging? The upgrade document says nothing about this. Best Regards, Sebastian

2 4

CNXCC compilation errors
by Infinicalls Infinicalls 07 Jul '16

07 Jul '16

Hi, Thanks. Well I tried it before also but could not proceed with since I got the compilation errors. Now also I am struck up with the same error mentioned in http://lists.sip-router.org/pipermail/sr-users/2016-January/091574.html Has anybody found a solution to this? Mine is Ubuntu 12.x LTS regards Ganesh Kumar On 7/7/16, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > look at cnxcc module for implementing a prepaid system. There should be > also a presentation about it at one of the kamailio world conferences -- > see kamailio.org/events/ > > Cheers, > Daniel > > > On 07/07/16 11:33, Infinicalls Infinicalls wrote: >> Hi Daniel, >> >> Thanks. I figured out later :-) >> >> I am actually trying to implement prepaid system with dialog and call >> control. In Siremis, after a call the duration and other details get >> updated but not the cost. This is inspite of having rates and >> kamailio_cdr / kamailio_rating() in cron. >> >> Any idea? >> >> regards >> Ganesh Kumar >> >> On 7/7/16, Daniel Tryba <d.tryba(a)pocos.nl> wrote: >>> On Wed, Jul 06, 2016 at 11:01:17PM +0530, Infinicalls Infinicalls wrote: >>>> " ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module >>>> matching <call_control> found >>>> " >>>> while adding modparam("call_control", "disable", 1). I have already >>>> included pv and dialog in the modules list. >>>> >>>> Any idea? >>> Try loading the module first :) >>> >>> https://www.kamailio.org/wiki/cookbooks/4.4.x/core#loadmodule >>> >>> >>> _______________________________________________ >>> 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 >>> >> _______________________________________________ >> 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://www.asipto.com - http://www.kamailio.org > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > > _______________________________________________ > 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 > -- -- http://www.infinicalls.com

1 0

A parameter with a life span of the entire dialog
by Jay Li 07 Jul '16

07 Jul '16

Dear All, I'm trying to set a few parameters for an "INVITE", and access their values in the failure routine if the call fails. $var() doesn't seem suitable for my purpose as it only lasts for one process. I think if I set a $var() in "INVITE", it will be gone when I receive the failure response. $sht() seems able to last long but it will be accessed by all processes which might include ones not belonging to the same dialog. I wonder if somebody would give me some suggestions here to handle such situations. Does Kamailio provide a mechanism to set up a parameter with a life span of the entire dialog? Thanks. Regards,Jay

2 1

call_control not found error
by Infinicalls Infinicalls 07 Jul '16

07 Jul '16

Hi, I am trying to implement a prepaid system and get this error " ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <call_control> found " while adding modparam("call_control", "disable", 1). I have already included pv and dialog in the modules list. Any idea? regards Ganesh Kumar

3 3

t_check_trans() exits route execution if transaction is found
by Sebastian Damm 07 Jul '16

07 Jul '16

Hi, on one of our systems we are checking a few times whether a transaction already exists. We noticed that route execution stops if the transaction is found. And after looking into the code, we found that it seems to be intended like this. >From tm.c: int t_check_trans(struct sip_msg* msg) { if (msg->first_line.type==SIP_REPLY) { [..] } else { switch(t_check_msg(msg, 0)){ [..] case 1: /* found */ [..] return 0; /* exit from the script */ } } } Can anybody explain, why script execution should stop when a running transaction is found? I would expect that then the function returns 1 so I can use it in my if statement and continue with my routing script. Thanks already for an explanation, Sebastian

2 1

Kamailio compact headers ?!
by Anonim Stefan 07 Jul '16

07 Jul '16

Hi, Is there a parameter to be enabled such that Kamailio will append headers in compacted form? (I'm thinking of Via: and Record-Route:) Thanks, Stefan

6 10

Kamailio and Mongodb
by C. 06 Jul '16

06 Jul '16

Hi, I am still trying to get Kamailio to work with MongoDB. Followed instructions here https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb <https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb>, but I cannot get things to install. Are there any instructions on the exact configuration changes that need to be made to support Mongodb? For example, kamctl points to MySQL, etc Thanks! Chuks.

1 0

Locally originated BYE dialog database handle
by Dirk Teurlings - Signet B.V. 06 Jul '16

06 Jul '16

Hi, There's something going on with a locally originated BYE and the dialog module running with a database backend. Here's the scenario. Call gets setup, and is being monitored by outside processes (non-Kamailio). After a while these processes determine the call is taking too long (or too expensive), so it has to be killed. To do this we're using the kamcmd dlg.end_dlg with h_entry and h_id as parameter. This works to end the call, but is throws some weird issues into the routing. First off, the event_route[dialog:end] is not thrown 99% of the time, though I would expect it to. Second, This error pops up in the log /usr/sbin/kamailio[25094]: ERROR: dialog [dlg_db_handler.c:199]: use_dialog_table(): invalid database handle As if the command issued by kamcmd isn't able to get connected to the dialog database. In any case, this error results in the dialog entry for this call not being removed from the database. So this piles up after a while. Anybody any thoughts on this? Cheers, Dirk

2 2

(no subject)
by John Petrini 06 Jul '16

06 Jul '16

Hello, I'm trying to append a branch, assign it a q value so it has priority and serialize the new branch and the main branch. The goal is to send the new branch off to our CNAM provider, capture the CNAM in their reply to an $avp then drop the branch, stick the cnam in the main branch and allow the call to continue as normal. We do this currently using OpenSIPS with serialize_branches(1) but the of load_contacts() behavior seems to be different than serialize_branches. route[CNAM_DIPS] t_on_branch("CNAM_DIPS"); t_on_reply("CNAM_DIPS"); t_on_failure("CNAM_DIPS"); t_load_contacts(); t_next_contacts(); append_branch("sip:2223334444@22.33.444.55:5600;trans-type=5", "0.5"); relay(); break; Two problems. First when t_next_contacts is called I get the following error: ERROR: pv [pv_branch.c:58]: pv_get_branchx(): error accessing branch [0] So it looks like the branches aren't actually being loaded? Second problem is that despite the first error my call goes out on the branch I appended and I get a reply from the CNAM provider but Kamailio forwards that reply back to the phone causing the phone to send a second invite. In OpenSIPs we just called drop, to drop the branch and let the main branch continue but this doesn't seem to work in Kamailio. I've read and reread the documentaion but it doesn't seem to behave the way the documentation says it should. Is there something I'm missing? Is there a better way to accomplish this in Kamailio? Thank You, ___ John Petrini

3 9

Issue with userblacklist module after 4.4.2 upgrade
by Tim Balmer 06 Jul '16

06 Jul '16

Hello, We are running Kamailio version 4.4 and have our sipproxy servers configured to do auto updates within te same major version based on a 4 weekly schedule (we have 4 sipproxy / Kamailio servers). So this morning as expected the first of our sipproxy servers automatically upgraded from 4.4.1 to 4.4.2, unfortunately after this update we have issues with the userblacklist module. The problem is that with the 4.4.2 release every number is now matching our userblacklists although it is not really blacklisted. After some digging in the logs I found this is probably because the userblacklist module is looking for the wrong database table. With every check_user_blacklist call I see errors like this appearing in the logs: Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Table 'kamailio.<number>' doesn't exist (1146) Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: userblacklist [db.c:68]: db_build_userbl_tree(): error while executing query. Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: userblacklist [userblacklist.c:277]: check_user_list(): cannot build d-tree I replaced the real number with <number> in the log output but it seems like the userblacklist module is now looking for a database table with the name of the number to check against. On our development server I tested setting the global userblacklist_table parameter and also passing the table parameter within the check_user_blacklist call but both are not making any difference. After downgrading the sipproxy back to 4.4.1 everything is working as expected again so it looks to me like a bug in the 4.4.2 release. Is anyone else experiencing the same issues after the 4.4.2 upgrade or is it just me doing something wrong? Best Regards, Tim Balmer

2 12

Kamailio summer-of-documentation - join the effort
by Olle E. Johansson 06 Jul '16

06 Jul '16

Hi! Looking for a summer-of-documentation project? Which README is the best? Which one is so full of Swedish that you totally fail to parse it? Check our documentation pages, suggest improvements on the mailing lists (or here on Facebook) and give us feedback! This is something you can spend your lazy summer days on - a nice cool drink, cool sunshades and Kamailio documentation! We constantly work to update our README files for every module and the cookbooks for the Kamailio core, the transformations and pseudovariables. But we’re developers, not casual administrators and propably fail to see some obvious things missing. Please tell us (feel free to use the mailing lists). And if you can - but not required at all - you can suggest rewrites in the XML source and upload to github. Let’s keep our docs up to date and make sure they really assist our users! http://kamailio.org/docs/modules/devel/ Have a great summer! Regards, /O

2 1

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by tegamjg 06 Jul '16

06 Jul '16

El 05/07/2016 11:36, Daniel Tryba <d.tryba(a)pocos.nl> escribió: > > Please keep the mailinglist in the loop, so everybody might benefit from > our ramblings :) > > > Still there are few things i dont understand, i am not using asterisk > > just as a voicemail server since they are actually handling also the > > calls passing first from kamailio and being load balanced to those > > asterisk boxes. May i still use call forwarding as you are using it? > > (Both asterisk have a shared storage with a clustered filesystem, so > > both will be able to see voice messages) > > Yes I think so. I use a seperate machine for voicemail but I see no > problem with other uses (I used to use it for playback of messages and > transcoding ebtween incompatible endpoints). > > By using the prefixes in kamailio to the username in $ru I have in the > extensions.conf: > > exten => _tovm-.,1,NoOp(leave voicemail) > exten => _tovm-.,n,Answer() > exten => _tovm-.,n,Set(CHANNEL(language)=nl) > exten => _tovm-.,n,Voicemail(${EXTEN:5},us) > exten => _tovm-.,n,Playback(Goodbye) > exten => _tovm-.,n,Hangup() > > exten => _getvm-.,1,NoOp(read voicemail) > exten => _getvm-.,n,Set(CHANNEL(language)=nl) > exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) > exten => _getvm-.,n,Hangup() > > > The other question is that i actually though that you need asterisk to > > have users configured in sipusers realtime table to associate their > > mailboxes, which i dont have since those users are stored in the > > subscriber table of kamailio. So am i still able to configure > > voicemail like you are doing it by syncing with the voicemail table?, > > i really hope so haha > > I forgot that fact. So yes I have a realtime sip users list (with > host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio > username, no password (this machine is not directly accessible from > outside)) Sorry, I think i rushed the last answer but if you could answer that one would be nice How are you handling the calls? Just with kamailio/rtpproxy? Because i am also using asterisk for calls with dial application and for nat issues (with kamailio behind nat) i am using also kamailio/rtpproxy for outside. All this with just handling users (registration and location) in the subscribe and location table of kamailio. That is why i am not using sipusers table of asterisk because of nat was behaving weird using it that way. Could it be possible to use both tables without expecting a different behaviour? Or is not, in the end, a good idea and i need to keep users in sipusers table? > > You might not be able to have endpoints able to subscribe to > notifications due to this. I baked something inspired by: > http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… > that appears to work for me. > > _______________________________________________ > 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

2 1

Kamailio out-of-mem
by José Seabra 06 Jul '16

06 Jul '16

I'm running burning tests against kamailio and after some time ~ 2 days I see kamailio out of mem: ERROR: tm [t_lookup.c:1271]: new_t(): ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed The result of command kamcmd mod.stats all is: Module: core { sip_msg_shm_clone(494): 10653392 build_req_buf_from_sip_req(2149): 2565480 msg_lump_cloner(978): 2943584 create_avp(175): 969736 xavp_new_value(94): 858469320 counters_prefork_init(207): 16896 cfg_clone_str(130): 88 cfg_shmize(217): 744 init_pt(110): 8 init_pt(105): 8 init_pt(104): 3080 dtrie_insert(157): 240 dtrie_insert(148): 48 dtrie_init(60): 160 dtrie_init(51): 32 register_timer(1011): 360 cfg_register_ctx(47): 32 init_tcp(4635): 8192 init_tcp(4629): 32768 init_tcp(4621): 8 init_tcp(4614): 8 init_tcp(4607): 8 init_tcp(4601): 8 init_tcp(4589): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(437): 16384 init_dst_blacklist(430): 8 timer_alloc(514): 96 init_dns_cache(366): 8 init_dns_cache(358): 16384 init_dns_cache(351): 16 init_dns_cache(345): 8 init_timer(283): 8 init_timer(282): 16384 init_timer(281): 8 init_timer(280): 8 init_timer(269): 8 init_timer(237): 8 init_timer(220): 278544 init_timer(219): 8 init_timer(206): 8 cfg_child_cb_new(830): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 shm_core_lock_init(153): 8 Total: 875992224 } Module: dmq { alloc_job_queue(229): 128 shm_str_dup(723): 24 build_dmq_node(156): 448 add_peer(67): 168 mod_init(243): 8 mod_init(237): 96 init_dmq_node_list(66): 24 init_peer_list(33): 24 Total: 920 } Module: auth_db { Total: 0 } Module: dialog { init_dlg_table(284): 131088 init_dlg_table(276): 8 init_dlg_table(269): 8 init_dlg_table(264): 8 init_dlg_timer(57): 8 init_dlg_timer(48): 32 new_dlg_profile(187): 624 Total: 131776 } Module: rr { Total: 0 } Module: xlog { Total: 0 } Module: kex { pkg_proc_stats_init(79): 1056 Total: 1056 } Module: db_postgres { lock_set_alloc(70): 80 Total: 80 } Module: pv { Total: 0 } Module: textops { Total: 0 } Module: htable { ht_cell_new(183): 1104 ht_init_tables(381): 8388608 ht_add_table(278): 928 Total: 8390640 } Module: sqlops { Total: 0 } Module: tm { build_uac_req(1543): 848 prepare_new_uac(479): 28880 _reply_light(542): 751504 build_local_reparse(341): 1168568 relay_reply(1884): 643088 build_cell(317): 9826624 prepare_new_uac(524): 8112 prepare_new_uac(509): 10192 init_tm_stats_child(60): 5632 insert_tmcb(137): 136 init_tmcb_lists(74): 16 init_tmcb_lists(72): 16 init_hash_table(467): 2097152 Total: 14540768 } Module: usrloc { new_urecord(65): 136 new_ucontact(98): 1096 new_urecord(58): 288 shm_str_dup(723): 1200 build_stat_name(51): 64 new_udomain(93): 40960 new_udomain(86): 48 new_dlist(573): 16 new_dlist(565): 32 register_ulcb(94): 32 init_ulcb_list(45): 16 lock_set_alloc(70): 4112 Total: 48000 } Module: tmx { tmx_check_pretran(250): 864 tmx_check_pretran(271): 1216 tmx_init_pretran_table(90): 256 Total: 2336 } Module: sl { init_sl_stats_child(125): 4224 sl_startup(83): 8 init_sl_stats(110): 8 Total: 4240 } Module: carrierroute { rule_fixup_recursor(584): 8 shm_str_dup(723): 56 add_route_rule(74): 152 add_route_flags(225): 48 create_domain_data(83): 32 create_carrier_data(59): 8 create_carrier_data(50): 64 load_route_data_db(295): 8 load_domain_map(182): 48 load_domain_map(171): 24 load_carrier_map(126): 48 load_carrier_map(115): 24 reload_route_data(172): 64 init_route_data(74): 8 Total: 592 } Module: dispatcher { reindex_dests(600): 1008 add_dest2list(350): 168 add_dest2list(324): 4200 init_data(204): 16 init_data(195): 16 mod_init(309): 8 mod_init(307): 8 ds_ping_active_init(102): 8 Total: 5432 } Module: db_text { dbt_init_cache(81): 256 dbt_init_cache(69): 8 dbt_init_cache(54): 8 Total: 272 } The result of command kamctl shmem is: shmem:fragments = 6353 shmem:free_size = 406208 shmem:max_used_size = 8464105472 shmem:real_used_size = 8463699264 shmem:total_size = 8464105472 shmem:used_size = 5190698208 The server has 8G of shared memory allocated, and the issue happens more or less after 2 days running tests. I'm using sipp to generate traffic, it is sending 500CAPS. There is any possibility of memory leak on Kamailio 4.4.0? Best Regards -- José Seabra

2 11

Kamailio blocked when using with Radius accounting when connection is lost with the database.
by Mx R 06 Jul '16

06 Jul '16

Hi there. I have the following scheme. 1. SIP server is running kamailio with module "acc", radiusclient-ng and freeradius. 2. DB server with MSSQL. The problem is the following, if the connection is lost with the database, no one can make calls. Even after the connection to the database is restored. We have to restart the radius and kamailio. And only after that can make calls. Kamailio logs rc_send_server: no reply from RADIUS server localhost: 1813 ERROR: acc [acc.c: 720]: acc_rad_request (): radius-ing failed Radius logs. Error: rlm_sql_unixodbc: SQL down 08S01 [unixODBC] [FreeTDS] [SQL Server] Write to the server failed Below are links to a similar problem. http://lists.opensips.org/pipermail/users/2012-March/021100.html http://lists.opensips.org/pipermail/users/2012-March/021102.html http://lists.opensips.org/pipermail/users/2012-March/021105.html http://lists.opensips.org/pipermail/users/2012-March/021128.html

1 0

Simple Kamailio configuration for Asterisk load balancing?
by Tickling Contest 06 Jul '16

06 Jul '16

Hello, I am beginning to front my Asterisk cluster with Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for this. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for Kamailio with the following features: (a) Support an arbitrarily large number of Asterisk servers (say, upto 10). (b) Offload SIP registration to a realtime/mysql DB used by the PBXs. Please also let me know if I should have to change my Asterisk PBX config in any way for this to happen. On a very related note, while I appreciate the fact that you need to _really_ understand SIP to configure Kamailio, it should be possible to get this setup by just running a script or GUI. It is difficult to find information on how to load balance multiple PBXs behind Kamailio. Thanks!

3 5

[rtprpoxy LB]
by Noriyuki Hayashi 06 Jul '16

06 Jul '16

Hi, Thank you for be used great KAMAILIO. Regarding rtpproxy LB. I configured as below. modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.136:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.137:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.138:7722") modparam("rtpproxy", "rtpproxy_disable_tout", 20) modparam("rtpproxy", "rtpproxy_tout", 2) modparam("rtpproxy", "rtpproxy_retr", 2) I could not restart kamailio when one rtpproxy server is down. Does anyone tell this solution by the parameter or somthing ? I am looking forward from anyone. Thank you so much. Noriyuki Hayashi

1 0

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by tegamjg 05 Jul '16

05 Jul '16

El 05/07/2016 11:36, Daniel Tryba <d.tryba(a)pocos.nl> escribió: > > Please keep the mailinglist in the loop, so everybody might benefit from > our ramblings :) > > > Still there are few things i dont understand, i am not using asterisk > > just as a voicemail server since they are actually handling also the > > calls passing first from kamailio and being load balanced to those > > asterisk boxes. May i still use call forwarding as you are using it? > > (Both asterisk have a shared storage with a clustered filesystem, so > > both will be able to see voice messages) > > Yes I think so. I use a seperate machine for voicemail but I see no > problem with other uses (I used to use it for playback of messages and > transcoding ebtween incompatible endpoints). > > By using the prefixes in kamailio to the username in $ru I have in the > extensions.conf: > > exten => _tovm-.,1,NoOp(leave voicemail) > exten => _tovm-.,n,Answer() > exten => _tovm-.,n,Set(CHANNEL(language)=nl) > exten => _tovm-.,n,Voicemail(${EXTEN:5},us) > exten => _tovm-.,n,Playback(Goodbye) > exten => _tovm-.,n,Hangup() > > exten => _getvm-.,1,NoOp(read voicemail) > exten => _getvm-.,n,Set(CHANNEL(language)=nl) > exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) > exten => _getvm-.,n,Hangup() > > > The other question is that i actually though that you need asterisk to > > have users configured in sipusers realtime table to associate their > > mailboxes, which i dont have since those users are stored in the > > subscriber table of kamailio. So am i still able to configure > > voicemail like you are doing it by syncing with the voicemail table?, > > i really hope so haha > > I forgot that fact. So yes I have a realtime sip users list (with > host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio > username, no password (this machine is not directly accessible from > outside)) That is what i was afraid of, so you are not handling kamailio users in the subscribe table of kamailio database but in sipusers table of asterisk database, is that correct? > > You might not be able to have endpoints able to subscribe to > notifications due to this. I baked something inspired by: > http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… > that appears to work for me. > > _______________________________________________ > 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

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Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by Daniel Tryba 05 Jul '16

05 Jul '16

Please keep the mailinglist in the loop, so everybody might benefit from our ramblings :) > Still there are few things i dont understand, i am not using asterisk > just as a voicemail server since they are actually handling also the > calls passing first from kamailio and being load balanced to those > asterisk boxes. May i still use call forwarding as you are using it? > (Both asterisk have a shared storage with a clustered filesystem, so > both will be able to see voice messages) Yes I think so. I use a seperate machine for voicemail but I see no problem with other uses (I used to use it for playback of messages and transcoding ebtween incompatible endpoints). By using the prefixes in kamailio to the username in $ru I have in the extensions.conf: exten => _tovm-.,1,NoOp(leave voicemail) exten => _tovm-.,n,Answer() exten => _tovm-.,n,Set(CHANNEL(language)=nl) exten => _tovm-.,n,Voicemail(${EXTEN:5},us) exten => _tovm-.,n,Playback(Goodbye) exten => _tovm-.,n,Hangup() exten => _getvm-.,1,NoOp(read voicemail) exten => _getvm-.,n,Set(CHANNEL(language)=nl) exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) exten => _getvm-.,n,Hangup() > The other question is that i actually though that you need asterisk to > have users configured in sipusers realtime table to associate their > mailboxes, which i dont have since those users are stored in the > subscriber table of kamailio. So am i still able to configure > voicemail like you are doing it by syncing with the voicemail table?, > i really hope so haha I forgot that fact. So yes I have a realtime sip users list (with host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio username, no password (this machine is not directly accessible from outside)) You might not be able to have endpoints able to subscribe to notifications due to this. I baked something inspired by: http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… that appears to work for me.

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dispatcher module not fetching attrs col from DB
by Jayesh Nambiar 05 Jul '16

05 Jul '16

Hi, I'm trying to use the dispatcher module with algo 9 for weighted distribution. I have a column attrs in the dispatcher table with value in the format "weight=90" for my destination addresses. But somehow the ds_list command always shows attrs as blank. I checked that kamailio doesn't fetch the attrs column at all; neither while starting up nor while calling ds_reload. This is the query that kamailio sends to my DB while starting up or calling the fifo ds_reload: select `setid`,`destination`,`flags`,`priority` from `dispatcher` I dont even see an attr column modparam in the documentation although kamailio didnt throw an error when I defined attr_col modparam for dispatcher module. Is there a different way to tell kamailio to fetch the attrs from the DB?? I'm using the 4.4.1 version. Thanks for any suggestion or pointers to solve this. - Jayesh

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Kamailio v4.2.8 Released
by Daniel-Constantin Mierla 05 Jul '16

05 Jul '16

Hello, Kamailio SIP Server v4.2.8 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.7. 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 v4.2.8 (or to 4.3.x/4.4.x series). Important: this version marks the end of planned releases from branch 4.2. From now on the focus is on maintaining the branches 4.4 and 4.3 for stable releases. For more details about version 4.2.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/07/kamailio-v4-2-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.4, at this moment with its latest release v4.4.2. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

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[SR-USERS] Kamailio issues with Asterisk Voicemail
by TEG AMJG 05 Jul '16

05 Jul '16

Dear list: I am quite new to Kamailio and i am really struggling to get some answer to some questions i have been trying to get answers by reading documentation and other questions or solutions around internet. Even though i am integrating Kamailio with two Asterisk boxes with the module dispatcher, i didnt follow the quite popular guide of Asterisk Kamailio Realtime Integration ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). The reason why i didnt is that it actually behaved weird since i wanted Kamailio to be a stateful proxy and also wanted Kamailio to act as a Registrar Server and Location Server using subscriber and location tables of kamailio database, keeping both Asterisk strictly as Media Servers. Also i have been able to solve some NAT Traversal issues using RTPProxy and not using those Asterisk boxes with their nat or qualify parameters to solve NAT issues. Most of my solution started following this guide ( http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html) which let me handle registration and location with Kamailio and also changed it putting kamailio behind NAT. Now to the real question, is there anyway to include Asterisk voicemail functionality with the solution I already mentioned? the question is because, as I far as I know, Asterisk needs to associate mailboxes to its users. So i believe that i have to practically change most of my solution and find a way to make Kamailio register and locate users with the Realtime integration of Asterisk and map the registration accordingly to the table fields. But it destroys most of my NAT and register solution since i also configured one of the private IPs as a peer in Asterisk sort of like in that guide [Kamailio] type=friend host=192.168.1.244 port=5060 disallow=all allow=gsm allow=g729 allow=alaw allow=ulaw Sorry for making the question so long and to be quite ignorant with some of the concepts involving Kamailio with Asterisk Thanks Alejandro

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Increase diversion counter
by Igor Potjevlesch 05 Jul '16

05 Jul '16

Hello, I'm looking for the best way to increase the Diversion counter. If my Kamailio is the first to generate a call forward I can manage this with static value into suffix params. But, if the Diversion counter is already set, I don't see how to handle this properly. Someone has already work on this? Regards, Igor.

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msgpack/format - new module?!
by Anonim Stefan 05 Jul '16

05 Jul '16

Hi, I'm thinking of creating a new module called "msgpack" or "format" which will export config msgpack_serialize() and msgpack_deserialize() for a given input. Is a new module the way to go, or do you think of a better place for this? Thanks, Stefan

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Releasing Kamailio v4.2.8
by Daniel-Constantin Mierla 05 Jul '16

05 Jul '16

Hello, I am planning to release tomorrow the v4.2.8 out of branch 4.2 to mark the end of official packaging for 4.2.x series. From now on, the focus will be 4.4.x and 4.3.x as stable branches. I just backported what I found relevant from branch 4.3 to 4.2 -- if anyone finds some other commits that need to be backported, reply to mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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register_dlgcb usage example
by Alexandr Dombrovsky 05 Jul '16

05 Jul '16

Hi! Where I can find any example of usage for this function: *register_dlgcb ?*

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Modify VIA header in replied message
by Huynh Quoc Bao 05 Jul '16

05 Jul '16

Hi all, As title, i want to add some additional parameters to the topmost via header in replied message. example. REGISTER ==> Kamailio ===> reply 200 OK. So in 200 OK message, i will add some additional parameters. And the same for the other message: INVITE, SUBSCRIBE, NOTIFY, MESSAGE message. -- Brs/Bao

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INFO: <core> [main.c:809]: sig_usr(): signal 15 received
by Aqs Younas 04 Jul '16

04 Jul '16

Greetings Again List. After being unable to use ndb_cassandra with kamailio I tried to use app_python. Below is relevant configuration. version: kamailio 4.4.2 (x86_64/linux) 892ad6 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, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 07:50:03 Jul 1 2016 with gcc 4.9.2 modparam("app_python", "script_name", "/usr/local/kamailio/etc/kamailio/handler.py") request_route { ..... python_exec("handler"); ..... } And i am using example provided in app_python module. *But i soon as i import Cassandra driver in script*. Kamailio terminate with there logs. Jul 1 19:57:19 debian ./kamailio[8643]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(20) Jul 1 19:57:19 debian ./kamailio[8642]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(19) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: tmx Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: tmx [tmx_mod.c:275]: child_init(): rank is (-4) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: sl Jul 1 19:57:19 debian ./kamailio[8616]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Jul 1 19:57:19 debian ./kamailio[8642]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa78988 and 0xabc988 Jul 1 19:57:19 debian ./kamailio[8642]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa787c0, 40, 1, (nil)), fd_no=0 Jul 1 19:57:19 debian ./kamailio[8644]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 21: kex Jul 1 19:57:19 debian ./kamailio[8645]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (22) Jul 1 19:57:19 debian ./kamailio[8640]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (17) Jul 1 19:57:19 debian ./kamailio[8639]: DEBUG: mi_rpc [mi_rpc_mod.c:104]: child_init(): initializing child[-2] for rpc handling Jul 1 19:57:19 debian ./kamailio[8646]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 23: tm Jul 1 19:57:19 debian ./kamailio[8641]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 18: corex Jul 1 19:57:19 debian ./kamailio[8635]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(-1) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: usrloc Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD Jul 1 19:57:19 debian ./kamailio[8636]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8621]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8620]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8625]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8623]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8638]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8624]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8632]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8647]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8633]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8637]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8619]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8622]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8626]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8627]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8628]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8629]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8642]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8630]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8635]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8631]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8634]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done Jul 1 19:57:19 debian ./kamailio[8616]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc I have tried using two different drivers for Cassandra in python but same results. I wonder if i import mysql driver kamailio starts perfectly. Below it my script. #!/usr/bin/python import sys import pymysql *import pycassa* from Router.Logger import LM_ERR from Router.Logger import LM_INFO class test: def __init__(self): LM_INFO('test.__init__\n') def child_init(self, y): LM_INFO('test.child_init(%d)\n' % y) return 0 def handler(self, msg): LM_INFO('test.handler(%s)\n' % (msg.Type)) if msg.Type == 'SIP_REQUEST': if msg.Method == 'INVITE': LM_INFO("Re-writing RURI") msg.rewrite_ruri('sip:0022@192.168.2.24:5073') LM_INFO('SIP request, method = %s, RURI = %s, From = %s\n' % (msg.Method, msg.RURI, msg.getHeader('from'))) LM_INFO('received from %s:%d\n' % msg.src_address) return 1 def mod_init(): return test() Any help is much appreciated. Best Regards, Aqs Younas

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Error while reading from Cassandra Cluster
by Aqs Younas 04 Jul '16

04 Jul '16

Greetings Lists. I am having hard time trying to use ndb_cassandra with kamailio. I have successfully complied the module and kamailio runs without any error. Below is relevant snippet. loadmodule "ndb_cassandra.so" modparam("ndb_cassandra", "host", "192.168.0.103") modparam("ndb_cassandra", "port", 9160) $var(keyspace) = 'kamailio'; $var(column_family) = 'numbers_directory'; $var(key) = 15; $var(column) = 'number'; xlog("L_INFO", "Valus to fetch are $var(keyspace), $var(column_family), $var(key), $var(column) \n"); * if (cass_retrieve("$var(keyspace)", "$var(column_family)", "$var(key)", "prefix", "$var(val_read)") > 0) {* xlog("L_INFO", " ===== ndb_cassandra. Sucess while reading from Cassandra. val_read: \"$var(val_read)\"\n"); } else { xlog("L_INFO", " ===== ndb_cassandra. Error while reading from Cassandra \n"); } Cassandra Table Structure. cqlsh:kamailio> describe numbers_directory; CREATE TABLE kamailio.numbers_directory ( id int PRIMARY KEY, description text, number text ) WITH bloom_filter_fp_chance = 0.01 AND caching = '{"keys":"ALL", "rows_per_partition":"NONE"}' AND comment = '' AND compaction = {'class': 'org.apache.cassandra.db.compaction.SizeTieredCompactionStrategy'} AND compression = {'sstable_compression': 'org.apache.cassandra.io.compress.LZ4Compressor'} AND dclocal_read_repair_chance = 0.1 AND default_time_to_live = 0 AND gc_grace_seconds = 864000 AND max_index_interval = 2048 AND memtable_flush_period_in_ms = 0 AND min_index_interval = 128 AND read_repair_chance = 0.0 AND speculative_retry = '99.0PERCENTILE'; cqlsh:kamailio> select * from numbers_directory; id | description | number ----+-------------+----------- 15 | null | 212456789 When i make a call i get this in logs. Jul 1 19:10:42 debian ./kamailio[8142]: INFO: <script>: Valus to fetch are kamailio, numbers_directory, 15, number Jul 1 19:10:42 debian ./kamailio[8142]: INFO: <script>: ===== ndb_cassandra. *Error while reading from Cassandra* I am stuck at this. Does anybody able to use kamailio with cassandra or can share their configuration? Any pointer on this is much appreciated. Best Regards, Aqs Younas

1 0

Kamailio and Mongodb
by C. 04 Jul '16

04 Jul '16

Hi, I have followed instructions on setting up Kamailio with MongoDB backend, and successfully compiled and installed kamailio with the db_mongo module. It can start successfully. However, when I tried to add users using the command: kamctl add test testpasswd test(a)mysipserver.com <mailto:test@mysipserver.com> It fails because it is trying to default to MYSQL. Here is the result of that command: WARNING: database engine not found - tried 'MYSQL' ERROR: This command requires a database engine - none was loaded When I take a look at the kamctl script, it lists all the other kinds of database engines, except MongoDb, so this is probably why it is failing. How can I use kamctl to create the required Mongodb collections (like users, subscribers, etc) that Kamaiio will then use?My kamailio.cfg seem to be working ok, as I seem to be able to connect to the database (I modified the file to use db_mongo module), and I create the initial collections as listed here - https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb <https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb> Thanks, Chuks.

2 1

Regarding config-engines
by Alexander Rudenko 04 Jul '16

04 Jul '16

Hi, Question about future config-engines. Will it be possible to use connectors/drivers via the engine script and is it a good Idea? E.g. I have specific database and python driver for that database. May I use simple 'import dbdriver', create connection to the db, use it inside python engine script, etc? In general, in case there is no kamailio connector module for scpecific database, I would use python connector. Thanks in advance! -- *Alexander Rudenko*

2 1

re-Invite in the failure route
by Jay Li 04 Jul '16

04 Jul '16

Dear All, Hopefully I'll be able to get some help here on re-INVITE in the failure route. For example I have a couple of PSTN gateway options. I'd like to try them by priorities. In case the one with top priority couldn't go through(e.g. returned 5XX on INVITE), I'd like to re-INVITE through another provider in the failure route. Not sure what I should do for such cases. Thanks. Regards,Jay

2 3

Issue loading ims_dialog module
by Vikram Chhibber 01 Jul '16

01 Jul '16

Hi All, I am trying to bring-up PCSCF on releae 4.4 and I am getting errors related to loading of ims_dialog module. The mpath is set correctly in kamailio.cfg and ims_dialog.so is present in that path. Interestingly, it is able to load rest of the modules from the same locaiton. ls -lrt /usr/local/lib64/kamailio/modules/ims_dialog.so -rwxr-xr-x 1 root root 953760 Jun 29 16:37 /usr/local/lib64/kamailio/modules/ims_dialog.so Any pointers will be appreciated. -- Unit kamailio.service has begun starting up. Jun 29 17:08:17 kamailio1 kamctl[2914]: ** INFO: Starting Kamailio : Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [sr_module.c:582]: load_module(): could not open module </usr/local/lib64/kamailio/modules/ims_dialog.so Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 169, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 313, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 314, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 315, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 316, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [cfg.y:3228]: yyparse(): cfg. parser: failed to find command set_dlg_profile (params 1) Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 1009, colu Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [cfg.y:3228]: yyparse(): cfg. parser: failed to find command set_dlg_profile (params 1) Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 1077, colu Jun 29 17:08:17 kamailio1 kamailio[2939]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 29 17:08:20 kamailio1 kamctl[2914]: ** ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed Jun 29 17:08:20 kamailio1 systemd[1]: kamailio.service: Control process exited, code=exited status=1 Jun 29 17:08:20 kamailio1 systemd[1]: Failed to start Kamailio SIP Server. -- Subject: Unit kamailio.service has failed Thanks

2 4

Distributed Authentication
by Colin Morelli 01 Jul '16

01 Jul '16

Hey all, I'm running a cluster of Kamailio instances as a proxy/registrar for another cluster of Freeswitch instances. I'm using http_async_client to make HTTP queries to my API to fetch credentials on auth challenges. Kamailio performs generating the header, and validating the result based on the data provided from my API. I'm fairly sure the answer is no, but I was wondering if Kamailio has any mechanism for getting access to the nonce/nc values in the challenges and responses so I can store them somewhere accessible to the whole cluster. Because my instances are transaction stateful, the request that is challenged and the subsequent request with the response may be routed to different instances and I want to validate the nonce correctly. I can move all of this into the API (the digest auth and verification), but my next question would be whether or not there are any APIs for getting access to this information in a structured format, or if I should just shove the whole digest auth header in the request to my API and parse/verify there. Thanks in advance. Best, Colin

2 1

Serialize Branches Based on q value
by John Petrini 01 Jul '16

01 Jul '16

Hello, I'm trying to append a branch, assign it a q value so it has priority and serialize the new branch and the main branch. The goal is to send the new branch off to our CNAM provider, capture the CNAM in their reply to a $var then drop the branch, stick the cnam in the main branch and allow the call to continue as normal. We do this currently using OpenSIPS and the serialize_branches(1) but the of load_contacts() seems to be different than serialize_branches. route[CNAM_DIPS] t_on_branch("CNAM_DIPS"); t_on_reply("CNAM_DIPS"); t_on_failure("CNAM_DIPS"); t_load_contacts(); t_next_contacts(); append_branch("sip:2223334444@22.33.444.55:5600;trans-type=5", "0.5"); relay(); break; Two problems. First when t_next_contacts is called I get the following error: ERROR: pv [pv_branch.c:58]: pv_get_branchx(): error accessing branch [0] Second problem is that my call goes out on the branch I appended and I get a reply from the CNAM provider but Kamailio forwards that reply back to the phone causing the phone to send a second invite. In OpenSIPs we just called drop, to drop the branch and let the main branch continue but this doesn't seem to work in Kamailio. I've read and reread the documentaion but it doesn't seem to behave the way the documentation says it should. Is there something I'm missing? Is there a better way to accomplish this in Kamailio? Thank You, ___ John Petrini

1 0

Dialplan module doubt
by José Seabra 01 Jul '16

01 Jul '16

Hello there, I'm evaluating if I can use dial-plan and dispatcher module to build a kamailio routing server to route calls based on number dialed(sometimes the number dialed can be Alphanumeric). I'm expecting have lot of dial-plan entries, some of them with manipulations, others without. Anyone can see here any limitation on this implementation? One thing that I have noticed is that dial-plan module enforces me to fill the fields substr_exp and repl_exp even if i don't have any manipulations to apply. I'm also concerning about performance issues with lot of dialplan entries. What are your thoughts about this solution? Thank you -- José Seabra

2 6

ACK / BYE transaction problem
by Oliver Roth 01 Jul '16

01 Jul '16

Hi all Follow scenario Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> gateway kamailio [gw] --> carrier [carr] I get Invites from [c5] with Request ,To, from, contact, pid in national format 0794445566 [lbl] dispatches this to [gw] For the [carr] I need international format. So doing these transactions in [gw] And sending to [carr] in international format Request, to, from, contact, ... => 417794445566 Everything ok Then I get a 100, 183 and even 200 from [carr] Ack is coming from [c5] to [lbl] and [gw] - but then it stocks The ACK is not sent to the [carr] I kamailio log I see DEBUG: RFC3261 transaction matching failed DEBUG: t_lookup_request: no transaction found So for me, the ACK cannot be assigned to a transaction and gets discarded by if ( is_method("ACK") ) { xlog(,"L_INFO", "WITHINDLG ACK - not loose route\n"); if ( t_check_trans() ) { xlog(,"L_INFO", "WITHINDLG ACK - t_check_trans() \n"); # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { xlog(,"L_INFO", "WITHINDLG ACK - not t_check_trans() DISCARD!!\n"); # ACK without matching transaction ... ignore and discard route(NATMANAGE); #t_relay(); #exit; Any idea? Problem with modifying the sip tags? Or problem with the dialog? Thanks for helping OIi

3 6

Kamailio Iterate All Headers
by Colin Morelli 01 Jul '16

01 Jul '16

Hey all, Back with more questions. I'm using Kamailio to make an HTTP call to my API to perform authentication and message routing. Currently, I'm trying to build up the post body that I send to my API to make those decisions. I've cherry picked a few of the headers that are important in my routing decisions. But, ideally, I'd like to just iterate over all the SIP message headers and append them as request parameters to my API call. Is this possible with Kamailio? I've been looking through the docs and can't seem to find a function to iterate the full list of headers. Thanks in advance. Best, Colin

4 6

Get registered user list
by Никита Лебедев 01 Jul '16

01 Jul '16

Good day! Can anyone point me to right direction? The question is - I want to get list of registered users (to find out who is online at the moment) from sip server. I'm using kamailio on server side and exosip library on client side. I tried to find something in RFC 3261 and failured... Can someone help and tell what to do or where to look for the answer?

3 3

P-Asserted-Identity - Pseudo-Variables are set as null
by José Seabra 30 Jun '16

30 Jun '16

Hello there, The Pseudo-Variablest hat store information about P-Asserted-Identity are all as set null: $pU=<null> $pd=<null> $pn=<null> Is there any known reason for that? The kamailio version that I'm using this 4.4.1 Thank you for you support Best regards -- José Seabra

3 3

Kamailio Load Balancer Architecture
by Daryn Johnson 30 Jun '16

30 Jun '16

I am interested in opinions and suggestions about load balancing with Kamailio. I work for an ITSP that currently uses Oracle & Broadsoft, and I am working to design and develop an open source solution using Kamailio (Proxy, Registrar, LB) & other Application/Media servers for more flexibility and freedom :) Thank you ALL for the work you have put in on Kamailio! After much reading and configuration I have a Kamailio 'proxy' setup, with endpoints registered using the Registrar module, and calls being sent/received to/from the PSTN. I am interested in separating the Load Balancers from the Registrars & logic, for security and in order to be able to scale appropriately. I will be using the dispatcher module for both load balancers and proxies using the following architecture: PublicIP = 5.5.5.5; Private IP = 192.168.1.0/24 USERS (Public Internet) ==> (public: 5.5.5.5) [ Kamailio (LoadBalancer, Firewall, Sanity Checks) ] (core:192.168.1.2) ==> [ Kamailio Registrar, Proxy, PSTN GW ] ==> AppServers or PSTN GW (we can access our PSTN gateways Via our Core using Private IPs) Questions: 1) Is it overkill to separate the LB & Proxy/Registrars? 2) Is this a common architecture & anyone configured this architecture successfully? Thanks in advance for your help! *Daryn Johnson* *Senior VoIP Engineer*

4 8

Kamailio v4.3.6 Released
by Daniel-Constantin Mierla 30 Jun '16

30 Jun '16

Hello, Kamailio SIP Server v4.3.6 stable release is out. This is a maintenance release of the previous stable branch, 4.3, that includes fixes since release of v4.3.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.6 (or to 4.4.x series). For more details about version 4.3.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/06/kamailio-v4-3-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.4, at this moment with its latest release v4.4.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

Releasing Kamailio v4.3.6
by Daniel-Constantin Mierla 30 Jun '16

30 Jun '16

Hello, as announced last week, I am planning to release a patch update from branch 4.3, respectively v4.3.6, tomorrow. Today I backported what I found to be relevant from branch 4.4 to 4.3. If anyone notices something missing, reply on the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

ipsec support in PCSCF
by Vikram Chhibber 30 Jun '16

30 Jun '16

Hi All, I did not get any concrete information on whether PCSCF supports transport mode ipsec in Kamailio release 4.4. It is mentioned that it is not supported in release 4.0. Could anyone please let me know the status on this? Thanks, Vikram

3 4

kamailio & redis
by Fabian Pignataro 29 Jun '16

29 Jun '16

Hello Community, I'm trying to read data (hash) stored in redis database, from kamailio script. I read the module documentation ndb_redis: http://www.kamailio.org/docs/modules/4.4.x/modules/ndb_redis.html#idp263007… But I'but I don't understand what I'm doing wrong :/ With someone has had experience this? *My platform:* *root@kamailio-fts:~# cat /etc/issue* *Ubuntu 14.04.4 LTS \n \l* *root@kamailio-fts:~# kamailio -V* *version: kamailio 4.4.1 (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, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES* *ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB* *poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.* *id: unknown * *compiled with gcc 4.8.2* *My script:* *modparam("ndb_redis", "server", "name=srvN;addr=127.0.0.1;port=6379;db=1")* *modparam("ndb_redis", "init_without_redis", 1)* *if(redis_cmd("srvN", "HMGET usuario:2002 status", "res")) {* *xlog("L_INFO","array size: $redis(res=>size) - type: $redis(res=>type) - value: $redis(res=>value[0]) $redis(res=>value[1]) \n"); * * }* *The output of syslog (I don't see errors): * *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7fc2fd008a30* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at 0x7fc2fd007020* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set at 0x7fc2fd008a98* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: db_postgres [km_dbase.c:433]: db_postgres_free_query(): PQclear(0xcc9060) result set* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [socket_info.c:564]: grep_sock_info(): checking if host==us: 13==13 && [172.16.20.219] == [172.16.20.219]* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0x7fc2fcfc09a0* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: registrar [lookup.c:240]: lookup_helper(): contact for [2004] found by address* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: INFO: <script>: es un mensaje y estoy por enviarlo* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: INFO: <script>: array size: 2 - type: 2 - value: <null>* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil)* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=36630, isACK=0* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG: trans=0x7fc2ed762d08, callback type 1, id 0 entered* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: d79552e0dafc752469a90c3599ba2d42* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_funcs.c:368]: t_relay_to(): SER: new transaction fwd'ed* It can be seen that $redis(res=>size) & $redis(res=>type) return values, not so with $redis(res=>value[0]). *By last, I share de output of redis CLI:* *127.0.0.1:6379 <http://127.0.0.1:6379>> hget usuario:2002 status * *"Online" * Cheers, Fabián - -- Ing. Fabián Pignataro

2 4

rtpengine allow_op problem
by david 29 Jun '16

29 Jun '16

hello all i have been taking a look to the rtpengine.c code and i found that the issue has relation with the queried_nodes_limit which had 50 defalut value, so before it was reached in the loop, we received a request/response retransmision doing all the same. anycase, the issue it's because when allow_op is applied, the select_rtpp_node gets a node even having the node disabled by the first attemp (having the rtpengine node down), and it does the loop. if it's not applied, we dont get a node at the retry. but adjusting the number of queried_nodes_limit to a value to be reached before a request retransmission arrives, i guess it should be enough (¿?) best regards david escartin

2 2

【Issue of kamailio with pgpool】
by Noriyuki Hayashi 29 Jun '16

29 Jun '16

Hi, I am testing on CentOS 6.6 x86.64 kamailio-4.3.5 postgresql-9.4.6 pgpool-II-pg94-3.5.2 When I using floating IP as 192.168.192.143 that kamailio is working. But I could not access to psql -h 192.168.192.143 -p 9999 -U postgres asterisk. Let me know if you have this issue with much appriciated. Thank you. #!define DBURL "postgres://asterisk:password@192.168.192.143:9999/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "postgres://asterisk:password@192.168.192.143:9999/asterisk" #!endif #!endif Noriyuki Hayashi Japan.

2 2

rtpengine allow_op problem
by david 29 Jun '16

29 Jun '16

hello all i have observed an issue using this parameter, which is useful in case you deactivate a node from the rtpengine sets, to let the calls to finish the negotiation. but i have tested that if a node fails down in the middle of a call, with that option activated, the kamailio gets blocked waiting until the rtpengine node is again up, to finish handling with the BYE request for instance. Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> could it be possible to make the kamailio leave that blocking state, or at least be able to activate the allow_op param by kamctl fifo? thanks a lot and regards david escartin

1 0

Documentation DMQ module
by Jan Hazenberg 29 Jun '16

29 Jun '16

Hi, Found a error in the documentation of the DMQ module (http://kamailio.org/docs/modules/devel/modules/dmq.html#dmq.p.num_workers). In the documentation there is a example: modparam("dmq", "num_threads", 4) If i'm correct this should be: modparam("dmq", "num_workers", 4) Thanks, Jan

2 1

rtpengine allow_op issue
by david escartin 28 Jun '16

28 Jun '16

hello all i have observed an issue using this parameter, which is useful in case you deactivate a node from the rtpengine sets, to let the calls to finish the negotiation. but i have tested that if a node fails down in the middle of a call, with that option activated, the kamailio gets blocked waiting until the rtpengine node is again up, to finish handling with the BYE request for instance. Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> could it be possible to make the kamailio leave that blocking state, or at least be able to activate the allow_op param by kamctl fifo? thanks a lot and regards david escartin

1 0

Kamailio - PSTN (kamailio) - rtpproxy issue
by Filippo Graziola 28 Jun '16

28 Jun '16

Hello everybody, I have a local kamailio+rtpproxy under NAT listening on multiple local interfaces (vlans) which works fluenty for all local user indipendently and engages rtpproxy correctly. I have another working kamailio+rtpproxy which is accessible from everywhere at a public address and I would like to route to it some specific calls from the local server. So, in the local kamailio configuration file I defined WITH_PSTN (all the necessary option) and INVITE/OK/BYE messages are going correctly to the destination. The problem is that I have no audio on both ways. After some packet sniffing, I tried killing local rtpproxy process and now calls to PSTN have also audio both ways. >From these tests, if my configuration is correct, when routing the call to PSTN gateway I should disable the local rtpproxy, and I'm in doubt about how I can do this. Could you help me? Should I change NAT flags in the PSTN route? Thanks

1 2

Kamailio v4.4.2 Released
by Daniel-Constantin Mierla 28 Jun '16

28 Jun '16

Hello, Kamailio SIP Server v4.4.2 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.2. For more details about version 4.4.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/06/kamailio-v4-4-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Enjoy the summer holidays! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

acc time mode 2 wrong timestamp
by pars3c 28 Jun '16

28 Jun '16

Hello, i've installled the new version of kamailio 4.4.1 . in 2 different server , with acc module and these parameters: modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "seconds") should be seconds.millisec On the first server in acc table , in field seconds i have values like this 1466493457.85, this is correct timestamp ( this server have centos 86_64) In the other server , i have values like 1913113.702 , that is wrong timestamp . (this is old centos 32 bit ) What can be the problem? The different version 32,64 bit ? or some compiler option ? Thanks

2 11

Planning release of v4.4.2 and other v4.x.y
by Daniel-Constantin Mierla 28 Jun '16

28 Jun '16

Hello, I am considering to release kamailio v4.4.2 next week along with the corresponding patch releases for branches 4.3 and 4.2. It will likely be the last release in series 4.2 to mark its end of official packaging. I would also consider a new patch release in 4.1.x, if that is still used by many out there to get packages that fix the libssl version matching -- Victor should say if the packages for this branch can still be generated in order to make sense doing this release. Anyhow, report any issues you are aware and are not yet announced for branches 4.4 and 4.3 to the bug tracker as soon as possible to get a chance to be fixed by release date. Likely 4.4.2 will be next Tuesday, June 29, with the other branches done in the following days. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

userblacklist use avp for check_blacklist
by Jan Hazenberg 28 Jun '16

28 Jun '16

Hello, I'm trying to use the userblacklist module to implement a global blacklist. I have a few questions about the module. Is it possible to use a avp for the check_blacklist function? If i'm correct it will only parse the request URI, but i'm rewriting the URI in a branch route to allow different formats to different destinations. Or is this only possible with the check_user_blacklist function? I also have a issue with numbers in E164 format with a leading +. I tried to escape this with a \ in the database but it will not load the entry's when i use the + in front of the number. Is the + supported by this module or should i strip it before the function and add it again afterwards? Thanks, Jan

2 12

cgrates - cdrs_tables.sql error
by Infinicalls Infinicalls 27 Jun '16

27 Jun '16

Hi, I am trying to install cgrates , but face this issue with database. " ERROR 1067 (42000) at line 7: Invalid default value for 'deleted_at' " is encountered in create_cdrs_tables.sql . Can somebody let me know how to overcome this? regards Ganesh Kumar

2 1

Re: [SR-Users] S-CSCF (kamailio) crashes after deregistration of client
by Christoph Valentin 27 Jun '16

27 Jun '16

2 1

behaviour of uac_redirect module
by Dirk Teurlings - Signet B.V. 27 Jun '16

27 Jun '16

Hi, Testing with latest stable 4.4 Kamailio at the moment, and running into the following issue. I have a setup that sends a 302 to do a Moved Temporarily to another extension. To update the Contact header accordingly, I use the get_redirects function in the failure_route. get_redirects('*', 'Redirects') This results in the following packet U 192.168.10.246:5060 -> 192.168.10.245:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 192.168.10.245:5060;branch=z9hG4bK40b58e09;rport=5060;received=192.168.10.245. From: "+31........" <sip:+31........@192.168.10.245>;tag=as42745abe. To: <sip:+31........@192.168.10.246>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f67d. Call-ID: 576349c40a7cb77a06230e181831672d@192.168.10.245:5060. CSeq: 102 INVITE. Contact: <sip:06........@sip.example.net>, <sip:06........@sip.example.net>;q=0.01. Server: sip.example.net. Content-Length: 0. . This works, but I don't want ACC to account for anything, so I tried: get_redirects('*') But this results in the following packet U 192.168.10.246:5060 -> 192.168.10.245:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 192.168.10.245:5060;branch=z9hG4bK185f4fdd;rport=5060;received=192.168.10.245. From: "+31........" <sip:+31........@192.168.10.245>;tag=as13f23689. To: <sip:+31........@192.168.10.246>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-b5fe. Call-ID: 336a16912db412d418f7f7495e2574f5@192.168.10.245:5060. CSeq: 102 INVITE. Contact: <sip:accountname@sip.example.net>, <sip:06........@sip.example.net>;q=0.01, <sip:06........@sip.example.net>;q=0.01. Server: sip.example.net. Content-Length: 0. . Notice the extra accountname Contact, this results in a forward that doesn't work for the mediaserver. To me the documentation about the function states that there should be no difference between using it with or without accounting as far as SIP messages go. Is this a bug, or is this by design? Cheers, Dirk

2 4

Question about gzcompress module
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, When using the gzcompress module, is it possible to compress/decompress only one specific header, or will it compress/decompress the whole body? Thanks, Joel.

2 2

Get full body from http POST request
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, I'm using http_query (POST method) to interact with an API. This API responds with a JSON. Is there a way to get the full body and after parse that JSON? Example response: { "error": { "code": 404, "message": "Destination not Found" } } Currently, I'm only being able to get the first line, in the previous example it would be: "{" According to the doc, that is the expected behavior: If HTTP server returns a class 2xx, 3xx or 4xx reply, the first line of the reply's body (if any) is stored in “result” parameter, which must be a writable pseudo variable. I don't have the ability to get the response in 1 single line like: { "error": { "code": 404, "message": "Destination not Found" } } How can I get the full body instead of only the first line? Thanks. Best regards, Joel.

1 1

Parsing diversion headers correctly
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, If you receive an INVITE with 2 diversion headers such as (in this order): Diversion: <sip:9975@X.X.X.X:5060>;privacy=off;screen=no; reason=user-busy; counter=1 Diversion: <sip:1234567890@Z.Z.Z.Z:5060>;privacy=off;screen=no; reason=unknown; counter=1 How is a nice way of extracting the user part from the URI of the last one? (in this case: 1234567890) Problems I have found: $(di{uri.user}) ---> Give mes the user part of the URI of the first header, in this case: 9975 Is there a way to tell $di to get its value from the last header instead of the first one? This example works: $(hdr(Diversion)[-1]{param.name,0}{s.strip,1}{s.striptail,1}{uri.user}) But I think it is very hacky... Any better approach anyone can suggest me? Thanks in advance. Best regards, Joel.

2 2

Public vs Internal IPs
by Colin Morelli 25 Jun '16

25 Jun '16

Hey all, When using Kamailio at the edge - what's the best practice around how to advertise your Record-Route? I assume it's going to involve the use of a double-RR with both the public and private IPs. However, I'm running in AWS where the host doesn't have two interfaces with both a public and private IP. So, I set the host's public IP as the advertised address while Kamailio binds to the local IP. But now how do I get Kamailio to know that it needs to double-RR in this case? Thanks, Colin

2 9

Specify alternative OpenSSL engine for Kamailio
by Joel Serrano | VOZELIA 25 Jun '16

25 Jun '16

Hi, This is a similar question on a previous thread: "[SR-Users] Compiling kamailio with custom openssl" Is it possible to specify a different OpenSSL engine for kamailio to use? For example: On nginx you have the config param: ssl_engine <engine> On apache you have the config param: SSLCryptoDevice <engine> Is there anything similar on Kamailio? We are using debian 8 with openssl-1.0.1t and these are the available engines: root@debian:/usr/lib/x86_64-linux-gnu/openssl-1.0.0/engines# ls -l total 404 -rw-r--r-- 1 root root 19512 Jun 21 11:07 lib4758cca.so -rw-r--r-- 1 root root 19784 Jun 21 11:07 libaep.so -rw-r--r-- 1 root root 15576 Jun 21 11:07 libatalla.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libcapi.so -rw-r--r-- 1 root root 24232 Jun 21 11:07 libchil.so -rw-r--r-- 1 root root 19864 Jun 21 11:07 libcswift.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libgmp.so -rw-r--r-- 1 root root 93304 Jun 21 11:07 libgost.so -rw-r--r-- 1 root root 15432 Jun 21 11:07 libnuron.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libpadlock.so -rw-r--r-- 1 root root 24096 Jun 21 11:07 libsureware.so -rw-r--r-- 1 root root 19784 Jun 21 11:07 libubsec.so root@debian:/usr/lib/x86_64-linux-gnu/openssl-1.0.0/engines# The reason is we have another server with a Cavium SSL card, which provides a dynamic engine: -rw-r--r-- 1 root root 127968 Jun 23 15:18 libcavium.so If it is not possible via config parameter, is there a way to specify the openssl engine to use before compiling? Thanks in advance. Best regards, Joel.

2 2

Count number of headers of type
by Joel Serrano | VOZELIA 25 Jun '16

25 Jun '16

Hi, It there an easy way to count the number of existing headers of certain type? For example: - How many 'via' headers: $hdrcount(Via) ? - How many 'diversion' headers: $hdrcount(Diversion) ? The $hdrcount is an example, just wondering if there is an easy way to get the number. Thanks. Best regards, Joel.

2 2

t_on_failure route not being called
by Colin Morelli 24 Jun '16

24 Jun '16

Hey all, I'm using a combination of http_async_client and rtjson to query my API and retrieve a JSON target route set for an incoming SIP request that Kamailio will forward to. The HTTP portion of it works great. I'm able to hit my API and get back a JSON document. Additionally, the rtjson parsing appears to work, because the request is hitting my freeswitch instance. However, FS is responding with a 503 (I'm intentionally doing this for testing), and Kamailio isn't calling my t_on_failure route. I've pasted the relevant portion of my config below - would be great if anyone has any insights. tl;dr: the "REROUTE" route section below isn't being reached even when FS responds with a 503. Best, Colin route[ROUTING_REPLY] { if $http_ok && $http_rs == 200 { xlog("L_INFO", "$ci|log|loaded target route set from document $http_rb\n"); rtjson_init_routes("$http_rb"); rtjson_push_routes(); xlog("L_INFO", "$ci|log|attempting to relay request towards $du\n"); t_on_failure("REROUTE"); t_set_fr(0, 1000); t_relay(); } else { send_reply("480", "Temporarily Unavailable"); } } route[REROUTE] { xlog("L_WARN", "$ci|log|relay to destination $du failed, checking next route\n"); if (rtjson_next_route()) { xlog("L_INFO", "$ci|log|additional route found, directing request to $du\n"); t_on_failure("REROUTE"); t_set_fr(0, 1000); t_relay(); } else { xlog("L_WARN", "$ci|log|route list exhausted, failing request\n"); send_reply("480", "Temporarily Unavailable"); exit; } }

1 1

S-CSCF (kamailio) crashes after deregistration of client
by Christoph Valentin 24 Jun '16

24 Jun '16

4 12

Kamailio behind ELB AWS
by yann christophe 24 Jun '16

24 Jun '16

Hello Everyone, I would like to know, if i can use ELB and route53 behind some Kamailio servers for the high availibility ? I use Kamailio behind Freeswitch. I would like deploy these servers on AWS. What do you suggest me for the high availibility on AWS with Kamailio. Thanks Best regards Yann

2 1

rtpengine installation delete ethernet interface on Centos
by Yasin CANER 24 Jun '16

24 Jun '16

1 0

How to understad that user is de-registrating
by Alexandru Covalschi 24 Jun '16

24 Jun '16

Hello list, I need to send to an external API events when user is registrated and de-registrated. As far as I understand standart behaviour is as follows: If user is not registered, he sends REGISTER and he is registrated (I can catch that because I make the auth). If user is registered and sends REGISTER he is de-register. (Please correct me if I'm wrong.) How can I catch that? Can I use event_route[usrloc:contact-expired]? Thanks in advance! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator tel: +37367398493 web: http://abriss.solutions/ <http://abs-telecom.com/>

3 4

uacreg flags field not changing
by Yuriy Gorlichenko 24 Jun '16

24 Jun '16

Hi. Im using kamailio 4.4.1 kamailio -v 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, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 99e1c3 compiled on 13:34:38 Jun 14 2016 with gcc 4.9.2 I using uacreg table for register trunks and want to know their status For now i see status onlu from kamcmd uac.reg_dump/info But if i trying to take flags from uacreg table (flags field) it always 0.

1 0

Compiling kamailio with custom openssl
by Cibin Paul 24 Jun '16

24 Jun '16

Hi, Can any one guide me to compile kamailio with custom openssl path please. Thanks & Regards Cibin

2 12

Dialog timeout before answer
by Alex Balashov 24 Jun '16

24 Jun '16

Hi, Will the dialog timeout - and specifically, $dlg_ctx(timeout_bye) - also apply to dialogs which have not yet been established? If so, does that mean the dialog module will send a CANCEL instead for a dialog in an early/progress state? -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 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

kamailio + fax "other fax not responding"
by Mx R 23 Jun '16

23 Jun '16

Hi there, We use kamailio 4.3.3. And we have one problem with the fax. We send and receive from and to our network, the fax does not work. Fax reports "other fax not responding". When we send through our network to another operator they receive fax. What could be wrong?

2 5

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 23 Jun '16

23 Jun '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0

3 5

Dialog in delete state is too old
by Emilio Soler Muñoz 23 Jun '16

23 Jun '16

Hello! Kamailio process has stopped. Anyone else has happened? Versión: 4.4 This is what I have in the logs. kamailio.log: 2016-06-23T10:10:33.983494+02:00 kam01 /usr/sbin/kamailio[2907]: NOTICE: dialog [dlg_hash.c:245]: dlg_clean_run(): dialog in delete state is too old (0x7f092e15b190 ref 3) 2016-06-23T10:10:34.005015+02:00 kam01 /usr/sbin/kamailio[2897]: WARNING: dialog [dlg_handlers.c:1348]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7f092e15b190 [3554:5262] with clid 'L, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PU' and tags 'as6b8e8af9' 'as25960d84' 2016-06-23T10:10:34.007989+02:00 kam01 /usr/sbin/kamailio[2913]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 2016-06-23T10:10:34.008119+02:00 kam01 /usr/sbin/kamailio[2823]: ALERT: <core> [main.c:739]: handle_sigs(): child process 2897 exited by a signal 11 2016-06-23T10:10:34.008128+02:00 kam01 /usr/sbin/kamailio[2823]: ALERT: <core> [main.c:742]: handle_sigs(): core was not generated kern.log: [363406.951756] kamailio[2897] general protection ip:7f092c9413a5 sp:7ffdd1065530 error:0 in dialog.so -- Emilio Soler Muñoz <emiliosoler(a)gmail.com>

1 0

Add a user through a http request
by Jay Li 22 Jun '16

22 Jun '16

Dear All, I wonder if Kamailio has the functionality to add a subscriber dynamically. For example, if I use event_route[xhttp:request] to expect a http get request, then based on the received request, add a user in the DB. Not sure what the best approach would be. Any suggestion is appreciated. Thanks. Regards,Jay

2 1

Dispatcher - Route call to alternate server if there is delay to answered call
by Amit Patkar 22 Jun '16

22 Jun '16

Hi I have a scenario where dispatcher is used to balance calls. Assume there are 2 media servers M1 & M2. Call are routed using round robin algorithm. There is scenario where call is handed over to M1 but it is not answered in defined time period (say 5 seconds due to load on server or network congestion), I wish to send this call to M2. What are the options to achieve it? Thanks & Regards, Amit Patkar

2 1

Re: [SR-Users] uac.reg_refresh not removing records from memory
by DanB 22 Jun '16

22 Jun '16

Hey Daniel, Thanks for so fast feedback! No difference as far as I can tell before and after removing the record from db. I have also not disabled specifically the record inside uacreg table but completely removed it from db. I should mention that I have just upgraded to latest 4.4.1 so I should have the flags also saved in db. Here is the record before removal: """ root@iPBXDev1:/etc/ipbxrp# kamcmd -s tcp:127.0.0.1:2046 uac.reg_dump { l_uuid: rsua1_172.16.254.101 l_username: rsua1 l_domain: 172.16.254.101 r_username: cust_sua1 r_domain: 172.16.254.102 realm: auth_username: cust_sua1 auth_password: check123 auth_proxy: sip:172.16.254.102:5060 expires: 3600 flags: 20 diff_expires: 3588 timer_expires: 1466349066 reg_init: 1466345451 reg_delay: 0 } """ and the one after: """ root@iPBXDev1:/etc/ipbxrp# kamcmd -s tcp:127.0.0.1:2046 uac.reg_dump { l_uuid: rsua1_172.16.254.101 l_username: rsua1 l_domain: 172.16.254.101 r_username: cust_sua1 r_domain: 172.16.254.102 realm: auth_username: cust_sua1 auth_password: check123 auth_proxy: sip:172.16.254.102:5060 expires: 3600 flags: 20 diff_expires: 3526 timer_expires: 1466349066 reg_init: 1466345451 reg_delay: 0 } """ Also here is the command refreshing the record (after that is not longer in db): """ POST /jsonrpc HTTP/1.1. Host: 172.16.254.101:5090. User-Agent: Go-http-client/1.1. Content-Length: 85. Content-Type: application/json. Accept-Encoding: gzip. . {"jsonrpc":"2.0","method":"uac.reg_refresh","params":["rsua1_172.16.254.101"],"id":7} # T 2016/06/19 16:17:47.275473 172.16.254.101:5090 -> 172.16.254.102:58172 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP 172.16.254.102:58172. Content-Type: application/json. Server: Sipean-RP 4.3.2. Content-Length: 36. . {"jsonrpc":"2.0","result":{},"id":7} """ Ta, DanB On 16.06.2016 12:00, sr-users-request(a)lists.sip-router.org wrote: > Message: 10 Date: Thu, 16 Jun 2016 08:17:53 +0200 From: > Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: > [SR-Users] uac.reg_refresh not removing records from memory > Message-ID: <76d9f2eb-26f7-c0db-8b74-37c93a5d3ef4(a)gmail.com> > Content-Type: text/plain; charset=utf-8 Hello, On 14/06/16 16:52, DanB > wrote: >> > >> >Hey Guys, >> > >> >I was wondering if there is any dynamic way today to remove a single >> >registration out of memory (and implicitly to force un-REGISTER) >> > >> >Eg: I add one record using uac.reg_refresh (which works fine) but after >> >removing the record from the database and executing uac.reg_refresh >> >again, the record stays in memory until expire. >> >Is there a better way to enforce un-register on remote side? >> > > do the flags in memory of the uac_reg account show that it is disabled? > > Cheers, > Daniel

4 3

WS to WS calls - No ACK received for 200OK
by Amit Patkar 21 Jun '16

21 Jun '16

Hi I am using Kamailio as Websocket proxy. User 1 & User 2 are registered on Kamailio over WebSocket. When User 1 calls User 2, User 2 gets ring and answers the call. 200 OK message is received by User 1 but ACK response sent by User 1 does not reach User 2. Since User 2 didn't get ACS, after 30 sec timeout it drops the call. Media is exchanged for 30 seconds, which means ICE is successful. I am running kamailio 4.3.3 on Kamailio is behind firewall and running on private IP. rtpengine is configured to handle media. All ports are forwarded to kamailio server and Kamailio is allowed to send data on any public IP. I used jssip & sipml framework to test. Result was same for both frameworks. No ACK received I can see following errors in log file. What could be the reason which indicate ACK was not forwarded Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6584]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: transaction answered: timestamp=1466501141;method=INVITE;from_tag=7926B9NR7U0X99ZfAr1T;to_tag=w3D2JGd2EUFJpobTcFyo;call_id=1edc8e15-f1f1-584d-df81-71c3d59d713d;code=200;reason=OK;src_user=10001;src_domain=xxxx.com;src_ip=xxx.yyy.zzz.aaa;dst_ouser=10002;dst_user=10002;dst_domain=df7jal23ls0d.invalid Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6584]: INFO: <script>: WS to WS Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:46 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:46 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:51 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:51 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:57 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:57 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:56:13 acstemplate /usr/sbin/kamailio[6581]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:56:13 acstemplate /usr/sbin/kamailio[6581]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) I searched list for similar issues but could not find match any Help to resolve this issue is appreciated. Thanks & Regards, Amit Patkar

3 4

Kamailio consult CNAM server through SIP Subscribe.
by José Seabra 21 Jun '16

21 Jun '16

Hello there, I need to use kamailio to consult an CNAM server in order to get the caller id name and number to the call, the diagram bellow represents what i need implement: Can I implement this behavior through the following kamailio modules: 1. tm and tmx module (call the function t_suspend and t_continue in order to stop INVITE transaction and resume it). 2. UAC module (generate the SUBSCRIBE sip message) 3. Some kamailio scripting logic to manage this behavior. what are your advices to implement this scenario? Thank you for your help Regards -- Cumprimentos José Seabra

4 17

kamailio question
by Irakli Nanobashvili 21 Jun '16

21 Jun '16

dear sr_users i have a little problem about kamailio sip server and who can help me ?

2 1

Equivalent of OpenSIP's binary interface in Kamailio
by Gholamreza Sabery 21 Jun '16

21 Jun '16

In OpenSIPS binary interface ( https://www.opensips.org/Documentation/Interface-Binary-2-2) provides a more efficient way of replicating state between multiple OpenSIPs instances. What is the equivalent feature in Kamailio? Regards

2 2

RTPENGINE rtp errors
by Dmitry 21 Jun '16

21 Jun '16

Hi 1)I encountered such errors when I make an internal call between 2 yate client SIP softphones:rtpengine is used and no audio: | Jun 19 10:02:35 kazootest2 rtpengine[2066]: [918508371(a)kazootest2.siplabs.local port 30942] Error parsing RTP header: short packet (header) when I turn off the rtpengine:THe Freeswitch handles such RTP packets correctly and audio is ok. I compared manually rtp packets from Yate: they comply to the RTP specification. 2) A call from WS client(jssip) to an ordinary sip softphone: | Jun 19 09:58:36 kazootest2 rtpengine[2066]: [s2m2k70tu0d12bjnnamr port 30814] Error parsing RTP header: invalid header version | How to fix these errors? The rtpengine looks through all incoming packets while looking for a new RTP stream. May be packets parsing is too strict? |

1 1

No other running SIP apllication on the machine.//答复: Kamailio ERROR "Address already in use" in tcp_init().
by joey.ren＠yulong.com 20 Jun '16

20 Jun '16

2 1

Kamailio dispatcher module - keep alives
by José Seabra 20 Jun '16

20 Jun '16

Hello there, I have a Kamailio server using dispatcher module to monitor several peers, some of them are in the same network, others are in an external network, for the peers that are in the external network kamailio has to send the OPTIONS(dispatcher keep alives) through another kamailio located in the border and acting as loadbalancer. If i set the dispatcher option "outbound_proxy" all SIP OPTIONS are sent to the outbound proxy configured and it isn't what i need, I need to have by some way a possibility to set outbound_proxy only for the peers that are located on the external network. How can I do that? Regards -- Cumprimentos José Seabra

3 5

Kamailio ERROR "Address already in use" in tcp_init().
by joey.ren＠yulong.com 20 Jun '16

20 Jun '16

3 2

Why SDPOPS does not remove attributes in SDP
by Dmitry 18 Jun '16

18 Jun '16

Hi allI have the following code: if($T_reply_code=="200") { if(has_body("application/sdp")) { xlog("L_INFO", "RTPENGINE received internal reply $T_reply_code $rr SDP extra lines will be removed"); set_rtpengine_set("0"); rtpengine_manage(); sdp_remove_line_by_prefix("a=rtcp"); sdp_remove_line_by_prefix("a=ssrc"); sdp_remove_line_by_prefix("a=ice"); sdp_remove_line_by_prefix("a=candidate"); xlog("L_INFO", "RTPENGINE received internal reply $T_reply_code $rr SDP extra lines removed with SDPOPS"); } }When I look through traces - I see that 200 ok(with SDP) has all these attributes and they are not removed. Why SDPOPS does not remove these attributes?

4 5

Help with routing block and Dispatcher Module
by pablo rosales 17 Jun '16

17 Jun '16

Hi everyone! I am a newbie with Kamailio. I'm trying to connect a kamailio server to an asterisk gateway. We are using basic Kamailio 4.4.x configuration file with mysql and rtpproxy we are having same issues with dispatcher module. we can not call between local subscribers. Basically, If I call to a local subscriber, the module always execute the route(DISPATCHER) code on the routing block, trying to send the call to my gateway, but if I try to call to an external number, it works perfect. All calls are being send to my gateway. Here is my dispatcher code: # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } ### xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); xlog(" --- DISPATCH: going to <$ru> via <$du>n"); t_on_failure("RTF_DISPATCH"); return; } ## Sample failure route failure_route[RTF_DISPATCH] { xlog(" --- DISPATCH: Starts RTF_DISPATCH \n"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } Can anybody help to see what I am doing wrong??

2 2

http_client module configuring User-Agent
by Jurijs Ivolga 17 Jun '16

17 Jun '16

Hi All, I tried today new module http_client. I just made GET request using http_client_query in following way: http_client_query("http://cool.api.com", "$var(result)"); It is working, but I can't see User-Agent in request. Below you can find my module options: modparam("http_client", "useragent", "kamailio") modparam("http_client", "connection_timeout", 1) Is it a bug, or am I doing something wrong? Thank you! Jurijs

2 3

Kamailio Problem carrying media
by Sirvan Paraste 16 Jun '16

16 Jun '16

Hi there We have got a situation that needed your help to overcome that. We have setup a Kamailio with voice, video call and text messaging and even push notification ability. it was working and till these days that we have lost media (voice and video and calling ability). seems that the clients do not see each other but strange is that they can send text messages and even push notification is work as well. Could you please give me an idea how to find the problem, or if you know how to solve it? Best regards. Sirvan Parasteh

2 1

Kamailio on multiple ports
by huseyin kalyoncu 16 Jun '16

16 Jun '16

Hello, We have kamailio running as proxy in front of a freeswitch box. This kamailio also acts as a registrar listening both 5060 and 5090 ports. We have two devices registered to kamailio: Device A on port 5060 Device B on port 5090 When Device B calls Device A everything is fine. Call released successfully. When Device A calls Device B, call setup is ok and they can talk. But when Device A tries to release the call, Device B remains open which cannot get a BYE message. When I dug up for the cause of the problem, I realized that: When Kamailo receives 200OK reply on port 5090 for the UPDATE message from Device B, it relays this reply internally and then sends this reply to freeswitch box on port 5060. This results in loosing alias in the contact field of the reply message and then which causes freeswitch to return BYE message with the wrong IP which is the local IP of the Device B. As a result this BYE message cannot be delivered to Device B. Is this a normal behaviour ? If it is so what changes should I make in the kamailio config to fix this? You can find the ngrep outputs of the both calls as attachment. Thanks Huseyin Kalyoncu

1 1

just a quick glance at my kamailio.cfg (suspect a rtpproxy_manage behavior)
by Sébastien Brice 16 Jun '16

16 Jun '16

Hi guys i am bridging call on a private ip asterisk behind kamailio/rtpproxy i am calling rtpproxy with the route [FROMASTERISK] route[FROMASTERISK] { if(ds_is_from_list()){ xlog("L_INFO","$dd Call from Media-Server Cluster\n"); rtpproxy_manage("cawei"); SIP INVITE with sdp always report a corrupt body!!! Session Name (s): Asterisk PBX 13.9.1 Connection Information (c): IN IP4 PUBLICIPPUBLICIP Connection Address: PUBLICIPPUBLICIP thats really strange since i am calling rtpproxy_manage only once i can't understand it appears twice. what do you think about that ? thx you

2 2

ACK not sent out
by Nagorny, Dimitry 16 Jun '16

16 Jun '16

Hello List, I have a setup of two WebRTC-Clients registered to kamailio. RegA calls RegB. Signalling itself for connection seems to work until the point RegA receives the 200Ok and wants to send the ACK, kamailio throws errors. Here are the 200Ok with the following ACK and the errors that kamailio writes: 200OK (from RegB to RegA, successful): SIP/2.0 200 OK Record-Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg> Via: SIP/2.0/WSS 192.0.2.1;rport=54839;received=10.250.5.17;branch=z9hG4bK415866 To: <sip:dnagorny@r5dev-kam1.robot5.de>;tag=ecmmlvc4ee From: "Testaccount" <sip:4@r5dev-kam1.robot5.de>;tag=3bege11omg Call-ID: vu8rkkgki2uupuf2cqf4 CSeq: 2017 INVITE Contact: <sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;alias=10.250.5.17~54838~6;transport=ws> Supported: ice,replaces,outbound Content-Type: application/sdp Content-Length: 1001 v=0 o=mozilla...THIS_IS_SDPARTA-47.0 4652302965311172452 0 IN IP4 0.0.0.0 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 1A:88:2D:6E:9F:82:14:E0:A2:DB:A7:16:56:00:2C:25:1B:D2:D1:EE:B2:D5:ED:6F:E1:7C:5C:B4:1A:30:9C:85 a=group:BUNDLE audio a=ice-options:trickle a=msid-semantic:WMS * m=audio 30710 UDP/TLS/RTP/SAVPF 111 c=IN IP4 10.250.5.74 a=candidate:0 1 UDP 2122252543 10.250.5.17 52581 typ host a=candidate:1 1 UDP 1686052863 87.158.246.174 49807 typ srflx raddr 10.250.5.17 rport 52581 a=end-of-candidates a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=fmtp:111 maxplaybackrate=48000;stereo=1 a=ice-pwd:4d722747e7d3b3caa360ebe9e646fb44 a=ice-ufrag:2b7e23b8 a=mid:audio a=msid:{5cea7c38-cc3e-4bd2-9a3b-5d2381f0acf9} {0236e1cf-dfdc-4c94-8ce0-59f52a23ebb1} a=rtpmap:111 opus/48000/2 a=setup:active a=ssrc:565594393 cname:{7df4b0da-093c-4c2d-b154-0b860b43f575} a=sendrecv a=rtcp:30710 a=rtcp-mux a=candidate:Y2XhN5oQNoH6nRph 1 UDP 2122252799 10.250.5.74 30710 typ host ACK (from RegA to RegB, unsuccessfull): ACK sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;transport=ws SIP/2.0 Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg> Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK2821443 Max-Forwards: 69 To: <sip:dnagorny@r5dev-kam1.robot5.de>;tag=ecmmlvc4ee From: "Testaccount" <sip:4@r5dev-kam1.robot5.de>;tag=3bege11omg Call-ID: vu8rkkgki2uupuf2cqf4 CSeq: 2017 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: outbound User-Agent: JsSIP 1.0.1 Content-Length: 0 Thrown Errors: kamailio[5009]: INFO: <script>: Is natted kamailio[5009]: INFO: <script>: From: sip:4@r5dev-kam1.robot5.de To: sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;transport=ws @ <null> Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK4386608 Route: <null> vu8rkkgki2uupuf2cqf4 kamailio[5009]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket kamailio[5009]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) I don't understand these errors. Can anybody explain and point me what needs to be fixed? Using Kamailio 4.3.5 on CentOS7.2, funny info if I call from outside(PSTN/FreeSWITCH) to kamailio into WebRTC-Client it works w/o problems. Respectfully Dimitry Nagorny Trainee

1 1

OpenSIPS balcklist/whitelist related question
by eyatsko 16 Jun '16

16 Jun '16

Hello! Could you, please, help me to make the above?.. I already have rough understanding about blst, permission modules etc. But I am about to compose CFG to count wrong (erroneous) attempts to REGISTER or INVITE from the same IP through several SIP dialogs. For example, some malicious user attempes to brute-force password. I would like to move it permanently into permissions' database after several (predefined) attempts. Kind regards, Ellad

1 0

uac.reg_refresh not removing records from memory
by DanB 16 Jun '16

16 Jun '16

Hey Guys, I was wondering if there is any dynamic way today to remove a single registration out of memory (and implicitly to force un-REGISTER) Eg: I add one record using uac.reg_refresh (which works fine) but after removing the record from the database and executing uac.reg_refresh again, the record stays in memory until expire. Is there a better way to enforce un-register on remote side? Thanks in advance! DanB

2 1

Question about testing RTPENGINE
by Dmitry 16 Jun '16

16 Jun '16

Hello I am trying to make a test setup to test the RTPENGINE. Case 1: Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.136)-----via NAT(10.111.101.3)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) Case 2: When A calls to B - the sound both ways, so it is ok. Subsriber A ---Asterisk (10.20.20.101)-----via NAT(10.11.97.12)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) WITH RTPENGINE: When A calls to B - the sound is absent, no sound. I compared both dumps the difference only in addressesAnd what I read in nathelper module: - 1 - Contact header field is searched for occurrence of RFC1918 addresses. - 2 - the "received" test is used: address in Via is compared against source IP address of signaling - 4 - Top Most VIA is searched for occurrence of RFC1918 addresses - 8 - SDP is searched for occurrence of RFC1918 addresses - 16 - test if the source port is different from the port in Via - 32 - test if the source IP address of signaling is a RFC1918 address All flags can be bitwise combined, the test returns true if any of the tests identified a NAT. in nat-traversal-role.cfgI see: if (nat_uac_test("3")) { force_rport(); fix_nated_contact(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } As I understand - both addresses are RFC1918but in case 2 the RTPENGINE is not kernalized Does the RTPENGINE distinguish the ip address? why it was not kernalized in case 2? Any help is appresiated.

1 1

RTPENGINE not kernalizing in some cases
by Dmitry 16 Jun '16

16 Jun '16

Hi I have the Kamailio and FS behind it (in Kazoo). Both are in one ip address 10.34.101.104 i use rtpengine_manage(). When I call from Asterisk which is behind NAT :1) Asterisk in VMware player (NAT, local ip 192.168.175.136) and public 10.34.101.103 - the voice was ok? in logs I saw Kernalizing media 2)Asterisk in KVM(NAT, local 10.28.40.101 public 10.101.29.45) and no Kernalizing in /var/log/messages and no voice. The Asterisk configs are absolutely the same. Kamailio log shows no errors In both cases the INVITE's SDP connection is 10.34.101.104 - the address of the Kamailioin the ' case 1' I see all RTP streams : 2 rtp streams - between asterisk and kamailio and vice versa, 2 -rtp streams between kamailio/rtpengine - Freeswitch and vice versa in the second case I see the following RTP streams: 1 rtp stream - from Kamailio to Asterisk, 1 rtp stream - from Freeswitch to Kamailio How to debug such cases? Any help is appreciated.

1 1

HTTP query at the end of a phone call
by Jay Li 15 Jun '16

15 Jun '16

Dear All, I'd like to send a HTTP call to a HTTP server whenever a call routed through a Kamailio server ends. By looking at the document, seems I should use http_query() function in the utils module. I wonder which route function I should put the http_query() in assuming we use the default kamailio.cfg. Thanks. Regards,Jay

4 3

404", "Not Found location"
by Sébastien Brice 15 Jun '16

15 Jun '16

Hi guys when i try to register i have route[LOCATION] that returns "404 Not Found location" due to this statement: $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 location"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } thats weird i did follow ubuntu realtime trying to get sip account registered and then uac_req_send to asterisk but the first step already failed. :-( i noticed there is no Username in my RURI (whatever UAC i am using) maybe this is the root of evil please find attache my kamailio.cfg i am using kamailio v 4.4 Any idea very welcome

2 1

Kamailio + ToS
by Marcelo Bonin 15 Jun '16

15 Jun '16

Hello, I need to know what are the available parameters to set ToS on Kamailio. I couldn't found much information about that, only this https://www.kamailio.org/wiki/cookbooks/4.0.x/core#tos but there is no reference for the available parameters. Thanks, Marcelo.

2 1

question about RTPengine
by Dmitry 14 Jun '16

14 Jun '16

hello i read the documentation about RTPengine.and the documentation says that: If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for rtpengine_answer() what does it mean? actually - according to 3261 RTP media (both ways) may be turned on after ACK

3 5

IP TOS
by Daniel W. Graham 14 Jun '16

14 Jun '16

Could I get some clarification on tos global setting as referenced by documentation below? Specifically, are DSCP values allowed? Or is this simply for scheduling purposes at the OS level? I can change the value once packets hit the network but if possible would like 0x60 to originate from Kamailio. tos The TOS (Type Of Service) to be used for the sent IP packages (both TCP and UDP). Example of usage: tos=IPTOS_LOWDELAY tos=0x10 tos=IPTOS_RELIABILITY Thanks! -Dan

2 3

Fwd: [Free-RTC] GSoC progress, more mentors?
by Daniel Pocock 14 Jun '16

14 Jun '16

This report was sent to the Free-RTC list https://lists.fsfe.org/mailman/listinfo/free-rtc Most of the projects have an emphasis on SIP If anybody wants to facilitate testing the new Telepathy/Empathy SIP connection manager code against Kamailio or other SIP proxies, registrars and presence servers that would be very welcome. -------- Forwarded Message -------- Subject: [Free-RTC] GSoC progress, more mentors? Date: Mon, 13 Jun 2016 22:53:46 +0200 From: Daniel Pocock <daniel(a)pocock.pro> Reply-To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org> To: Free-RTC(a)lists.fsfe.org Hi all, The GSoC students have officially been working for 3 weeks now. I created a summary of all the students here: https://wiki.debian.org/SummerOfCode2016/RTCProjects They've been publishing weekly reports on the debian-outreach list: https://lists.debian.org/debian-outreach/ It would be really helpful to increase the size of the mentoring team for the students working on RTC projects: - we probably need at least one more Python developer and one more C++ developer to help - other people can also help test the projects, for example, if you are using Wordpress, please try Mesut's new WPCall module: https://github.com/mesutcang/wpcall As a bonus, some mentors are selected every year for a fully paid trip to California for Google's GSoC mentor summit. If anybody has more questions about it or would like to be more involved please contact me privately. Regards, Daniel _______________________________________________ Free-RTC mailing list Free-RTC(a)lists.fsfe.org https://lists.fsfe.org/mailman/listinfo/free-rtc

1 0

Kamilio as Scalable Registrar + Proxy
by Colin Morelli 13 Jun '16

13 Jun '16

Hello, I currently have Freeswitch acting as a B2BUA, handling registrations, routing, etc for a prototype voice application I have built. Now I'm at the point where it's time to actually scale it out, and I'm looking at a few different options for the SIP proxy + registration later. Essentially, what I would like to have (I think), is: 1) Stateless SIP proxy with consistent hashing to route to: 2) Transaction-stateful SIP proxy handling registrations and initial routing Ideally, the only thing that's stored across the cluster (if possible) would be registrations, in something like memcache/redis. Then, of course, each node might have its own internal state of transactions that it's tracking. I'd prefer the only thing to care about dialog state to be Freeswitch. Finally, the decisions about which FS instance to route to would be made by my application as well, based on runtime rules. Is this something people here have done before? Are there other suggestions on how to scale out kamailio to multiple instances. Thanks in advance, Colin

2 1

BLF lights
by Bill 13 Jun '16

13 Jun '16

I have spent a couple days beating my head against the Kamilio BLF feature. But cannot seem to get presence signals from one ua to another. I am testing with a Zoiper softphone and a Yealink hardware phone, both registering to Kamailio and able to make and receive calls. But getting absoultely nowhere with BLF. Have gone through all the suggestions in whitepaper 43, but still no results. Anybody had success with BLF, please steer me in the ritght direction. Thanks, Bill

2 1

Kamailio benchmark app_python + OPTIONS
by Anonim Stefan 13 Jun '16

13 Jun '16

Hi, I am doing some basic benchmarks with Kamailio. I am using a VM with 1 CPU i7 2,50 GHz and Kamailio with children=1. This is what I measure for each request, inside the benchmark block: - app_python processing (some euclidian distance based on a dlg_var input) -> same for all requests - uac_req_send() OPTIONS -> same for all requests I am running SIPp tests for 1000 calls with 1,10,50,100,150,200 calls per second. What I see is that the Global average processing time per request decreases with the increasing rate; I expected it to increase with the increasing rate. Is there an explanation for this, or am I doing something wrong ?! Thanks, Stefan

3 5

BYE message to an individual client
by kai.ohnacker＠cbc.de 13 Jun '16

13 Jun '16

Hello Community, is it possible to send a BYE message to an individual client via mi command line (siremis)? Cheers, Kai

2 1

TM Module t_suspend question
by gmele 10 Jun '16

10 Jun '16

Hello, we are using the TM Module* t_suspend *function in our kamailio to suspend transactions (mainly INVITE requests). We do this as we send some push notification to the callee that will force it to re-REGISTER. Once we get the REGISTER, we resume the transaction with a *t_continue()*. If the callee never re-REGISTER, the FR Timer will terminate the INVITE transaction. This is working fine for us when the callee has a phone that can receive the push notification. Now, we want to be able to contact users having multiple devices: phone and desktop VoIP clients. The problem with current implementation is that if we wait for the re-REGISTER from the mobile phone, we may never send the INVITE to the desktop VoIP client if the mobile phone is shutdown. So, I'd like to now if it is possible to set a timer to the t_suspend command which will automatically resume the suspended transaction (without issuing an ERROR like the FR timer is doing)? I didn't see such possibility in the TM module, but is there another way to implement this? Hope I was clear enough Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TM-Module-t-suspend-question-tp1493… Sent from the Users mailing list archive at Nabble.com.

3 6

Re: [SR-Users] Push notification question
by Vasiliy Ganchev 10 Jun '16

10 Jun '16

Hi! To cope with your specific cases try connection-oriented transport (tcp/tls). e.g. if app is closed - the connection is closed as well and the usrloc record will be removed. After this use standard approach (with late forking - tsilo module) imho, it is not a good idea to suspend the call if there is records in location. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/Push-notification-question-tp149025… Sent from the Users mailing list archive at Nabble.com.

1 0

Handling ReINVITEs
by Grant Bagdasarian 09 Jun '16

09 Jun '16

Hello, How should reINVITEs be handled in the kamailio script and also in combination with rtpengine? I was reading the RR docs and loose_route function pretty much handles this if I'm correct. We're currently receiving reINVITEs for checking if a session is still active, not for session modifications like SDP changes. Does kamailio handle reINVITEs correctly out of the box or should extra logic be implemented? Also, what happens when rtpengine_manage is called for a reINVITE inside WITHNDLG or in onreply_route for 200 OK? Is the module smart enough to ignore port (de-)allocations when the SDP is still the same as the initial INVITE or 200 OK? Regards, Grant

2 2

forward calls from kamailio(RTPengine) to FREESWITCH
by Ankitt Sharma 09 Jun '16

09 Jun '16

hello i am using using kamailio + RTPengine in my setup and i want to forward every single call to freeswitch(for recording purpose), all three are running on the same machine (testing in the local network) my requirement for media flow is: UA1-->RTPproxy-->>freeswitch-->RTPproxy-->UA2 my user agents are using webrtc media profile so my rtp engine flags are : rtpengine_manage("trust-address replace-origin replace-session-connection RTP/SAVPF"); i've attached my configuration file i've configured freeswitch for media_webrtc=true ,still when i try calling the freeswitch send 488 without sending call to other end should media-address flag be used to forward media through, if yes how it should be done ??

2 1

ACK does not match transaction after stripping Route header
by Francisco Valentin Vinagrero 09 Jun '16

09 Jun '16

Hi there, I'm having an issue in a SBC (ACME) -> KAMAILIO -> Asterisk scenario with an ACK that gets ignored in Kamailio because it does not match any transaction. The INVITE coming from the SBC looks like this (only relevant headers and hidden numbers for simplicity - SBC has IP .12 , Kamailio .30 and Asterisk .34) INVITE sip:mynumber@10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip: mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip: a-number @10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:41754112601@10.15.1.12:5060;transport=udp> And its forwarded to Asterisk with the Record-Route header: INVITE sip: mynumber @10.15.1.30:5060 SIP/2.0 *Record-Route: <sip:10.15.1.30;lr=on;ftag=SDkbo9901-42090> *Via: SIP/2.0/UDP 10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip:mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip:a-number@10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:a-number@10.15.1.12:5060;transport=udp> Then, 200 OK from Asterisk: SIP/2.0 200 OK *Via: SIP/2.0/UDP 10.15.1.30;rport=5060;received=10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> Which is sent to the SBC like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> And finally the SBC sends the ACK: ACK sip:10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdt7p9k00dounet8ic600.1 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 ACK Contact: <sip:a-number@10.15.1.12:5060;transport=udp> *Route: sip:10.15.1.30;lr;ftag=SDkbo9901-42090 The problem: this ACK gets not retransmitted to Asterisk At first, I thought it was some sanity check but after disabling that I realized that it was in the WITHINDLG route. For the incoming ACK I get in the logs: Jun 8 11:56:47 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53240]: ALERT: <script>: Inside LOOSE route for ACK proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route=sip:10.15.1.30;lr;ftag=SDkbo9901-42090 src_ip=10.15.1.12 And once the ACK is ready to be sent to Asterisk, the Route header has been removed and no Record-Route has been added so it fails. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: INFO: rr [rr_mod.c:402]: pv_get_route_uri_f(): No route header present. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: ALERT: <script>: ACK does not match transaction!! proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route= src_ip=10.15.1.30 My WITHINDLG route looks like this: # 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()) { if (is_method("BYE")) { xlog("L_ALERT","Inside LOOSE route\n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_ALERT","Inside LOOSE route for ACK proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_ALERT","ACK does not match transaction!! proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Thanks for reading this :) Any idea about how to validate the transaction? t_check_trans is not being validated... Cheers, Francisco.

2 2

rtpproxy question
by Albert Petit 09 Jun '16

09 Jun '16

Hi list I have installed a newer version of rtpproxy (2.0) in our development server , i was happy because it brings lot of performance improvements However i have found an issue when clients use TLS+SRTP In that scenario i do not want rtpproxy to decrypt/reencrypt the traffic as my B2BUA does that or in some scenarios we do plan to add end to end encryption. RTPProxy 1.X was doing that perfectly Instead since i use rtpproxy 2.0 all SRTP and SRTCP traffic is decrypted by rtpproxy and i receive it clear in the B2BUA How can i disable this new feature and that rtpproxy just keeps sending the received SRTP/SRTCP stream without decrypting it?

3 3

IPv6 Only client-rtp proxy configuration
by aman jeff 09 Jun '16

09 Jun '16

Hi team, How to do the Configuration of Kamailio+RTP-Proxy for SER in IPv6, Currently I can establish SIP Session,while rtp start call are not audible.So Some configuration need to change,Am bit confused on it,Please help to resolve the issue,For my proper VoIP-Communication in IPv6 Surveillance. Regards Arnab

3 2

kamailio mysql+ldap+radius authentication
by wanmon 08 Jun '16

08 Jun '16

Dear list, I have Kamailio 4.4.1 running with local mysql authentication (WITH_MYSQL/WITH_AUTH). i have RADIUS and LDAP services running. I would like to use all of them (MYSQL+RADIUS+LDAP) for Authentication. How is the best "kamailio.cfg" approach to use all defined users on RADIUS and LDAP? so, when a user try to login, kamailio will try MYSQL, them LDAP, them RADIUS. If user/password are not define on any of those 3 system, authentication will fail. Do i need modify my current LDAP tree? Im trying to using standart "userPassword" attribute. This is my reference: http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth i would like to "add" route[LDAPAUTH], not to "replace" route [AUTH]. Kind Regards

2 1

A: README files for modules generated automatically
by Daniel-Constantin Mierla 08 Jun '16

08 Jun '16

Hello, following past discussions about different formats resulted when generating readme files for modules on various systems, which typically led to conflicts when backporting, I put in place a cron job running on kamailio.org to fetch from github the docbook files, generate the readme and push them back. The cron job is running every 15min. >From now on it is no longer needed to generate the readme yourself. Just commit the changes to xml files and push them to git repository. The readme will be refreshed automatically in a bit. The commits in this case are done by a generic git account named kamailio-dev. The commit message looks like: modules: readme files regenerated - app_mono ... Only the first module that has the readme updated is listed (in this example being app_mono). The readmes are automatically generated for master and 4.4 branches. If you do consistent changes to the docbook file, you can generate the readme locally to be sure there is no error, then restore the old version before committing - relevant commands: make modules-readme modules=modules/async rm modules/async/README git checkout modules/async/README Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

4.4 compiler warnings on debian jessie
by Juha Heinanen 08 Jun '16

08 Jun '16

I'll come back to mem/tlsf_malloc.c issue. Camille asked if the compiler warnings on mem/tlsf_malloc.c were occurring on x86 32 bit machine. I now tried and, yes, they do not appear on i386 jessie, but not on amd64 jessie. So looks like this is the same issue that Ovidiu has reported. -- Juha

2 3

4.4 fails to start on debian jessie due to tls issue
by Juha Heinanen 08 Jun '16

08 Jun '16

Was this fixed on master? Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: : tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.1t 3 May 2016" (0x1000114f), compiled "OpenSSL 1.0.1k 8 Jan 2015" (0x100010bf).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: CRITICAL: <core> [main.c:2514]: main(): could not initialize tls, exiting... If so, please backport to 4.4 since it is now impossible to start kamailio on debian jessie. -- Juha

2 6

ack with cscall-id header
by Rawad Khalil 07 Jun '16

07 Jun '16

Dear kamailio community, I have a problem that seems really weird. i have the same problem using kamailio 4.3 and 4.4.1. The scenario is the following: I make a call from a sip phone to kamailio. kamailio routes the Invite to an asterisk server. After a while asterisk sends a Bye request to kamailio. Now instead of routing the Bye to the sip phone, I want to create a re-invite request from kamailio and send it to the sip phone. The purpose of the re-invite is to connect the sip phone to another sip phone that already has an active call. This part is working fine but the problem occurs after the sip phone sends the ‘200 OK’ message for the re-invite. Kamailio sends an ACK for the 200 message but this ACK has a ‘CSCall-id’ header instead of the ‘Call-id’ header. To create the re-invite I am using the UAC module with one change to the source code. In the uac.c file found in the tm module folder, I added 2 lines of code to the request function in order to stop the UAC module from creating a from tag(attached). I also attached a trace that shows the ack message with cscall-id. Thank you, Rawad

2 1

Re: [SR-Users] Regarding kamailio-4.4.0 msrp module
by Daniel-Constantin Mierla 07 Jun '16

07 Jun '16

Hello, what is missing? What is highlighted seems to be in the code. Can you provide a patch or make a pull request on github project? That will make it simpler to understand what you want to change. Cheers, Daniel On 07/06/16 11:51, Surendra.P wrote: > > Hello Daniel, > > > > Switch case control is missing while parsing MSRP > pseudo variables. Please find the code snippet below: > > > > int pv_parse_msrp_name(pv_spec_t *sp, str *in) > > { > > if(sp==NULL || in==NULL || in->len<=0) > > return -1; > > > > switch(in->len) > > { > > case 3: > > if(strncmp(in->s, "buf", 3)==0) > > sp->pvp.pvn.u.isname.name.n = 1; > > else goto error; > > break; > > case 4: > > if(strncmp(in->s, "body", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 2; > > else if(strncmp(in->s, "code", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 3; > > else if(strncmp(in->s, "hdrs", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 4; > > else goto error; > > break; > > case 5: > > if(strncmp(in->s, "msgid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 5; > > else if(strncmp(in->s, "conid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 21; > > else goto error; > > case 6: > > if(strncmp(in->s, "method", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 6; > > else if(strncmp(in->s, "buflen", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 7; > > else if(strncmp(in->s, "sessid", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 8; > > else if(strncmp(in->s, "reason", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 9; > > else if(strncmp(in->s, "crthop", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 12; > > else goto error; > > break; > > > > > > Regards > > Surendra > > > > > > _______________________________________________ > 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://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Regarding kamailio-4.4.0 msrp module
by Surendra.P 07 Jun '16

07 Jun '16

Hello Daniel, Switch case control is missing while parsing MSRP pseudo variables. Please find the code snippet below: int pv_parse_msrp_name(pv_spec_t *sp, str *in) { if(sp==NULL || in==NULL || in->len<=0) return -1; switch(in->len) { case 3: if(strncmp(in->s, "buf", 3)==0) sp->pvp.pvn.u.isname.name.n = 1; else goto error; break; case 4: if(strncmp(in->s, "body", 4)==0) sp->pvp.pvn.u.isname.name.n = 2; else if(strncmp(in->s, "code", 4)==0) sp->pvp.pvn.u.isname.name.n = 3; else if(strncmp(in->s, "hdrs", 4)==0) sp->pvp.pvn.u.isname.name.n = 4; else goto error; break; case 5: if(strncmp(in->s, "msgid", 5)==0) sp->pvp.pvn.u.isname.name.n = 5; else if(strncmp(in->s, "conid", 5)==0) sp->pvp.pvn.u.isname.name.n = 21; else goto error; case 6: if(strncmp(in->s, "method", 6)==0) sp->pvp.pvn.u.isname.name.n = 6; else if(strncmp(in->s, "buflen", 6)==0) sp->pvp.pvn.u.isname.name.n = 7; else if(strncmp(in->s, "sessid", 6)==0) sp->pvp.pvn.u.isname.name.n = 8; else if(strncmp(in->s, "reason", 6)==0) sp->pvp.pvn.u.isname.name.n = 9; else if(strncmp(in->s, "crthop", 6)==0) sp->pvp.pvn.u.isname.name.n = 12; else goto error; break; Regards Surendra

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Admin Arnab

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, How to configure rtp proxy for IPv6 Only client,now we are not able to get the audio, Please help Regards Arnab

2 1

Kamailio Sorce code Documents
by aman jeff 07 Jun '16

07 Jun '16

Hi team, I like to know the source code documention of kamailio, I like to customize textops.c,like to get the variables and structure documents gparam_p, etc.. Regards Jeff

2 1

Yealink BLF
by Bill 06 Jun '16

06 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

2 1

RTPENGINE not kernalizing in some cases
by Dmitry 06 Jun '16

06 Jun '16

is it possible that RTPENGINE failes because 2 rtpengine_offer (2 invites during autharization) and the 183 Reply as I see in the log - is seen as an ANSWER Both INVITES are with SDP RTPENGINE_MANAGEThis call is not answered Received command 'offer' from 127.0.0.1:54490 Creating new call] offer time = 0.000134 sec Replying to 'offer' from 127.0.0.1:54490 Received command 'offer' from 127.0.0.1:53879 offer time = 0.000060 secReplying to 'offer' from 127.0.0.1:53879 Received command 'answer' from 127.0.0.1:40099 answer time = 0.000169 sec Replying to 'anser' from 127.0.0.1:40099 ICE negotiated: local interface 10.34.101.104Confirmed peer address as 10.34.101.104:7458Confirmed peer address as 10.34.101.103:62608

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 06 Jun '16

06 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 1

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