- sr-users - kamailio.org

unknown command <acc_rad_request>, missing loadmodule?
by Satish Patel 16 Aug '14

16 Aug '14

What is this? I have acc.so module loaded but still getting this error. I am running opensip 1.11 root@sip:~# opensips -c -f /etc/opensips/opensips.cfg Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 252, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 259, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 266, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 273, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 280, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] ERROR:core:main: bad config file (5 errors)

2 1

Text Operations on a Message
by AliReza Khoshgoftar Monfared 15 Aug '14

15 Aug '14

Hi list, How can I extract a specific part of my message body? I know that "textops" has the function "search_body(re)" but it returns True/False upon finding "re" in the body of the text, how can I return the actual text that regular expression matches? As an example, my message body is rate is 172#012 I want to extract the number "172" out of this. Thanks Alireza

3 11

Handling 407 reply
by Yuriy Gorlichenko 15 Aug '14

15 Aug '14

Hello I use UAC module to register trunks to my providers. I've try to call to my provider. I send INVITE to my porvider and then It recieve me 407 reply "proxy authentication required". Offcourse I can handle this reply manualy (by adding headers throuch header_add) and add md5 summ, but I think it`s not write and harder for Kamailio. I find module or function at some module that helps me send write INVITE with auth parameters as a reply ar 407 response of provider. Thanks for help.

2 2

Rate Limiting with Kamailio
by AliReza Khoshgoftar Monfared 15 Aug '14

15 Aug '14

Hi list, A quick question, What is the best way to limit the rate of the kamailio, Say I want it to process only 200 INVITEs per second I know this can be done using ratelimit module, but what if I want to change this rate dynamically? say, in an overload control scenario, I am getting a recommended rate piggybacked in my replies, and I read it, then how can I change the messge processing rate of kamailio in runtime inside my script? Is there a way to change the rate using ratelimit but inside script as messages come? The RPC function (rl.set_pipe) does it, but can I call it inside my script? Thanks, Alireza

3 3

Kamamilio ignores RTP engine returned string
by Yuriy Gorlichenko 15 Aug '14

15 Aug '14

Hello Rtpengine (rtpproxy-ng module) works fine with kamailio till today. Without any changes at kamailio or rtpengine kamailio ignores changed by rtpengine SDP content. To check this I use sdp_get() and after tying to call I print avp from this function and returned sdp body by rtpengine != this content My log: Jul 16 17:17:27 test rtpengine[5521]: Got valid command from 127.0.0.1:36484: offer - { "sdp": "v=0#015#012o=- 8380958077563183406 2 IN IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-semantic: WMS o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012m=audio 56984 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 192.168.56.1#015#012a=rtcp:56984 IN IP4 192.168.56.1#015#012a=candidate:2999745851 1 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2999745851 2 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2448668656 1 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:2448668656 2 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:1206525147 1 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:1206525147 2 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:3374335715 1 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:3374335715 2 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:2602400937 1 udp 2121998079 192.168.214.1 56988 typ host generation 0#015#012a=candidate:2602400937 2 udp 2121998079 192.168.214.1 56988 typ host generation 0#015#012a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0#015#012a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0#015#012a=candidate:3748678400 1 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:3748678400 2 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:157933611 1 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:157933611 2 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:2275654163 1 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:2275654163 2 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:3583934553 1 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=candidate:3583934553 2 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=ice-ufrag:Oglxt4kDKBr8sfXY#015#012a=ice-pwd:iZAp/WE5TCqIoPA0hIpxztqq#015#012a=ice-options:google-ice#015#012a=fingerprint:sha-256 E3:E7:04:57:80:03:27:12:D6:88:F6:E7:3E:64:4A:55:DA:AD:3F:DA:4E:41:6E:6F:F0:D0:69:A6:AB:AE:79:4F#015#012a=setup:actpass#015#012a=mid:audio#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time#015#012a=sendrec… opus/48000/2#015#012a=fmtp:111 minptime=10#015#012a=rtpmap:103 ISAC/16000#015#012a=rtpmap:104 ISAC/32000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:106 CN/32000#015#012a=rtpmap:105 CN/16000#015#012a=rtpmap:13 CN/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=maxptime:60#015#012a=ssrc:3151729659 cname:wvwdxd7V5hVETGHP#015#012a=ssrc:3151729659 msid:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=ssrc:3151729659 mslabel:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012a=ssrc:3151729659 label:bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012", "ICE": "remove", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "60m6hpu9sclqnjvhql2q", "received-from": [ "IP4", "213.108.168.163" ], "from-tag": "6oo4g9fr4c", "command": "offer" } Jul 16 17:17:27 test rtpengine[5521]: Unknown flag encountered: 'force' Jul 16 17:17:27 test rtpengine[5521]: [60m6hpu9sclqnjvhql2q] Creating new call Jul 16 17:17:27 test rtpengine[5521]: [60m6hpu9sclqnjvhql2q] Returning to SIP proxy: d3:sdp770:v=0#015#012o=- 8380958077563183406 2 IN IP4 123.33.24.45#015#012s=-#015#012t=0 0#015#012a=msid-semantic: WMS o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012m=audio 30008 RTP/AVP 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 123.33.24.45#015#012a=rtpmap:111 opus/48000/2#015#012a=fmtp:111 minptime=10#015#012a=rtpmap:103 ISAC/16000#015#012a=rtpmap:104 ISAC/32000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:106 CN/32000#015#012a=rtpmap:105 CN/16000#015#012a=rtpmap:13 CN/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=maxptime:60#015#012a=ssrc:3151729659 cname:wvwdxd7V5hVETGHP#015#012a=ssrc:3151729659 msid:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=ssrc:3151729659 mslabel:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012a=ssrc:3151729659 label:bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=sendrecv#015#012a=rtcp:30009#015#012a=rtcp-mux#015#0126:result2:oke Jul 16 17:17:27 test /usr/sbin/kamailio[6969]: INFO: <script>: Faltured {INVITE} Jul 16 17:17:27 test /usr/sbin/kamailio[6969]: INFO: <script>: This is not falture {INVITE} Jul 16 17:17:27 test /usr/sbin/kamailio[6969]: INFO: <script>: SDP is {v=0#015#012o=- 8380958077563183406 2 IN IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-semantic: WMS o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012m=audio 56984 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 192.168.56.1#015#012a=rtcp:56984 IN IP4 192.168.56.1#015#012a=candidate:2999745851 1 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2999745851 2 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2448668656 1 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:2448668656 2 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:1206525147 1 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:1206525147 2 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:3374335715 1 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:3374335715 2 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:2602400937 1 udp 2121998079 192.168.214.1 56988 typ host generation 0#015#012a=candidate:2602400937 2 udp 2121998079 192.168.214.1 56988 typ host generation 0#015#012a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0#015#012a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0#015#012a=candidate:3748678400 1 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:3748678400 2 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:157933611 1 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:157933611 2 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:2275654163 1 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:2275654163 2 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:3583934553 1 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=candidate:3583934553 2 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=ice-ufrag:Oglxt4kDKBr8sfXY#015#012a=ice-pwd:iZAp/WE5TCqIoPA0hIpxztqq#015#012a=ice-options:google-ice#015#012a=fingerprint:sha-256 E3:E7:04:57:80:03:27:12:D6:88:F6:E7:3E:64:4A:55:DA:AD:3F:DA:4E:41:6E:6F:F0:D0:69:A6:AB:AE:79:4F#015#012a=setup:actpass#015#012a=mid:audio#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time#015#012a=sendrec… opus/48000/2#015#012a=fmtp:111 minptime=10#015#012a=rtpmap:103 ISAC/16000#015#012a=rtpmap:104 ISAC/32000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:106 CN/32000#015#012a=rtpmap:105 CN/16000#015#012a=rtpmap:13 CN/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=maxptime:60#015#012a=ssrc:3151729659 cname:wvwdxd7V5hVETGHP#015#012a=ssrc:3151729659 msid:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=ssrc:3151729659 mslabel:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012a=ssrc:3151729659 label:bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012}

3 3

can you define an array and loop in kamailio.cfg
by ANTHONY HESNAUX 15 Aug '14

15 Aug '14

hello all, Can you define an array and a loop in kamailio.cfg not work !!!! $var(i) = 0; tab[0]="192.168.0.2"; tab[1]="192.168.0.3 »; while($var(i)<2) { $var(rip) = $tab[$var(i)]; $uac_req(method)="REGISTER »; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $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(); }

2 1

sipcapture and haproxy
by Grant Bagdasarian 15 Aug '14

15 Aug '14

Hello, I just discovered HAproxy, and I was wondering if it is possible to use it in combination with the sipcapture module to store SIP messages on multiple database servers? Has anyone done this before? Regards, Grant

2 3

New Kamailio module Obfuscate
by Muhammad Shahzad 15 Aug '14

15 Aug '14

Hi, As already discussed in detail in following email thread, https://www.mail-archive.com/sr-users@lists.sip-router.org/msg19922.html The new Kamailio module obfuscate is ready for testing and can be downloaded at, http://webrtc.voip-demos.com/obfuscate.tbz2 It contains full code, with documentation and usage scenarios etc. Adding Kamailio dev team, can you guys review the code and add this module to Kamailio trunk if possible? I will be responsible for its maintenance and bug fixes etc. if any. Thank you.

2 23

what language is using in kamailio.cfg
by ANTHONY HESNAUX 15 Aug '14

15 Aug '14

hello every body, I would like to know what language is using in Kamailio.cfg because I don’t arrive to create a loop thank you anthony

3 2

RPMs - yum repositories
by Daniel-Constantin Mierla 15 Aug '14

15 Aug '14

Hello, courtesy lod.com (sponsoring the hardware and bandwidth) and Fred Posner helping with that, we got a server located in USA to use for the project. Main purpose for now is using it as a RPM repository, available as rpm.kamailio.org . The server takes the packages from open suse build service, making a local mirror. It is not yet automated, but should happen in the future (anyone that can help here with some script, is more than welcome to join the team) -- anyhow, we generate the rpms only at stable releases. The mirror should be usable as an yum repository or individual package download via http, more details at: - http://rpm.kamailio.org I am not an user of a distro using RPMs, therefore I am asking interested people in the community to give it a try and report here eventual issues. Also, if you build packages for other distributions, we can mirror them on the server -- get in touch to setup the process. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Send socket and sequential requests
by Alex Balashov 14 Aug '14

14 Aug '14

Hi, My proxy is bound to a.b.c.d:5060 and secondarily to e.f.g.h:5060. Most traffic comes into a.b.c.d:5060 and is relayed out that interface. Some initial INVITEs are routed out of e.f.g.h:5060, forced via $fs. The problem is that the Record-Route header still reflects a.b.c.d:5060 and that sequential messages from both ends end up entering and exiting a.b.c.d:5060. Many endpoints don't like that; for instance, e2e ACKs from the caller that come from a.b.c.d:5060 while the initial INVITE came from e.f.g.h:5060 can be a problem in that they are not matched by some UASs. What is the best way to approach this? Is this something covered by Kamailio's 'mhomed' core param? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 8

INVITE to provider
by Yuriy Gorlichenko 14 Aug '14

14 Aug '14

Hello. I Use UAC module for register to my porviders (I have several porviders at my server and). I successfully ring from provider to my asterisk servers and route calls to users that registers at Kamailio. So now I need route calls from users to my providers, But can not understand how to send INVITE to provider? Doesanyone has working example for it?

1 0

Fwd: Call disconnects within 30 seconds
by Rahul MathuR 14 Aug '14

14 Aug '14

Hello Davy, My constraint is that I cannot use TCP for this solution. I am attaching my kamailio.cfg file, please please help me to resolve this issue as soon as possible (it has already chewed many nights :( ) On Wed, Aug 13, 2014 at 8:50 PM, davy van de moere < davy.van.de.moere(a)gmail.com> wrote: > Hey Rahul, > > thank you for your pcap file. Your astpp is indeed not succeeding in > getting the OK delivered correctly to your pjsua. Also those 400 Bad > Session Description are no good. > > I do see as well fragmented packets passing by, perhaps an easy test would > be to have everything be sent over TCP. Big sip packets tend to suffer over > UDP networks. From your sip packets your setup looks quite standard, > although with a lot of headers (which is fine), so I'ld give it a try with > sending everything over tcp, t_relay_to_tcp should help you out! > > Good luck! > > > --- In reply to --- > I guess the attachment size was greater than 60Kb thereby getting > quarentined. > Re-sending it to you so that this issue could be resolved. > > > 2014-08-12 14:42 GMT+02:00 davy <davy.van.de.moere(a)gmail.com>: > > If your Kamailio setup is close to vanilla, it should do it by default. >> But Kamailio is a very powerfull tool, it can easily be setup to stop >> passing over ACKs :) >> >> To attempt to answer what explicitly sends along the ACKs, that will most >> likely be the t_relay function, together with the logic which went before >> it… >> >> A good old tcpdump will most likely enlighten us. >> >> Op 12-aug.-2014, om 14:39 heeft Rahul MathuR <rahul.ultimate(a)gmail.com> >> het volgende geschreven: >> >> Hello Davy, >> >> Thanks for writing back.. >> >> Tonight I'll take the tcpdump on Kamailio box and share the file. >> >> Please note that Kamailio and Freeswitch are both on public IP & at >> Freeswitch param, enable_timer=false is set. >> >> Is there any explicit way wherein ACKs can be transmitted to FS ? >> >> >> Thanks in advance ! >> >> >> On Tue, Aug 12, 2014 at 1:15 PM, davy van de moere < >> davy.van.de.moere(a)gmail.com> wrote: >> >>> Are you sure you're getting the ACK correctly to FS? >>> >>> FS typically has this behavior when it did not correctly receive a >>> confirmation of an answer, and after 30 seconds disconnects, as for FS the >>> call has failed. >>> >>> Do you have a trace of the packets? >>> >>> grtz, >>> Davy Van De Moere >>> >>> >>> 2014-08-12 13:37 GMT+02:00 Rahul MathuR <rahul.ultimate(a)gmail.com>: >>> >>>> Hello, >>>> >>>> I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and >>>> Sip server FreeSwitch. >>>> Whenever I call directly from UAC to Sip server, the call gets >>>> established for as long as I want, however when I use the proxy in between, >>>> it gets disconnected within 30 seconds. It seems that FS sends a BYE within >>>> 30 seconds. >>>> >>>> I would really appreciate if anybody please guide me where I am going >>>> wrong in this case ? >>>> >>>> -- >>>> Warm Regds. >>>> MathuRahul >>>> >>>> _______________________________________________ >>>> 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 >>> >>> >> >> >> -- >> Warm Regds. >> MathuRahul >> _______________________________________________ >> 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 > > -- Warm Regds. MathuRahul -- Warm Regds. MathuRahul

1 1

Keep Number of Messages per Source
by AliReza Khoshgoftar Monfared 13 Aug '14

13 Aug '14

Hi List, I am trying to keep a counter for number of messages received per source ip in my kamailio script. The basic challenge is to keep this data in a neat structure. Ideally, I want some array with keys named after source IP addresses, sth like number_of_messages["10.10.10.10"] = 23 Of course, this data structure shall be script-persistent (as opposed to transaction persistent) Is there any suggestion to do this? (keep a record of # of messages received from each source IP address and log them)?

3 7

How do I make subst_hf() use variables in search (not replacement)?
by Alex Villacís Lasso 13 Aug '14

13 Aug '14

Consider the following snippet: if (is_present_hf("Contact")) { xlog("L_ALERT", "===== reply to SUBSCRIBE has Contact: $ct\n"); xlog("L_ALERT", "===== want to replace with $td\n"); xlog("L_ALERT", "===== regexp to use is /$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asterisk.bindport)/$td/\n"); if (subst_hf("Contact", "/127.0.0.1:5080/$td/", "a")) { xlog("L_ALERT", "===== reply had Contact modified\n"); } } If I use the hardcoded regexp "/127.0.0.1:5080/$td/", subst_hf() replaces the Contact value correctly. However, if I use "/$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asterisk.bindport)/$td/" (where asterisk.bindip and asterisk.bindport are the supposed values 127.0.0.1 and 5080), subst_hf() fails to replace the Contact value. How do I make use of the existing configuration variables in order to substitute in the Contact header?

2 6

Kamailio + FreeRadius + MySQL Accounting STOP not updating
by Satish Patel 13 Aug '14

13 Aug '14

I have install Kamailio 4.1 with MySQL database so all users stored in MySQL (authentication), Later I have installed FreeRadius for accounting only, Just accounting (Invite/Bye). In short MySQL for authentication and FreeRadius for accounting ( CDRTool). Problems: If i register user using SIP phone and call other SIP extension then freeradius start accounting and insert entry in MySQL/radaccts tables if i end call then it doesn't update END accounting ( Bye ). I have google it everywhere but didn't get single clue so i hope i get something here from you guys -- kamailio.cfg ( default file i just add following radius section) loadmodule "acc.so" loadmodule "acc_radius.so" ... ... # ------- FreeRadius Accounting ------- modparam("acc_radius", "radius_flag", 1) modparam("acc_radius", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc_radius", "service_type", 15) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc_radius", "radius_extra", "User-Name=$Au") -- /etc/freeradius/sql.conf sql { driver = "rlm_sql_mysql" server = "localhost" login = "radius" password = "radius" radius_db = "radius" sqltrace = no sqltracefile = ${logdir}/sqltrace-%Y%m%d.log num_sql_socks = 25 connect_failure_retry_delay = 60 accounting_start_query = "\ CALL insert_radacct_record( \ 'radius', \ '%{Acct-Session-Id}', \ '%{Acct-Unique-Session-Id}', \ '%{Billing-Party}', \ '%{Billing-Party}', \ '%{SIP-Proxy-IP}', \ '%{NAS-Port}', \ FROM_UNIXTIME('%S'), \ '0', \ '0', \ '0', \ '0', \ '%{Called-Station-Id}', \ '%{Calling-Station-Id}', \ '%{Sip-Response-Code}', \ '%{Service-Type}', \ '%{ENUM-TLD}', \ '%{Framed-IP-Address}', \ UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ '%{Sip-Response-Code}', \ '%{Sip-Method}', \ '%{Sip-Translated-Request-URI}', \ '%{Sip-To-Tag}', \ '%{Sip-From-Tag}', \ '%{Sip-RPId}', \ '%{Source-IP}', \ '%{Source-Port}', \ '%{Canonical-URI}', \ '', \ '', \ '%{Sip-Application-Type}', \ '%{User-Agent}', \ '%{From-Header}' \ )" accounting_stop_query = "\ CALL update_radacct_record( \ 'radius', \ FROM_UNIXTIME('%S'), \ UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ '', \ '%{X-RTP-Stat}', \ '%{Acct-Session-Id}', \ '%{Sip-To-Tag}', \ '%{Sip-From-Tag}' \ )" -- Radius logs rad_recv: Accounting-Request packet from host 127.0.0.1 port 36379, id=11, length=152 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Invite Event-Timestamp = "Aug 12 2014 22:45:36 EDT" Sip-From-Tag = "59eb1875" Sip-To-Tag = "d443e36e" Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ." User-Name = "2002(a)192.168.1.5" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Tue Aug 12 22:45:36 2014 : Info: # Executing section preacct from file /etc/freeradius/sites-enabled/default Tue Aug 12 22:45:36 2014 : Info: +- entering group preacct {...} Tue Aug 12 22:45:36 2014 : Info: ++[preprocess] returns ok Tue Aug 12 22:45:36 2014 : Info: [acct_unique] Hashing 'NAS-Port = 5060,Client-IP-Address = 127.0.0.1,NAS-IP-Address = 127.0.0.1,Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ.",User-Name = "2002(a)192.168.1.5 "' Tue Aug 12 22:45:36 2014 : Info: [acct_unique] Acct-Unique-Session-ID = "117c7740de62ba6c". Tue Aug 12 22:45:36 2014 : Info: ++[acct_unique] returns ok Tue Aug 12 22:45:36 2014 : Info: [suffix] Looking up realm "192.168.1.5" for User-Name = "2002(a)192.168.1.5" Tue Aug 12 22:45:36 2014 : Info: [suffix] No such realm "192.168.1.5" Tue Aug 12 22:45:36 2014 : Info: ++[suffix] returns noop Tue Aug 12 22:45:36 2014 : Info: ++[files] returns noop Tue Aug 12 22:45:36 2014 : Info: # Executing section accounting from file /etc/freeradius/radiusd.conf Tue Aug 12 22:45:36 2014 : Info: +- entering group accounting {...} Tue Aug 12 22:45:36 2014 : Info: [detail] expand: /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d -> /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:45:36 2014 : Info: [detail] /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:45:36 2014 : Info: [detail] expand: %t -> Tue Aug 12 22:45:36 2014 Tue Aug 12 22:45:36 2014 : Info: ++[detail] returns ok Tue Aug 12 22:45:36 2014 : Info: [sql] expand: CALL insert_radacct_record( 'radius', '%{Acct-Session-Id}', '%{Acct-Unique-Session-Id}', '%{Billing-Party}', '%{Billing-Party}', '%{SIP-Proxy-IP}', '%{NAS-Port}', FROM_UNIXTIME('%S'), '0', '0', '0', '0', '%{Called-Station-Id}', '%{Calling-Station-Id}', '%{Sip-Response-Code}', '%{Service-Type}', '%{ENUM-TLD}', '%{Framed-IP-Address}', UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', '%{Sip-Response-Code}', '%{Sip-Method}', '%{Sip-Translated-Request-URI}', '%{Sip-To-Tag}', '%{Sip-From-Tag}', '%{Sip-RPId}', '%{Source-IP}' Tue Aug 12 22:45:36 2014 : Debug: rlm_sql (sql): Reserving sql socket id: 24 Tue Aug 12 22:45:36 2014 : Debug: rlm_sql (sql): Released sql socket id: 24 Tue Aug 12 22:45:36 2014 : Info: ++[sql] returns ok Sending Accounting-Response of id 11 to 127.0.0.1 port 36379 Tue Aug 12 22:45:36 2014 : Info: Finished request 0. Tue Aug 12 22:45:36 2014 : Info: Cleaning up request 0 ID 11 with timestamp +13 Tue Aug 12 22:45:36 2014 : Debug: Going to the next request Tue Aug 12 22:45:36 2014 : Info: Ready to process requests. rad_recv: Accounting-Request packet from host 127.0.0.1 port 35683, id=149, length=152 Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = "Aug 12 2014 22:46:20 EDT" Sip-From-Tag = "59eb1875" Sip-To-Tag = "d443e36e" Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ." User-Name = "2002(a)192.168.1.5" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Tue Aug 12 22:46:20 2014 : Info: # Executing section preacct from file /etc/freeradius/sites-enabled/default Tue Aug 12 22:46:20 2014 : Info: +- entering group preacct {...} Tue Aug 12 22:46:20 2014 : Info: ++[preprocess] returns ok Tue Aug 12 22:46:20 2014 : Info: [acct_unique] Hashing 'NAS-Port = 5060,Client-IP-Address = 127.0.0.1,NAS-IP-Address = 127.0.0.1,Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ.",User-Name = "2002(a)192.168.1.5 "' Tue Aug 12 22:46:20 2014 : Info: [acct_unique] Acct-Unique-Session-ID = "117c7740de62ba6c". Tue Aug 12 22:46:20 2014 : Info: ++[acct_unique] returns ok Tue Aug 12 22:46:20 2014 : Info: [suffix] Looking up realm "192.168.1.5" for User-Name = "2002(a)192.168.1.5" Tue Aug 12 22:46:20 2014 : Info: [suffix] No such realm "192.168.1.5" Tue Aug 12 22:46:20 2014 : Info: ++[suffix] returns noop Tue Aug 12 22:46:20 2014 : Info: ++[files] returns noop Tue Aug 12 22:46:20 2014 : Info: # Executing section accounting from file /etc/freeradius/radiusd.conf Tue Aug 12 22:46:20 2014 : Info: +- entering group accounting {...} Tue Aug 12 22:46:20 2014 : Info: [detail] expand: /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d -> /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:46:20 2014 : Info: [detail] /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:46:20 2014 : Info: [detail] expand: %t -> Tue Aug 12 22:46:20 2014 Tue Aug 12 22:46:20 2014 : Info: ++[detail] returns ok Tue Aug 12 22:46:20 2014 : Info: [sql] expand: CALL update_radacct_record( 'radius', FROM_UNIXTIME('%S'), UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', '', '%{X-RTP-Stat}', '%{Acct-Session-Id}', '%{Sip-To-Tag}', '%{Sip-From-Tag}' ) -> CALL update_radacct_record( 'radius', FROM_UNIXTIME('2014-08-12 22:46:20'), UNIX_TIMESTAMP('2014-08-12 22:46:20') - 'Aug 12 2014 22:46:20 EDT', '', '', 'M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ.', 'd443e36e', '59eb1875' ) Tue Aug 12 22:46:20 2014 : Debug: rlm_sql (sql): Reserving sql socket id: 23 Tue Aug 12 22:46:20 2014 : Info: [sql] expand: stop packet with zero session length. [user '%{User-Name}', nas '%{NAS-IP-Address}'] -> stop packet with zero session length. [user '2002(a)192.168.1.5', nas '127.0.0.1'] Tue Aug 12 22:46:20 2014 : Info: [sql] stop packet with zero session length. [user '2002(a)192.168.1.5', nas '127.0.0.1'] Tue Aug 12 22:46:20 2014 : Debug: rlm_sql (sql): Released sql socket id: 23 Tue Aug 12 22:46:20 2014 : Info: ++[sql] returns noop Sending Accounting-Response of id 149 to 127.0.0.1 port 35683 Tue Aug 12 22:46:20 2014 : Info: Finished request 1. Tue Aug 12 22:46:20 2014 : Info: Cleaning up request 1 ID 149 with timestamp +57 Tue Aug 12 22:46:20 2014 : Debug: Going to the next request Tue Aug 12 22:46:20 2014 : Info: Ready to process requests. -- SQL tables ( Time is wired 1969 ) mysql> select * from radacct196912; +-----------+----------------------------------------------+------------------+----------+-------+--------------+-----------+-------------+---------------------+---------------------+-----------------+---------------+-------------------+------------------+-----------------+------------------+-----------------+------------------+--------------------+-------------+---------+-----------------+----------------+---------------+-----------+-----------------+----------+------------+-------------------------+---------------+--------------------+-----------+---------+---------------+----------+------------+--------------+-----------+-----------+---------------+------+-------+------------+-----------+-----------+---------------+------------+-----------+---------+ | RadAcctId | AcctSessionId | AcctUniqueId | UserName | Realm | NASIPAddress | NASPortId | NASPortType | AcctStartTime | AcctStopTime | AcctSessionTime | AcctAuthentic | ConnectInfo_start | ConnectInfo_stop | AcctInputOctets | AcctOutputOctets | CalledStationId | CallingStationId | AcctTerminateCause | ServiceType | ENUMtld | FramedIPAddress | AcctStartDelay | AcctStopDelay | SipMethod | SipResponseCode | SipToTag | SipFromTag | SipTranslatedRequestURI | SipUserAgents | SipApplicationType | SipCodecs | SipRPID | SipRPIDHeader | SourceIP | SourcePort | CanonicalURI | DelayTime | Timestamp | DestinationId | Rate | Price | Normalized | BillingId | MediaInfo | RTPStatistics | FromHeader | UserAgent | Contact | +-----------+----------------------------------------------+------------------+----------+-------+--------------+-----------+-------------+---------------------+---------------------+-----------------+---------------+-------------------+------------------+-----------------+------------------+-----------------+------------------+--------------------+-------------+---------+-----------------+----------------+---------------+-----------+-----------------+----------+------------+-------------------------+---------------+--------------------+-----------+---------+---------------+----------+------------+--------------+-----------+-----------+---------------+------+-------+------------+-----------+-----------+---------------+------------+-----------+---------+ | 1 | Y2ExZjhkMTU4ODRmN2MwMGM1N2Q1Y2E5YTA5ZjAzNjg | e46f9d0439141495 | | | | 5060 | | 1969-12-31 19:00:00 | 1969-12-31 19:00:00 | 0 | NULL | NULL | | 0 | 0 | | | 200 | Sip-Session | | | 1407883391 | 0 | Invite | 200 | 70cc9216 | e2cb1f0c | | | | | | | | | | | 0 | | | NULL | 0 | | NULL | | | | | | 2 | MzYyY2JhM2Q3MzEyN2U1ODJkMDA1NGNiMTMzYmE0OTc | aa9b8bcb6effa2c7 | | | | 5060 | | 1969-12-31 19:00:00 | 1969-12-31 19:00:00 | 0 | NULL | NULL | | 0 | 0 | | | 200 | Sip-Session | | | 0 | 0 | Invite | 200 | c6ea6056 | 407f047e | | | | | | | | | | | 0 | | | NULL | 0 | | NULL | | | | | | 3 | M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ. | 117c7740de62ba6c | | | | 5060 | | 1969-12-31 19:33:34 | 1969-12-31 19:33:34 | 0 | NULL | NULL | | 0 | 0 | | | 200 | Sip-Session | | | 1407897936 | 1407897980 | Invite | 200 | d443e36e | 59eb1875 | | | | | | | | | | | 0 | | | NULL | 0 | | NULL | | | | | +-----------+----------------------------------------------+------------------+----------+-------+--------------+-----------+-------------+---------------------+---------------------+-----------------+---------------+-------------------+------------------+-----------------+------------------+-----------------+------------------+--------------------+-------------+---------+-----------------+----------------+---------------+-----------+-----------------+----------+------------+-------------------------+---------------+--------------------+-----------+---------+---------------+----------+------------+--------------+-----------+-----------+---------------+------+-------+------------+-----------+-----------+---------------+------------+-----------+---------+ 3 rows in set (0.00 sec)

2 3

Kamailio IMS - forking based on the SIP method
by Daniel Ciprus 13 Aug '14

13 Aug '14

Hello, is there a way to make SCSCF do forking based on SIP method ? The problem is that our implementation on app server is not fully done and we want INVITE to be forked the way it is but OPTIONS shouldn't be forked there since forking is happening twice in this case which results to storm of OPTIONS on the network. thanks -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

1 0

freeradius Duplicate attribute name Event-Timestamp
by Satish Patel 13 Aug '14

13 Aug '14

somehow i am getting following error, if i start radius Error: Errors reading dictionary: dict_init: /etc/freeradius/dictionary.opensips[93]: dict_addattr: Duplicate attribute name Event-Timestamp If i remove it from /etc/freeradius/dictionary.opensips file it works! but my it screwed my time in mysql table ATTRIBUTE Event-Timestamp 230 string Just run grep to find out but it is not anywhere define but still i am getting error root@sip:/etc/freeradius# grep "Event-Timestamp" * -r dictionary.opensips:ATTRIBUTE Event-Timestamp 230 string sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: FROM_UNIXTIME(%{Event-Timestamp}), \ sql.conf: FROM_UNIXTIME(%{Event-Timestamp}), \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \

2 3

Dialog module event_route problem
by betul.cerit＠netgsm.com.tr 13 Aug '14

13 Aug '14

Hello; I am trying to build a prepaid accounting system. I use dialog module's start/end event-route. When even_route[dialog:started] triggered, a http_query works for the started call. When event_route[dialog:end] triggered , a http_query works for call is ended. When http_query works in event_route[dialog:started], i get some CRITICAL ERROR like 'Aug 12 12:16:09 /usr/local/sbin/kamailio[10151]: CRITICAL: dialog [dlg_timer.c:205]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f7634d925d8 tl->next=(nil) tl->prev=(nil)' 'Aug 12 12:16:09 /usr/local/sbin/kamailio[10151]: ERROR: dialog [dlg_handlers.c:1263]: dlg_onroute(): failed to update dialog lifetime' Http_query works very well. I can get all call information on Web Servers but default_time out or other timeout isn't working. After i commented out http_query in event_route[dialog:started] , it stopped giving ERROR but i realized that dialog cannot uptade dialog hash map .I can see all finished call with "kamctl dialog show". After i closed http_query in even_route[dialog:end] , all problems are solved. I think a problem occured between http_query and dialog event_routes.It cannot control dialog lifetime.Any idea about this problem or suggest an alternative way will be appreciated.My configuration is as follows. Best Regards Betül. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu ; useropkodu ") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #!endif route[CHCK_DLG]{ xlog("L_INFO","DLG:AVP=> :>:$avp(i:77):$avp(op_kodu):$avp(i:69) "); $dlg_ctx(timeout_bye)=1; $avp(unique_id)=$sruid; /* Time out suresi i:10 konusabilecegi sure saniye */ $avp(i:10)=0; $avp(i:10)=(int)$avp(i:77); #$avp(i:10)=10; $dlg_ctx(timeout_route)=33; $dlg_var(uniqueid)=$avp(unique_id); $dlg_var(userid)=$avp(userid); $dlg_var(opkodu)=$avp(op_kodu); set_dlg_profile("userid","$avp(userid)"); set_dlg_profile("opkodu","$avp(op_kodu)"); set_dlg_profile("useropkodu","$avp(userid):$avp(op_kodu)"); if(get_profile_size("useropkodu","$avp(d_user_opkodu)")){ xlog("L_INFO","Userid_Size:$avp(d_user_opkodu) kk : $avp(userid):$avp(op_kodu) "); xlog("L_INFO","Userid_Limit:$sht(sayac=>$avp(userid):$avp(op_kodu))"); if(!($sht(sayac=>$avp(userid):$avp(op_kodu))>$avp(d_user_opkodu))){ sl_send_reply("403","Kapasite Asildi."); exit; } } dlg_manage(); return; } event_route[dialog:start]{ xlog("L_ALERT","START:CI:$dlg(callid):u_id:$dlg_var(uniqueid):U_id:$dlg_var(userid) "); xlog("L_ALERT","START:lifetime:$avp(i:10):opkodu:$dlg_var(opkodu):$avp(aranan)" ); #!ifdef WITH_UTILS $var(http_res)=http_query("http://bla.com/somestuff.php?userid=$avp(userid)&aranan=$avp(aranan)&callid…","$var(result)"); if($var(http_res)=="200"){ json_get_field("$var(result)","sonuc","$var(snc)"); json_get_field("$var(result)","yorum","$var(yorum)"); json_get_field("$var(result)","yorum","$var(debug)"); if($var(snc)!=200){ xlog("L_ERR","START:CI:$ci:Sonuc:$var(snc):Yorum:$var(yorum):Debug:$var(debug)"); }else{ xlog("L_INFO","START:CI:$ci:Sonuc:$var(snc)"); } }else{ xlog("L_ERR","START:CI:$ci:HTTP_RESULT:$var(http_res)"); } #!endif } event_route[dialog:end]{ xlog("L_ALERT","END ::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::: :"); xlog("L_ALERT","END:CI:$ci:START_TIME:$dlg(start_ts):NOW:$TS"); $var(billsec)=$TS-$dlg(start_ts); xlog("L_ALERT","END:SURE : $var(billsec): u_id: $dlg_var(uniqueid):code:$T_reply_code:$DLG_status "); xlog("L_ALERT","END:U_ID:$dlg_var(userid):O_Kodu:$dlg_var(opkodu):"); if($var(billsec)<0){ $var(billsec)=$DLG_lifetime; xlog("L_ERR","START:CI:$ci:Billsec Negatif geldi : lifetime alinacak:$DLG_lifetime"); } #!ifdef WITH_UTILS $var(http_res)=http_query("http://bla.com/somestuff.php?uniqueid=$dlg_var(uniqueid)&billsec=$var(bills…","$var(result)"); if($var(http_res)=="200"){ json_get_field("$var(result)","sonuc","$var(snc)"); json_get_field("$var(result)","yorum","$var(yorum)"); json_get_field("$var(result)","yorum","$var(debug)"); if($var(snc)!=200){ xlog("L_ERR","START:CI:$ci:Sonuc:$var(snc):Yorum:$var(yorum):Debug:$var(debug)"); } }else{ xlog("L_ERR","START:CI:$ci:HTTP_RESULT:$var(http_res)"); } #!endif xlog("L_ALERT","END ::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::: :"); }

2 2

Call disconnects within 30 seconds
by Rahul MathuR 13 Aug '14

13 Aug '14

Hello, I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and Sip server FreeSwitch. Whenever I call directly from UAC to Sip server, the call gets established for as long as I want, however when I use the proxy in between, it gets disconnected within 30 seconds. It seems that FS sends a BYE within 30 seconds. I would really appreciate if anybody please guide me where I am going wrong in this case ? -- Warm Regds. MathuRahul

3 5

SDP ending with CRLF and a=nortpproxy:yes
by Narsay, Deep 12 Aug '14

12 Aug '14

Hello, Please let me know if there is any work around for this. Details of the issue: kamailio.cfg has this sequence in kamailio.cfg, ...... modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)" ) ........ sdp_remove_line_by_prefix("a=candidate"); set_iattr("ice_priority", "1"); rtpproxy_manage("frwie"); I'm seeing an issue towards the bottom of the SDP message. .... a=ssrc:738345272 mslabel:VPELrlXvdfGmtevlYLEeVm6zq4JHvI4g a=ssrc:738345272 label:VPELrlXvdfGmtevlYLEeVm6zq4JHvI4ga0 a=ice-ufrag:YJuJWr9clIJHhENw a=ice-pwd:3v1D1ShCg8xIlDZ9 a=candidate:1297581420 1 udp 659136 10.1.2.3 17534 typ host generation 0 a=candidate:1297581420 2 udp 659136 10.1.2.3 17535 typ host generation 0 a=ptime:20 \r\n a=nortpproxy:yes\r\n The call gets rejected by UA due to an invalid (empty) line in SDP. Very old but somewhat similar issue: http://lists.sip-router.org/pipermail/users/2008-September/019605.html (Module authors, I'll be happy to code/test or assist in any way possible if needed.) Just to test it, I commented the following code snippet and the call worked. But that can't be the permanent fix. Thank you, - Deep N Code snippet:rtp_proxy.c::force_rtp_proxy starting from line 2850 * if (proxied == 0 && nortpproxy_str.len) { cp = pkg_malloc((nortpproxy_str.len + CRLF_LEN) * sizeof(char)); if (cp == NULL) { LM_ERR("out of pkg memory\n"); return -1; } anchor = anchor_lump(msg, body.s + body.len - msg->buf, 0, 0); if (anchor == NULL) { LM_ERR("anchor_lump failed\n"); pkg_free(cp); return -1; } memcpy(cp, CRLF, CRLF_LEN); memcpy(cp + CRLF_LEN, nortpproxy_str.s, nortpproxy_str.len); if (insert_new_lump_after(anchor, cp, nortpproxy_str.len + CRLF_LEN, 0) == NULL) { LM_ERR("insert_new_lump_after failed\n"); pkg_free(cp); return -1; } }*

2 3

kamctl not creates /tmp/kamailio_fifo file
by Yuriy Gorlichenko 12 Aug '14

12 Aug '14

Hello I try to use kamctl dispatcher reload cmd but in gives me an error kamctl dispatcher reload ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. when i try to user kamcmd dispather.reload it gives me: ERROR: connect_unix_sock: connect(/tmp/kamailio_ctl): No such file or directory At kamailio.cfg I defined modparams for ctl.so and enables ctl.so module too modparam("ctl", "mode", 0600) modparam("ctl", "user", "root") modparam("ctl", "fifo", "fifo:/tmp/kamailio_fifo") modparam("ctl", "fifo", "/tmp/kamailio_fifo2") modparam("ctl", "fifo", "udp:*:2050") # fifo protocol over udp modparam("ctl", "fifo", "tcp:*:2050") # fifo over tcp At the kamctlrc I defined FIFOPATH="/tmp/kamailio_fifo" So I can not understand what is wrong with my FIFO file?

3 5

New developer: Jorj Bauer
by Daniel-Constantin Mierla 11 Aug '14

11 Aug '14

Hello, I want to announce that a new developer got GIT write access to repository: Jorj Bauer from University of Pennsylvania, USA - he is mainly joining the efforts to maintain and develop further the SCA module: * http://kamailio.org/docs/modules/stable/modules/sca.html His git commit id is: jorj My warm welcome and thanks for joining to help the project. Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Routing for case of client's changing networks
by Dmytro Bogovych 11 Aug '14

11 Aug '14

Greetings all. I have following use case: 0) Peer A and B registers on kamailio via TLS. 1) Peer A establishes call to peer B. 2) Peer A changes network and call has to be refreshed. Peer A reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer. The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering. Is there way to solve or avoid this situation? Thank you :)

2 4

http_query results question
by Jonathan Hunter 11 Aug '14

11 Aug '14

Hi Guys, I have used the http_query function in the past, in conjunction with the json module, on version 3.3 and unbuntu. I am now using version 4.1 on Centos 6.5, and I am using the http_query, which I'm aware records in the results the first line of the response. Can you select the result to be recorded depending on the line in the response, for example if a number is returned only in line 4, can this be extracted from a 200 ok response using this function? Or is this then a need to use JSON module, and if so which repository for centos 6.5 contains it! :) Sorry for the question! Thanks Jon

3 8

Dispatcher caches IP
by Yuriy Gorlichenko 11 Aug '14

11 Aug '14

Hello. I try to integrate dispathcer module (with db table dispatcher) to my Kamailio server. I have problem with changing IP at db table, but when I calling - call goes to old IP that alread deleted from db. I tested my table with query that selects destination address from table, and it replyes to me correct address. I use round-robin ("4") algorythm modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.0.2") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") How to disable this cache?

4 5

Accessing Message Buffer in Kamailio
by AliReza Khoshgoftar Monfared 08 Aug '14

08 Aug '14

Hi list, Is there a way to survey the message buffer (iei the queue where arriving sip messages wait to be processed by worker threads) in Kamailio? Is there a way to check the number of messages in this buffer? Additionally, is it possible to survey the number of messages by method (INVITE, BYE, etc.)? I know that the ratelimit module ( http://kamailio.org/docs/modules/stable/modules/ratelimit.html) shall access such buffers, but I do not see any variable accessible in the config offered by that module. Am I missing sth or is there a way to do so? Thanks Alireza

2 2

[Kamailio-4.1.4]Make Install Error
by Djamel Bahamid 08 Aug '14

08 Aug '14

Hello everybody, I am new on this environment ( Kamailio). I try to install(settle) Kamailio 4.1.4, on an O.S Suse Sles11 SP3, all the compilation took place, with the exception of the installation (make install), where I obtain the following error: make[2]: Nothing to be done for `install-if-newer'. touch /usr/local/lib64/kamailio/modules/userblacklist.so install -m 755 userblacklist.so /usr/local/lib64/kamailio/modules make[2]: `libkmi.so.1.0' is up to date. make[2]: `libsrdb1.so.1.0' is up to date. make[2]: `libkcore.so.1.0' is up to date. make[2]: `libsrutils.so.1.0' is up to date. LD (gcc) [M usrloc.so] usrloc.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 18: syntax error near unexpected token `fi' */bin/sh: -c: line 18: ` fi ; \' make: *** [install-cfg] Error 2 * Could you help me please to solve this error, Best Regards, Djamel.

4 14

ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid"
by Manuel Camarg 08 Aug '14

08 Aug '14

I have this config: WebRTC sipml5 -> Kamailio -> Asterisk -> Kamailio -> WebRTC sipml5 When placing a communication between two sipml5 points I get this errors: ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid" ERROR: <core> [forward.c:532]: forward_request(): ERROR: forward_request: bad host name df7jal23ls0d.invalid, dropping packet ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) If I place a comm between WebRTC sipml5 -> Kamailio -> Asterisk -> Softphone (connected to Asterisk) Everything works fine I'm running RTPEngine as RTP Proxy for DTLS/SRTP issues SIP INVITE signalling Ringing 180 message and 200 OK gets done fine Any ideas where this issue might come from? Looks something related with media traffic (RTP). Kind regards!

2 2

Complex NAT Scenario,HELP!
by Agiftel 08 Aug '14

08 Aug '14

Hi all, I need help configuring a little bit complex NAT/PAT scenario. Idea is that clients on Internet registers themselvs against kamailio and then it routes requests to Alcatel PBX. kamailio acts also as NAT proxy ( so that clients on internet does not need vpn connection. Something like EDGE SERVER for MS Lync clients) Below the scenario Alcatel PBX: 10.9.6.3 Kamailio internal NIC: 172.16.52.240 Kamailio DMZ NIC: 10.9.23.180 PU.BL.I.C.IP: 1.2.3.4 ALCATEL PBX(LAN 10.9.6.3)————-------------- | |———-(LAN INT 172.16.52.240) | |KAMAILIO (2 NICs) ———————————— | | | | |————(LAN dmz 10.9.23.180) PAT/NAT | |————firewal | | PU.BL.I.C.IP ^ | | Client on internet As you can see there is a firewall that make NAT and PAT from PU.BL.I.C.IP to internal Kamailio DMZ nic. natted ports are 5060 and range from 30000 to 65000 ( for RTP ) I'am tryng to use kamailio.cfg that comes with installation.. Now I am making some changes but is not working. Internal client ( lan 172.16.52.x ), registered on kamailio, can make call to alcatel phones and viceversa. External client ( on internet ) can register to kamailio but cannot call alcatel phone nor client registered on kamailio. Here some traces CALL FROM EXTERNAL CLIENT TO INTERNAL CLIENT: *U 2014/07/25 11:26:55.709196 2.2.2.2:39464 -> 10.9.23.180:5060 INVITE sip:1001@lwsbc.longwave.eu SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:casa@2.2.2.2:39464>. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4.6.1 73073-dadc6769-W. Content-Length: 275. . v=0. o=- 13050753989744621 1 IN IP4 192.168.1.102. s=X-Lite release 4.6.1 stamp 73073. c=IN IP4 2.2.2.2. t=0 0. m=audio 65220 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:26:55.822346 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 100 LW-SBC is Trying. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Server: LW-SBC. Content-Length: 0. . U 2014/07/25 11:26:55.891270 172.16.52.240:5060 -> 172.16.52.53:1384 INVITE sip:bria@172.16.52.53:1384;rinstance=b31d64b3d4ead28c SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Max-Forwards: 69. Contact: <sip:casa@2.2.2.2:39464>. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4.6.1 73073-dadc6769-W. Content-Length: 510. . v=0. o=- 13050753989744621 1 IN IP4 192.168.1.102. s=X-Lite release 4.6.1 stamp 73073. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30108 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30109. a=ice-ufrag:wa6qnn48. a=ice-pwd:L2qWms8nR1SA9Z8rWtDiMpzjyEHV. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30108 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30109 typ host. U 2014/07/25 11:26:55.973284 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Content-Length: 0. . U 2014/07/25 11:26:55.986747 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. User-Agent: Bria 3 release 3.5.5 stamp 71243. Allow-Events: hold, talk. Content-Length: 0. . U 2014/07/25 11:26:56.040076 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. User-Agent: Bria 3 release 3.5.5 stamp 71243. Allow-Events: hold, talk. Content-Length: 0. . U 2014/07/25 11:26:58.473501 10.9.23.180:5060 -> 2.2.2.2:39464 OPTIONS sip:2.2.2.2:39464 SIP/2.0. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0. From: sip:pinger@lwsbc.eu;tag=uloc-53cfd16f-38c-1-b7eb1a5f-72d84677. To: sip:2.2.2.2:39464. Call-ID: b2033011-c4175f65-f074de1(a)10.9.23.180. CSeq: 1 OPTIONS. Content-Length: 0. . U 2014/07/25 11:26:58.551746 2.2.2.2:39464 -> 10.9.23.180:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0;received=1.1.1.1. Contact: <sip:2.2.2.2:39464>. To: <sip:2.2.2.2:39464>;tag=0867c967. From: <sip:pinger@lwsbc.eu>;tag=uloc-53cfd16f-38c-1-b7eb1a5f-72d84677. Call-ID: b2033011-c4175f65-f074de1(a)10.9.23.180. CSeq: 1 OPTIONS. Accept: application/sdp. Accept-Language: en. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: replaces, eventlist. User-Agent: X-Lite release 4.6.1 stamp 73073 52ef741c-W. Content-Length: 0. . U 2014/07/25 11:27:01.110665 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:01.152416 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:01.611529 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:01.649682 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:02.563132 172.16.52.53:1384 -> 172.16.52.240:5060 . . .............. U 2014/07/25 11:27:02.612548 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:02.629843 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:04.613706 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:04.617573 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:08.614089 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:08.615487 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:12.614697 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:12.616644 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:14.950640 2.2.2.2:39464 -> 10.9.23.180:5060 . . .............. U 2014/07/25 11:27:16.615213 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:16.617182 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:20.615928 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:20.617554 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:24.616354 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:24.617435 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:28.478391 10.9.23.180:5060 -> 2.2.2.2:39464 OPTIONS sip:2.2.2.2:39464 SIP/2.0. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0. From: sip:pinger@lwsbc.eu;tag=uloc-53cfd16f-38c-1-b7eb1a5f-82d84677. To: sip:2.2.2.2:39464. Call-ID: b2033011-d4175f65-d274de1(a)10.9.23.180. CSeq: 1 OPTIONS. Content-Length: 0. . U 2014/07/25 11:27:28.584706 2.2.2.2:39464 -> 10.9.23.180:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0;received=1.1.1.1. Contact: <sip:2.2.2.2:39464>. To: <sip:2.2.2.2:39464>;tag=86694f0d. From: <sip:pinger@lwsbc.eu>;tag=uloc-53cfd16f-38c-1-b7eb1a5f-82d84677. Call-ID: b2033011-d4175f65-d274de1(a)10.9.23.180. CSeq: 1 OPTIONS. Accept: application/sdp. Accept-Language: en. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: replaces, eventlist. User-Agent: X-Lite release 4.6.1 stamp 73073 52ef741c-W. Content-Length: 0. . U 2014/07/25 11:27:28.617317 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:28.618441 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:32.564887 172.16.52.53:1384 -> 172.16.52.240:5060 . . .............. U 2014/07/25 11:27:32.618675 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:32.620366 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. * KAMAILIO CODE: cfg: #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_DEBUG #!define WITH_ACCDB #!define WITH_XMLRPC #!define WITH_MULTIDOMAIN #!define WITH_ALIASDB #!define WITH_ALCATEL ##------!define WITH_PSTN #!define WITH_XHTTP #!define WITH_ANTIFLOOD #------#!define WITH_WEBSOCKETS #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif include_file "modules_lw.cfg" ####### 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 { # sip_trace(); # setflag(22); #!ifdef WITH_WEBSOCKETS if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } #!endif # per request initial checks route(REQINIT); # NAT detection #!ifdef WITH_ALCATEL if(($rU=~"^25[0-9][0-9]$")) { route(ALCATEL); } #!endif route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # dispatch destinations to ALCATEL OXE route(ALCATEL); # user location service route(LOCATION); route(RELAY); } #!ifdef WITH_WEBSOCKETS route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); rtpproxy_manage("froc+SP"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); rtpproxy_manage("froc-sp"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); rtpproxy_manage("co"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($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 $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } #!endif route[RELAY] { #!ifdef WITH_WEBSOCKETS route(SETUP_BY_TRANSPORT); #!endif # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif 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; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && is_uri_host_local()) { # 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 exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } xlog("L_NOTICE","#### message flags are: $mf [$mF]\n"); if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ALCATEL # do not auth traffic from ALCATEL - trusted! if(route(ALCATEL)) return; #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || is_from_local()) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!is_from_local() && !is_uri_host_local()) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT xlog("L_NOTICE","####$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); force_rport(); if (!nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); xlog("L_NOTICE","####$rm from $fu (IP:$si:$sp) in Route[NAT] fix_nat-register\n"); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT xlog("####NATMANAGE M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if((src_ip==ALCATEL_LAN) || (src_ip==LONGWAVE_LAN)) { rtpproxy_manage("ie"); xlog("L_NOTICE","####$rm $cs from $fu (IP:$si:$sp) in route[RTPPROXY] RTPproxy with IE Flags\n"); } else { rtpproxy_manage("ei"); xlog("L_NOTICE","####$rm $cs from $fu (IP:$si:$sp) in route[RTPPROXY] RTPproxy with EI Flags\n"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(!is_from_local()) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_NOTICE","####new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies #!ifdef WITH_WEBSOCKETS onreply_route[REPLY_TO_WS] { xlog("L_INFO", "####Reply from softphone: $rs"); if (t_check_status("183")) { change_reply_status("180", "Ringing"); remove_body(); exit; } if(!(status=~"[12][0-9][0-9]")) return; rtpproxy_manage("froc+SP"); route(NATMANAGE); } onreply_route[REPLY_FROM_WS] { xlog("L_INFO", "####Reply from webrtc client: $rs"); if(status=~"[12][0-9][0-9]") { rtpproxy_manage("froc-sp"); route(NATMANAGE); } } #!endif onreply_route[MANAGE_REPLY] { xlog("L_NOTICE","####incoming reply Code: $T_reply_code from IP:$si:$sp \n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } # ====================================================== # Populate CDRs Table of Siremis # ====================================================== route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); } # -- Charts route[STATS] { # clean very old records $var(tmc) = $var(tmc) + 1; $var(x) = $var(tmc) mod 144; if($var(x) == 0) sql_query("ca", "delete from statistics where time_stamp<$Ts - 864000", "ra"); # insert values for Kamailio internal statistics sql_query("ca", "insert into statistics (time_stamp,shm_used_size," "shm_real_used_size,shm_max_used_size,shm_free_used_size," "ul_users,ul_contacts) values ($Ts,$stat(used_size)," "$stat(real_used_size),$stat(max_used_size),$stat(free_size)," "$stat(location-users),$stat(location-contacts))", "ra"); # init the values for first execution, compute the diff for the rest if($var(tmc)==1) { $var(rcv_req_diff) = $stat(rcv_requests); $var(fwd_req_diff) = $stat(fwd_requests); $var(2xx_trans_diff) = $stat(2xx_transactions); } else { $var(rcv_req_diff) = $stat(rcv_requests) - $sht(stats=>last_rcv_req); $var(fwd_req_diff) = $stat(fwd_requests) - $sht(stats=>last_fwd_req); $var(2xx_trans_diff) = $stat(2xx_transactions) - $sht(stats=>last_2xx_trans); } # update the values for stats stored in cache (htable) $sht(stats=>last_rcv_req) = $stat(rcv_requests); $sht(stats=>last_fwd_req) = $stat(fwd_requests); $sht(stats=>last_2xx_trans) = $stat(2xx_transactions); # insert values for stats computed in config sql_query("ca", "update statistics set tm_active=$stat(inuse_transactions)," "rcv_req_diff=$var(rcv_req_diff),fwd_req_diff=$var(fwd_req_diff)," "2xx_trans_diff=$var(2xx_trans_diff) where time_stamp=$Ts", "ra"); } # ALCATEL GW routing route[ALCATEL] { #!ifdef WITH_ALCATEL # check if ALCATEL GW IP is defined if (strempty($sel(cfg_get.alcatel.gw_ip))) { xlog("SCRIPT: ALCATEL routing enabled but alcatel.gw_ip not defined\n"); return; } # route to ALCATEL dialed numbers starting with '25xx' # - update the condition to match your dialing rules for ALCATEL routing if(!($rU=~"^25[0-9][0-9]$")) return; # only local users allowed to call if(!is_from_local()) { sl_send_reply("403", "Not Allowed!!!!!"); exit; } if (strempty($sel(cfg_get.alcatel.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.alcatel.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.alcatel.gw_ip) + ":" + $sel(cfg_get.alcatel.gw_port); } set_advertised_address("10.9.23.180"); route(RELAY); exit; #!endif return; } modules: ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 mhomed=1 log_facility=LOG_LOCAL1 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT #tcp_connection_lifetime=3605 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 server_header="Server: LW-SBC" server_signature=yes user_agent_header="User-Agent: LW-SBC" #!ifdef WITH_XHTTP tcp_accept_no_cl=yes #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLT_OUT 8 #!define ALCATEL_LAN 10.9.6.0/24 #!define LONGWAVE_LAN 172.16.52.0/24 #!substdef "!MY_IP_ADDR_DMZ!10.9.23.180!g" #!substdef "!MY_IP_ADDR_INT!172.16.52.240!g" #!substdef "!MY_DOMAIN!lwsbc.longwave.eu!g" #!substdef "!MY_WS_PORT!10080!g" #!substdef "!MY_WSS_PORT!10443!g" #!substdef "!ALCATEL_IP!10.9.6.3!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR_DMZ:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR_DMZ:MY_WSS_PORT!g" listen=MY_IP_ADDR_DMZ:5060 advertised_address = "1.1.1.1" listen=MY_IP_ADDR_INT:5060 #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif alias = "MY_DOMAIN:5060" alias = "MY_IP_ADDR_DMZ:5060" alias = "MY_IP_ADDR_INT:5060" alias = "MY_DOMAIN:10080" alias = "MY_IP_ADDR_DMZ:10080" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=10080 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "10.9.6.3" desc "PSTN GW Address" pstn.gw_port = "5060" desc "PSTN GW Port" #!endif #!ifdef WITH_ALCATEL # ALCATEL routing # alcatel.gw_ip = "10.9.6.3" desc "Alcatel OXE GW, Longwave" alcatel.gw_port = "5060" desc "Alcatel OXE GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/i386-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "textopsx.so" #!ifdef WITH_WEBSOCKETS loadmodule "websocket.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "siptrace.so" loadmodule "sdpops.so" # -- for siremis CDRs -------------- loadmodule "rtimer.so" loadmodule "sqlops.so" # -- Charts Panel #loadmodule "htable.so" #------- GIA' CARICATO CON L'ANTIFLOOD---------- # -- MI Commands Panel loadmodule "mi_datagram.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy-ng.so" #----loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_XHTTP loadmodule "xhttp.so" #!endif #!ifdef WITH_XHTTP_RPC loadmodule "xhttp_rpc.so" #!endif #!ifdef WITH_XHTTP_PI loadmodule "xhttp_pi.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ------- Modifico il valore di default ( trying -- your call is important to us ) modparam("tm", "auto_inv_100_reason", "LW-SBC is Trying") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy-ng", "rtpproxy_sock", "udp:127.0.0.1:12222") #----modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@lwsbc.eu") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "a=>size=8;autoexpire=920;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #!ifdef WITH_XHTTP_RPC modparam("xhttp_rpc", "xhttp_rpc_root", "http_rpc") #!endif #!ifdef WITH_XHTTP_PI modparam("xhttp_pi", "xhttp_pi_root", "http_pi") modparam("xhttp_pi", "framework", "/etc/kamailio/pi_framework.xml") #!endif # -- for siremis CDRs -------------- modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@localhost/kamailio") # -- For CHARTS modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://kamailio:kamailiorw@localhost/kamailio") modparam("htable", "htable", "stats=>size=6;") # -- MI Commands Panel modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8033") # ----- siptrace params ----- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_on", 1) #modparam("siptrace", "trace_sl_acks", 1) #modparam("siptrace", "trace_local_ip", "10.9.23.180:5060") /etc/default/ngcp-rtpengine-daemon RUN_MEDIAPROXY=yes LISTEN_TCP=127.0.0.1:25060 LISTEN_UDP=127.0.0.1:22222 LISTEN_NG=127.0.0.1:12222 ADDRESS=10.9.23.180 ADV_ADDRESS=1.1.1.1 # ADDRESS_IPV6=... # ADV_ADDRESS_IPV6=... TIMEOUT=60 SILENT_TIMEOUT=3600 PIDFILE=/var/run/ngcp-rtpengine-daemon.pid FORK=yes # TOS=184 TABLE=0 NO_FALLBACK=yes PORT_MIN=30000 PORT_MAX=50000 # REDIS=127.0.0.1:6379 # REDIS_DB=1 # B2B_URL=http://127.0.0.1:8090/ LOG_LEVEL=7 Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Complex-NAT-Scenario-HELP-tp129226.… Sent from the Users mailing list archive at Nabble.com.

2 5

sdpops and rtpproxy/rtpengine
by Alex Balashov 08 Aug '14

08 Aug '14

Hi, Is it safe to use sdpops and the rtpproxy* modules together, given that the latter modify SDP? In other words, can I prune some codecs first, and then call rtpproxy_offer()? Is it safe? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 4

SDPOPS issue or append_hf
by Igor Potjevlesch 07 Aug '14

07 Aug '14

Hello, I have an issue with the module SDPOPS while using "sdp_keep_codecs_by_name". If the calling party sends only one codec description like: Content-Type: application/sdp Content-Length: 202 v=0 o=UserA 2966746938 1790378070 IN IP4 10.141.0.21 s=Session SDP c=IN IP4 10.141.0.21 t=0 0 m=audio 49152 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 The result of the function "sdp_keep_codecs_by_name("PCMA,PCMU,G729a");" is: Content-Type: application/sdp Content-Length: 170 P-Asserted-Identity: "+0123456789" <sip:+0123456789@sip.tld> v=0 o=UserA 2485672881 3000549892 IN IP4 a.b.c.d s=Session SDP c=IN IP4 a.b.c.d t=0 0 m=audio 40330 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=nortpproxy:yes If I open the capture in Wireshark, the PAI is not in the SDP part, and the end of the capture after "a=rtpmap:8 PCMA/8000" is seen as "Data (18 bytes)". I don't understand why the PAI is inserted within the SDP part. Adding the PAI is done after "sdp_keep_codecs_by_name": if (!is_present_hf("P-Asserted-Identity")) { $var(pai) = $(fU{re.subst,/^0/+33/g}); append_hf("P-Asserted-Identity: \"$var(pai)\" <sip:$var(pai)@$fd>\r\n"); } I guess that this cause my INVITE being dropped by 488 Media Not Acceptable Here. Regards, Igor.

2 6

SIP over WebSocket
by Azamat Ailbayev 07 Aug '14

07 Aug '14

Hello, I have a question regarding to SIP over WebSocket implementation. I have changed kamailio.cfg file accordingly with websocket configurations. However, when I am trying to run Kamailio, now I am getting 3 errors on the same line which is "syn_branch = 0". What could be the issue? And How it is possible to fix it? Thank you in advance. BR, Azamat Ailbayev, Kazakhstan

2 1

planning the release of v4.1.5
by Daniel-Constantin Mierla 07 Aug '14

07 Aug '14

Hello, I am considering to release v4.1.5 sometime next week, most likely on the 6th of August. Checking the 4.1 branch, there are not many fixes, few are on my list for backporting. That's good, indicating a high level of stability. If anyone is aware of issues not reported on tracker or patches that have not been backported, add to the tracker or write a message to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 5

Kamailio v4.1.5 Released
by Daniel-Constantin Mierla 06 Aug '14

06 Aug '14

Hello, Kamailio SIP Server v4.1.5 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.4. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.5. For more details about version 4.1.5 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/08/kamailio-v4-1-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio with open ims
by Senthil K 06 Aug '14

06 Aug '14

Hi All, i have a query here, the user is registered with the open IMS and also with Kamailio ( as a third party registration via SCSCF ) Here my issue is ; when the user makes a call, the INVITE reaches to kamailio from SCSCF( Scscf port is 6060) , when the Kamailio sends the INVITE back to IMS, it sends to the port 5060. problem is , the kamailio registers the SCSCF port as 6060, but when sending the INVITE back it uses the port 5060. why this is So ?? i checked in registered entry as well, it shows that the contact is SCSCF with 6060 port only. please help in this situation, Kind regards SenthilK

2 2

Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
by Olli Heiskanen 06 Aug '14

06 Aug '14

Hello, I've been experimenting with Kamailio with ws and sip clients and could need a hand in getting a call between those two to work. I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS 6.5 and a mediaproxy-ng running. I have clients wsclient(a)testers.com and gsclient(a)testers.com and I try to make call from wsclient to gsclient. The wsclient is a jssip client running on chrome and gsclient is a grandstream desk phone. My config file is the default one enhanced by online examples. I use a html5 <audio> element for the media streams, and configured my jssip phone to accept audio options like this: var options = { 'eventHandlers': eventHandlers, 'mediaConstraints': {'audio': true, 'video': false } }; sipUA.call(callto, options); I used the instructions from here: http://www.slideshare.net/crocodilertc/webrtc-websockets What I get is gsclient ringing, and as I answer there is no audio and call hangs up in a few seconds. I guess this is a SDP problem, something between Kamailio and Mediaproxy-ng but SDP is not my strong point so I'd appreciate advice. Question is where's my misconfiguration/problem? I would like to learn why this problem occurs and how to fix it rather than getting a solution right away, but please bear in mind I don't know much about SDP. In Kamailio log I see: kamailio[27059]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Error rewriting SDP kamailio[27058]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id kamailio[27057]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id Following are the INVITEs and 200 OKs from my SIP trace (1.1.1.1 is the ip of my Kamailio & mediaproxy-ng box and 2.2.2.2 is the public ip behind which both my clients are). The gsclient has port 5066. ****************************************************************************** U 2014/04/01 20:03:41.060009 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 2211. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/SAVPF 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=candidate:2999745851 1 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:2999745851 2 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:3350409123 1 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:3350409123 2 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0. a=candidate:4233069003 2 tcp 150995 T 2014/04/01 20:03:41.119806 2.2.2.2:38986 -> 1.1.1.1:5060 [A] ...... U 2014/04/01 20:03:41.159086 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Warning: 304 GS "Media type not available". Content-Length: 0. . U 2014/04/01 20:03:41.159392 1.1.1.1:5060 -> 2.2.2.2:5066 ACK sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Max-Forwards: 16. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 ACK. Content-Length: 0. . U 2014/04/01 20:03:41.161085 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 3136. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/AVP 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=fingerprint:sha-256 72:54:87:EC:D2:4C:D1:70:C2:FE:69:08:20:5C:92:1D:E0:EA:BD:45:09:E0:90:62:27:B6:34:60:54:E2:99:28. a=setup:actpass. a=mid:audio. a=sendrecv. a=rtpmap:111 opus/48000/2. a=fmtp:111 minptime=10. a=rtpmap:103 ISAC/16000. a=rtpmap:104 ISAC/32000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:106 CN/32000. a=rtpmap:105 CN/16000. a=rtpmap:13 CN/8000. a=rtpmap:126 telephone-event/8000. a=maxptime:60. a=ssrc:3298511848 cnam And here are the 200 OK messages when answering the call: U 2014/04/01 20:03:46.049711 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. T 2014/04/01 20:03:46.051127 1.1.1.1:5060 -> 2.2.2.2:38986 [AP] .~.dSIP/2.0 200 OK. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. ****************************************************************************** cheers, Olli

4 26

no fork mode and more than one listen address found (will use only the first one)
by Yuriy Gorlichenko 06 Aug '14

06 Aug '14

Hello. I Installed kamailio 4.1.4, when I starting my server I see following Warning no fork mode and more than one listen address found (will use only the first one) So I see only first listening port at my netstat At my kamailio.cfg file I write fork=yes? but it not helps me. I found the same issue at this mailing list http://comments.gmane.org/gmane.comp.voip.openser.user/10686 But it very old issue (2007 year) and I think patch at this issue does not help me. How I can start my Server for runnings with many interfaces?

3 5

Understanding DNS resolution done by Kamailio
by Igor Potjevlesch 05 Aug '14

05 Aug '14

Hello, I'm using Kamailio 4.1 and I'm wondering how to avoid external DNS resolution. I have the following config: dns_cache_init=no use_dns_cache=no dns=no rev_dns=no Even with this config, I have many and many DNS query on SRV _sip for the hostnames set in carrierroute module. These requests are not useful because the A resolution is done by /etc/hosts. Major problem with this, is that when I have a DNS issue or IP transit issue, Kamailio waits for resolution timeout and becomes overloaded. As a consequence, Kamailio can't treat others SIP requests like REGISTER because he stuck in DNS resolution. Regards, Igor.

2 4

[rtpengine] continuous 'Successful STUN binding request from' messages
by Paul Belanger 05 Aug '14

05 Aug '14

I was hoping somebody could confirm the following is a _normal_ log file for rtpengine. Specifically I am curious of the 'Successful STUN binding request from' messages are continuously logged. Are they part of the normal operation or does this indicate some issue with media? I only ask because I am getting one way audio at the moment. I get audio from rtpengine however rtpengine does not hear me. http://pastebin.com/hztxMFL3 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 3

rtp relaying and cseq - rtpengine/rtpproxy
by Daniel-Constantin Mierla 05 Aug '14

05 Aug '14

Hello, apparently none of the rtp relays I was looking at (rtpproxy and rtpenging) are taking in consideration the CSeq value. Quite recently I met a case with a b2bua that sends the invite without sdp, then after its ACK with SDP sends quickly a re-invite to get itself out of media stream. However, due to network and parallel processing races, the ACK gets processed after the re-INVITE, updating the rtp relay session with attributes that should be no longer used. The impact is practically no audio. I got it working using the 'brave' htable and tracking cseq numbers, skipping calling the relay for lower cseqs, but I wonder if wouldn't be better to push it to the relay, to avoid also races on the network between kamailio and the relay. I would be also curious to hear if others were facing such issues. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Modify Refer-To header
by Stefan Kohlhauser 05 Aug '14

05 Aug '14

2 1

Re: [SR-Users] missing BYE record on acc table when usingtimeout_avp
by betul.cerit＠netgsm.com.tr 05 Aug '14

05 Aug '14

Hi, In ‘event_route[dialog:end]’I can not get any variable they are all empty but using acc_db_requests() worked for me. Thanks for your help. BEST REGARDS From: Daniel-Constantin Mierla Sent: Monday, August 04, 2014 5:28 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] missing BYE record on acc table when usingtimeout_avp Hello, with latest version you can get the cdrs at once via dialog+acc module. If you want to use the classic acc, then add either event_route[dialog:end] or event_route[tm:local-request]. There you can get actions when the dialog is ended (including the time out), and for the second you get the BYE sent out. Perhaps SQLOPS with sql_query() or acc_db_requests() should be used to record the event in db table. Cheers, Daniel On 04/08/14 15:19, betul.cerit(a)netgsm.com.tr wrote: Hi to all, I want to limit call duration according to the credit of the user.To accomplish this I use timeout_avp .When the limit is reached call ends but no record associated with BYE method is inserted to the acc table and that makes impossible to bill that call.What can I do to solve this problem or is there an alternative way to limit call duration?My configuration and logs are as follows. BEST REGARDS. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif -------------------------------------------------------------------------------- route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # 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()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } 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 route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } -------------------------------------------------------------------------------- Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 kamailio /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012! _______________________________________________ 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://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

2 1

[rtpengine] DTLS error: 1 (message digest is null)
by Paul Belanger 04 Aug '14

04 Aug '14

Greetings, I'm having some trouble getting dtls-srtp working with kamailio 4.1 (mediaproxy-ng) and rtpengine (master). I believe I finally have my branch logic setup properly in kamailio, however when the calls get placed into rtpengine, it appears DTLS is not initialized properly. Attached is my debug log file. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 3

4.1 Compilation issue
by Gilbert T. Gutierrez, Jr. 04 Aug '14

04 Aug '14

CentOS6.5 Kamailio 4.1 I compiled Kamailio as follows so that the configuration file would be in "/etc/kamailio" rather than "/usr/local/etc/kamailio/"... make cfg_prefix=/ all make cfg_prefix=/ install When I try to start kamailio I get the following error... Not starting : invalid configuration file! ERROR: loading config file(/usr/local/etc/kamailio/kamailio.cfg): No such file or directory I can find no reference in the init file to that location (which does not exist). Did I use the wrong options in make? Gilbert

3 3

missing BYE record on acc table when using timeout_avp
by betul.cerit＠netgsm.com.tr 04 Aug '14

04 Aug '14

Hi to all, I want to limit call duration according to the credit of the user.To accomplish this I use timeout_avp .When the limit is reached call ends but no record associated with BYE method is inserted to the acc table and that makes impossible to bill that call.What can I do to solve this problem or is there an alternative way to limit call duration?My configuration and logs are as follows. BEST REGARDS. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif -------------------------------------------------------------------------------- route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # 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()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } 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 route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } -------------------------------------------------------------------------------- Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 kamailio /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012!

2 2

LM_ERR and LM_DBG
by aawaise 04 Aug '14

04 Aug '14

I want to ask some queries regarding logging in kamailio. 1. First of all if I want to add LM_DBG or LM_ERR in lookup.c file of registrar module, where to add the command as on extraction of kamailio-3.1.1.zip, I found registrar module files in modules_k folder as well as modules_s folder. 2. I was advised to us macros from dprint.h to see LM_DBG and LM_ERR statements in log file, how to do so ?? 3. Third, will these logs also print in /var/log/messages in debian linux along with other kamailio.cfg logs ?? 4. What changes should be made in kamailio.cfg to see these logs, i.e log level etc ?? Any help will be highly appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/LM-ERR-and-LM-DBG-tp129392.html Sent from the Users mailing list archive at Nabble.com.

2 1

See XDBG and XERR Messages in Logs
by aawaise 04 Aug '14

04 Aug '14

I have added some XLOG and XDBG commands in registrar module of kamailio. I want to see messages in result of these commands being printed in logs to make sure that changes are being made surely. How can it be done ? I am using fedora and I see kamailio related logs in /var/log/messages.log -- View this message in context: http://sip-router.1086192.n5.nabble.com/See-XDBG-and-XERR-Messages-in-Logs-… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration - Problem
by Reza Kiani Moghaddam 03 Aug '14

03 Aug '14

Hi, I Installed Kamailio & Asterisk on Ubuntu 14.04 LTS, and both of them started successfully. I can register with Jitsi, But I can't call other users.. I don't know what to do ...

2 1

Kamailio behind NAT- alias issue
by Yuriy Gorlichenko 03 Aug '14

03 Aug '14

Hello. I have Kamailio running behind NAT. It lesten eth0 with ip 192.168.0.3 and I have external IP that have domain name (for example sip.myserver.com). Register packets from clients comes from external IP. If I write at kamailio.cfg: alias=sip.myserver.com I see error at log - bad_uri sip.myserver.com I think it happens because kamailio does not know atything about external ip pecause kamamiliol working with server interfaces (eth0, eth1, etc.) So my question - how to listen external IP with domain name on kamailio that running behind NAT? P.S. I do not have any access to router, that present NAT for me.

2 2

SIPp load testing
by bob sacamano 02 Aug '14

02 Aug '14

I am trying to follow the guide shown here:http://www.kamailio.org/docs/openser-performance-tests/ to load test my kamailio system.I am a little unclear as to how many instances of sipp are running. For the first part i see the command ./sipp -sf uac_msg.xml -rsa 192.168.2.102:5060 192.168.2.102:5070 -m 200000 -r 10000 -d 1 -l 70for generating the UAC part but is there another instance of sipp running on the kamailio computer acting as a UAS? I have seen come sites use for example: sipp 192.168.1.100:5060 -sn uas -p 5060 but is this not necessary? thanks for any help

2 3

Sample Kamailio Config for Siremis
by Noah Mehl 02 Aug '14

02 Aug '14

I have managed to install Kamailio and Siremis 4.x, but I have noticed that neither Kamaliio or Siremis come with a sample config that makes sense for Siremis (as far as I can tell). For instance, if I want to use LCR with dispatchers, the sample Kamilio configs don’t have the dispatcher module loaded or configured. Am I missing something? Can someone please share theirs? Thanks! ~Noah

3 2

ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene>
by Rene Montilva 01 Aug '14

01 Aug '14

Hi list i've been working with t_suspend, t_continue and async sleep, now the problem it's with register users, the softphones are frozen when try to establish a session with kamailio, now my users can't register and the only error i got is this: ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene> my kamailio config: # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 30) # max value for expires of registrations #modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "hash_size", 15) #!endif

1 0

kamailio routes packets with invalid From/To headers with uac.restore_mode=auto when incoming packet does not use exact same replaced From/To header
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am currently handling a system that runs kamailio and asterisk in the same machine. The kamailio instances are being used to emulate multiple SIP domains, by means of From/To mangling of incoming packets, which are then routed to Asterisk. The attached kamailio.cfg does this work. There is an problem when handling SUBSCRIBE requests (as required for BLF and voicemail indications). My configuration is written so that these SUBSCRIBE requests are not handled by kamailio, but instead routed to asterisk. There is a failure to check From/To headers to see whether NOTIFY packets generated as part of a subscription can be restored using the information in Record-Route. The end result is that kamailio ends up sending packets with garbled tags that are (rightly) rejected by the SIP endpoint. The following is an example that demonstrates the issue (using Jitsi as endpoint): After registration, Jitsi sends a SUBSCRIBE request: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com> Max-Forwards: 70 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Proxy-Authorization: Digest username="avillacisIM",realm="pbx.villacis.com",nonce="U9lZJlPZV/r06Xep/ukc1UzAIO0V3TbS",uri="sip:avillacisIM@pbx.villacis.com",response="0e18f4913c2693f6154c91f158fb17fe" Content-Length: 0 This packet is mangled by the configuration, and is sent to asterisk like this: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> Max-Forwards: 69 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Content-Length: 0 The asterisk response for the SUBSCRIBE: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080>;expires=3600 Content-Length: 0 This is in turn transformed back by kamailio, and sent to Jitsi like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080;alias=127.0.0.1~5080~1>;expires=3600 Content-Length: 0 Now asterisk wants to send a NOTIFY to the endpoint for the subscription. The NOTIFY looks like this: NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport Max-Forwards: 70 Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes>,<sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Here is where the bug appears. The autoprocessing does not recognize that the From header (From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e) from the above request has nothing to do with the saved information (vsf parameter). Instead, it blindly mangles the From header, and does not even run a sanity check on the result before routing it. The end result is shown below. NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as5562e95e> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as5562e95e> Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bK8333.8bfe7bc2bd554a8631f0d00d463b28ee.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport=5080 Max-Forwards: 69 From: "asterisk" <sip:asterisk@12(.0.0.1:5080.....@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) From examination of the source code, the vsf and vst strings are base64 encodings of the result of XORing the byte strings of the old and new tags. For this to work, the headers of future packets should match. However, here kamailio does not realize that the header does not match (by the ftag), and also does not check that the resulting "restored" header is a valid header.

2 2

SCA module - sca_call_info_update() function for 180 responses
by João Vitor Arruda 01 Aug '14

01 Aug '14

Hello all, I've had proposed a fix to SCA module in the tracker ( https://sip-router.org/tracker/index.php?do=details&task_id=446) but did´t hear anything about it until now. I would appreciate some comments on that to see if its feasible to be committed or even hear for anyone else using SCA module with Polycom phones to see if they have the same issue described. Thanks, João Vitor Arruda

4 5

Kamailio V4.1.3 - Dialog timeout -ACC ERROR
by Yasin Caner 01 Aug '14

01 Aug '14

Hello; i am trying to limit call duration with dialog module. So i use *timeout_avp *for limiting it. there is no problem to close calls but when it happens i got some errors and kamailio can not write Bye result in DB. There is code sample and result in below. 2. question : is there a big difference between dialog_ng and dialog? Which one is more stable? 3. question : After restarting kamailio , Dialog hash vanish so timeout isn't working. Can DB mode solve this problem? PS: event_route[dialog:end] isn't working in Kamailio V 4.1.4. I got some error like t_w_relay. Thanks for helps. ------------------------------------------------------------------------ #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif ------------------------------------------------------------------------ route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # 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()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } 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 route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } ------------------------------------------------------------------------ Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 host-91-93-189-136 /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012!

1 0

sip
by Nar 01 Aug '14

01 Aug '14

Hi, I am developing a social application where I want to enable voice & video calling function and still I don't know what to use in server side, do you have any product that I can use to develop the server side(for 10.000+ users)? Thank you in advance.

2 1

[Kamailio]list members
by Djamel Bahamid 01 Aug '14

01 Aug '14

Hi, I would like to subscribe to the list. Best Regards, Djamel BAHAMID.

2 1

SIGUSR1 for memory status not working as documented - only one process reports back
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once memlog is set in the configuration file, a kamailio process will dump a report of the allocation map when shutting down, or when receiving a SIGUSR1. I have configured my kamailio.cfg with memlog=1 and no other change, and I see the memory report on shutdown for all processes. However, when I send a SIGUSR1 to a kamailio process, the process does absolutely nothing, with the exception of the first kamailio process (the one reported as Type=attendant by "kamctl ps"). All of the other processes just ignore SIGUSR1. What is going on? It is inconvenient to force a shutdown of all the kamailio processes just to get the memory report.

2 1

Unable to SUBSCRIBE for presence using sip.js through WSS
by Alex Villacís Lasso 31 Jul '14

31 Jul '14

My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using SIP.js via a websocket, Kamailio is unable to send the NOTIFY for the presence event. However, MESSAGE packets are routed correctly between peers, even through websockets. The location table after registration looks like this: +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | id | ruid | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods | instance | reg_id | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | 192 | uloc-53d96398-1afc-1 | admin | pbx.villacis.com | sip:nfinhooe@192.0.2.90;transport=ws | sip:192.168.3.2:41478;transport=WS | NULL | 2014-07-30 17:18:57 | -1.00 | 6a2dot9ihoobhng9ackqmq | 90 | 2014-07-30 17:08:57 | 0 | 64 | SIP.js/0.5.0 | tls:192.168.2.18:5061 | 782 | <urn:uuid:b8888e95-7833-46a4-a21e-8d2e809a66b8> | 1 | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ When running the following javascript code: var subs = ua.subscribe('admin(a)pbx.villacis.com', 'presence'); subs.on('notify', function (n) { alert(n.request.body); }); ... I see the following traffic in the browser console area: SUBSCRIBE sip:admin@pbx.villacis.com SIP/2.0 Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279 Max-Forwards: 70 To: <sip:admin@pbx.villacis.com> From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Proxy-Authorization: Digest algorithm=MD5, username="admin", realm="pbx.villacis.com", nonce="U9lswVPZa5WSu5jz1P/jIOJTvthKXw6v", uri="sip:admin@pbx.villacis.com", response="f7246e64573b5888d69896a70d4204aa" Event: presence Expires: 3600 Contact: <sip:nfinhooe@192.0.2.90;transport=ws> Allow: ACK,CANCEL,BYE,OPTIONS,MESSAGE Supported: outbound User-Agent: SIP.js/0.5.0 Content-Length: 0 SIP/2.0 202 OK Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279;rport=41478;received=192.168.3.2 To: <sip:admin@pbx.villacis.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-84c3 From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.2.18:5061;transport=ws> Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 However, I also get the following messages in the kamailio log: Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription I have found a discussion at http://marc.info/?l=sr-users&m=136624316816288&w=2 which recommends fix_nated_contact(), but I am not really sure about how to add it, or what conditions to check on whether it should be used. I added the call to fix_nated_contact() like this: # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); # # 1 - Contact header field is searched for occurrence of RFC1918 or rfc6598 addresses. # 2 - the "received" test is used: address in Via is compared against source IP address of signaling # 16 - test if the source port is different from the port in Via # 64 - test if the source connection of signaling is a WebSocket # ----- # 83 #!ifdef WITH_WEBSOCKET # 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. if (nat_uac_test("83")) { #!else if (nat_uac_test("19")) { #!endif if (is_method("REGISTER")) { fix_nated_register(); #!ifdef WITH_WEBSOCKET if (nat_uac_test("64")) fix_nated_contact(); #!endif } else { if (is_first_hop()) if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); } #!endif return; } After registration, "kamctl ul show" reports this: [root@elx3 js]# kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: admin(a)pbx.villacis.com Contact:: sip:85017jar@192.168.3.2:41777;transport=ws Q= Expires:: 584 Callid:: argl25kg0r4qe8lbtbhn9f Cseq:: 82 User-agent:: SIP.js/0.5.0 Received:: sip:192.168.3.2:41777;transport=WS State:: CS_NEW Flags:: 0 Cflag:: 64 Socket:: tls:192.168.2.18:5061 Methods:: 782 Ruid:: uloc-53d971c7-297f-1 Instance:: <urn:uuid:75464840-96b8-4e22-8eaa-5012ba327086> Reg-Id:: 1 Last-Keepalive:: 1406759390 Last-Modified:: 1406759390 However, it still does not work: Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription

1 2

Obscuring SIP traffic and using with NoSIP
by Muhammad Shahzad 31 Jul '14

31 Jul '14

Hi, As the mobile voip is getting more and more popular these days, there has been a strong opposition from GSM operators against mobile voip apps. They often use tactics like blocking voip ports, or detect and block voip traffic and in some cases restricting udp traffic altogether to very low upload and download speeds. See below link for some details, http://www.linphone.org/eng/blog/linphone-over-3g.html While not all the problems can be solved right now (especially the limiting udp traffic, since RTP always uses udp transport) I was wondering if we can at least handle the sip related problems. The most important of them is SIP traffic detection. While some forks would suggest using TCP/TLS to encrypt SIP traffic, it has a few problems, e.g. 1. It requires somewhat high resources on mobile devices, so many low-end android phones simply can't use it. 2. There is possibility that encryption signature may identify it as SIP traffic. There exists firewalls (often deployed in middle eastern countries) which have huge database of encryption signatures and patterns which although may not decrypt the sip packet but at least identify it as sip packet and block it. Also with rough agencies of evil empires spying over millions of users worldwide makes the current encryption standards pretty much pointless, at least in terms of user privacy and network security. So there is a strong need to experiment with new ideas and concepts to regain internet freedom. Some of such ideas are, 1. Convert sip traffic which is plain text to binary format just before transmitting it and revert it to plain text upon reception. 2. XOR the sip traffic (pretty much same as binary sip). 3. Use some very lightweight but effective / non-standard encryption algorithm, e.g. https://github.com/mshary/itv All these ideas require that SIP server such as Kamailio is able to adopt to these, preferably with minimal or no change in native code. The NoSIP module seems an interesting module in this regard. It provides all traffic it thinks is not the SIP traffic to configuration script, where we can do our own parsing and do whatever we want with it. I have two questions about this, 1. If parsed message is SIP, we can we send it back to kamailio core to get it processed as if it is a normal SIP message received by kamailio? 2. Can this module or any other module available in kamailio, that can provide us full sip packet that is about to be transmitted over sip socket, so we can "encode" it just before it is sent to next hop? I know this would be like writing a SIP transport in kamailio script which would be very tough if not impossible to implement in native core. But it will really help in winning the modern mobile voip challenges. Thank you.

3 8

[Serusers] how to enable logs in SER
by pawan 31 Jul '14

31 Jul '14

Sir pl help for registration of soft phone to my local sip server my email vijay8538(a)gmail.com --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

1 0

multi-homed rtpengine
by Paul Belanger 30 Jul '14

30 Jul '14

Greetings, We have had good success in the past using rtpproxy with internal and external interfaces however, we are attempting to migrate to rtpengine now but are running into an issue (apparently this feature is not supported). So, my question is as follows, does this mean we need to multiple rtpengine and bind them to each interface? Additionally, in this setup can audio be bridged between the 2 rtpengine processes? If not, how do people over come this difference? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

1 0

Re: [SR-Users] LCR
by Keith 30 Jul '14

30 Jul '14

Hi, I want to do LCR based on from uri. I get I need to do regular expression matching but have no idea where to look to start. Any pointers would be good. Thanks Keith

3 2

Warning message on SIP websocket
by Alex Villacís Lasso 29 Jul '14

29 Jul '14

Just a quick question. What is supposed to be wrong with the below message? Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message (offset: 22) Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Via: SIP/2.0/TLS 192.168.3.2:58657#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: Er/CgVyBNzVGREBmt+w3FTff8s4=#015#012Server: kamailio (4.1.4 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012>

2 1

Plz help: MSILO on TLS + cert based client auth enabled server
by Marc M. 29 Jul '14

29 Jul '14

Hi, I have a SIP server with TLS + client certificate based authentication. The server is listening for SIP(UDP) on port 5060, and for SIPS(TLS) on port 5061) The client certificate based authentication is enforced on port 5061. I would like to fire up MSILO and face now a strange problem: At each REGISTER, the stored messages get dumped. Dumped messages get sent to the clients from senders logged in over port 5060(UDP) as SIP:xxxxxxxxx@xx.xx.xx.xx however Dumped messages are NOT sent to clients from senders logged in on port 5061(TLS) as SIPS:xxxxxxxx@xx.xx.xx.xx It looks like the messages get dumped so M-dump() is working, but they will never enter the route. I would assume TLS is stopping the dumped messages to entering the route logic. I would either 1. somehow force m_dump() to pipe the messages over SIP(and not SIPS), or 2. disable client auth for messages recieved from local ip Can you help me how to proceed? What would be the correct approach? Can you help me with either 1 or 2. You help is greatly appreciated!!! Marc

2 1

Only a redirect server
by Jack Smith 29 Jul '14

29 Jul '14

Dear friends, Just new to this mailing list, new to Kamailio, and new to running a sip server. I would like to try only a redirect server. I had a look at http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg which was friendly, but then the default file kamailio.cfg looked frightening ! Could someone help me set up a minimal Kamailio installation only to handle state-less redirect? Best, Jack

4 4

append_branch and t_replicate issue
by Asgaroth 28 Jul '14

28 Jul '14

Hi All, I have an issue that I cant seem to get to the bottom of, I would appreciate if someone could point me in the right direction. I have 3 registrar's and would like to replicate the registration regest from 1 to the other 2, I am trying this with t_replicate, the docs say that to send to multiple destinations an append_branch should be performed prior to the t_replicate, which is what I am trying to do. What appears to happen is that the replication only sends the REGISTER to the server defined in the t_replicate command, it doesnt appear to be sending it to the destination defined in append_branch. Kamailio Version: version: kamailio 4.1.4 (x86_64/linux) 84c1ff flags: STATS: Off, USE_TCP, USE_TLS, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 84c1ff compiled on 14:36:04 Jul 20 2014 with gcc 4.4.7 Here is what I have tried so far (on registrar 1): #!define BACKUP_REGISTRAR_1 '1.1.1.1' #!define BACKUP_REGISTRAR_2 '1.1.1.2' #!define BACKUP_REGISTRAR_3 '1.1.1.3' save("location"); if ( src_ip != BACKUP_REGISTRAR_2 && src_ip != BACKUP_REGISTRAR_3 ) { append_branch("sip:" + BACKUP_REGISTRAR_2 + ":5060"); t_replicate("sip:" + BACKUP_REGISTRAR_3 + ":5060"); }; With the above in place, I only see the system attempting to replicate to BACKUP_REGISTRAR_3, I dont see any request to BACKUP_REGISTRAR_2. Am I doing something wrong here? Thanks Bruce

4 29

possible regression on 4.1.3 vs 4.1.4
by Kelvin Chua 26 Jul '14

26 Jul '14

using the default cfg file, whenever a WITHINDLG BYE is received, 4.1.4 needs a retransmission before passing it through, the first BYE would generate this in syslog tm [tm.c:1518]: _w_t_relay_to(): ERROR: w_t_relay_to: unsupported route type: 64 and the 2nd retransmitted BYE would go on fine. i believe route type 64 is branch failure, but using the same cfg and reverting back to 4.1.3 solves it. so whatever changes was done to 4.1.4 broke it. Kelvin Chua

2 4

Rtpengine vs. TURN?
by Peter Villeneuve 25 Jul '14

25 Jul '14

Hi, On my server, I have the option of using either Rtpengine for NAT traversal or pure TURN without rtpengine. Rtpengine has the obvious plus that it only needs 1 public IP, while TURN (with STUN) will need 2 public IPs, although that's not a problem in my case. Having said that, I'd like to take advantage of the huge experience that users of this list have in real world deployments. in your experience, which option is more reliable in a real world deployment? Cheers, Peter

2 3

MSILO not checking Expires header if Allow: is absent and Contact: is present but without expires
by Anthony Messina 25 Jul '14

25 Jul '14

In setting up Kamailio with MSILO, I've noticed that some sip clients (I'm using CSipSimple), upon un-REGISTER-ing do not send an Allow header, but do send an "Expires: 0" header as well as a Contact header. However as the Contact header doesn't contain "expires=0", it appears as though the MSILO skips over checking the Expires header. As such, I've had to add the following workaround to my route[REGISTRAR] block or else these clients trigger the m_dump. # Ensure we aren't unregistering # Right now, CSipSimple doesn't send an Allow header with an un-REGISTER, # so Kamailio skips to checking the Contact header for expires, and appears # to ignore the fact that an Expires: 0 header is present. if($hdr(Expires) != 0) m_dump(); Looking through the msilo check_message_support function, it appears that it never checks for the Expires header if there is no Allow header and the Contact header does not specify an expire setting. I do believe that we're missing the check for the Expires header, regardless of whether or not there is an Allow header, or a Contact header with expires=0 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 4

407 Proxy Authentication Required with Cisco
by proLogika 25 Jul '14

25 Jul '14

Hello, I am using Kamailio as SIP register with asterisk integration describe from hire: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . asterisk is listen on public ip: 2.3.4.5:5060 kamailio is listen on public ip: 1.2.3.4:5060 everything is working well except some software VoIP clients (like Yate) and CISCO phone like Cisco-CP7940G/8.0 and the new one from Cisco series. I’m testing now with Yate client and Cisco. They are register OK but when a call is made Kamailio is answer back with 407 Proxy Authentication Required. When I register Yate or Cisco to asterisk directly the call is passing normaly. I was trying to manipulate kamailio.cfg and more specifically the part: #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; If i commented out this part the call is passing, but I do not have auth anymore (everyone can register) Here is ngrep: U 2014/07/23 19:17:08.108458 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.108805 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. Proxy-Authenticate: Digest realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I". Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.128626 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. User-Agent: YATE/5.3.0. Content-Length: 0. . U 2014/07/23 19:17:08.129076 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.129622 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.130107 1.2.3.4:5060 -> 2.3.4.5:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Record-Route: <sip:1.2.3.4;lr=on;ftag=838449717;nat=yes>. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Content-Type: application/sdp. Content-Length: 499. . v=0. o=yate 1406132227 1406132227 IN IP4 1.2.3.4. s=SIP Call. c=IN IP4 1.2.3.4. t=0 0. m=audio 21888 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. a=nortpproxy:yes. U 2014/07/23 19:17:08.130593 2.3.4.5:5060 -> 1.2.3.4:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0;received=1.2.3.4;rport=5060. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.130770 1.2.3.4:5060 -> 2.3.4.5:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Content-Length: 0. . U 2014/07/23 19:17:08.131361 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.149847 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. User-Agent: YATE/5.3.0. Content-Length: 0. — And kamailio.cfg attached: proLogika Sent with Airmail

2 1

MESSAGE forking bug(?)
by Allen Zhang 24 Jul '14

24 Jul '14

Hi, I believe there is a bug on MESSAGE forking. Test scenario: User A has two records in the location table, with different instance ids: 111 and 222. One of user A's instance is killed, hence 222 is not reachable. User B send a MESSAGE to user A. Kamailio forked the MESSAGE. Kamailio got a '200 Message delivered' response from 111 and forwarded it to user B. The MESSAGE forked to 222 timed out. Kamailio send a '408 Time out' to user B. This violates RFC 3428 section 4. It states that the forking UAS should only send ONE final response to the sender UAC. Forking is done by t_relay(), and the registrar parameter 'append_branches' is set to default value 1. Did I do something wrong, or there is a bug? Regards, Allen

2 5

Late parallel forking
by Guillaume Bour 24 Jul '14

24 Jul '14

Hi, I would like to do some kind of "late parallel forking" (see the example below). Is there a way to do such thing with Kamailio ? Here is an example: . A call B . B device 1 is connected so INVITE is forwarded to it . B device 2 registers succesfully . as device 1 as not yet answered and INVITE transaction is not expired, I want INVITE message to be forwarded to device 2 (so both are ringing) Regards, Guillaume Bour

4 6

Outbound registration?
by James Cloos 23 Jul '14

23 Jul '14

I don't see anything relevant in the docs, nor greping the src, but in case I missed something: Does the core, or do any of the modules support originating registration requests to other proxies a/o endpoints? I'd like to move registration responsibility to kama, and have it add a header on incoming INVITEs and the like from anything with which it has REGISTERed indicating which outbound registration is relevant to said request. If I'm right that there isn't support for that, do any of the app modules expose enough sip capability easily to write such? -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

3 4

Configuration kamailio + siremis
by Tomasz Sidło 23 Jul '14

23 Jul '14

Hello all I'm looking for the person who configures my kamailio + siremis I'll pay for the full configuration. Kamailio is to work together with a asterisk if anyone willing please contact me mail tomasz(a)sidlo.pl

1 0

Kamailio RtpProxy MHomed
by Pascal Fautré 23 Jul '14

23 Jul '14

Hi, I tried to use Kamailio / RTPProxy in mhomed setup without any luck. I had no problem to configure it with only 1 interface, without mhomed, everything worked perfectly. The RTP streams where not established correctly even if I managed to have to proper IP in the SIP INVITE (C & O). Versions: version: kamailio 4.1.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 04:23:19 Jun 13 2014 with gcc 4.7.2 RTPProxy -v: Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Here is my RTPProxy config (/etc/default/rtpproxy) : CONTROL_SOCK=udp:127.0.0.1:7722 EXTRA_OPTS=“-l PU.BL.IC.IP/PRI.VA.TE.IP -m 11000 -M 12000 -d DBUG:LOG_LOCAL3 Here are snippets of my kamailio.cfg: port=5060 mhomed=1 # 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; xlog("NATMANAGE M=$rm OU=$ou RURI=$ru RD=$rd F=$fu T=$tu NH=$nh(d) IP=$si ID=$ci\n"); if(dst_ip == PUBLIC_IP) { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coei\n"); rtpproxy_manage("coei", PRIVATE_IP); } else { xlog("NATMANAGE coee\n"); rtpproxy_manage("coee", PUBLIC_IP); } } else { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coii\n"); rtpproxy_manage("coii", PRIVATE_IP); } else { xlog("NATMANAGE coie\n"); rtpproxy_manage("coie", PUBLIC_IP); } } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } Calls were correctly going to the desired rtpproxy_manage options. Now I’m not quite sure I’m using the correct ones. I had to specify the PUBLIC_IP or PRIVATE_IP in the rtpproxy_manage calls in order to have the correct IP address in the C and O headers of the SIP INVITE. Without that, the public IP would be sent as C and O params to phones on the private subnet. In fact not a single call direction would give correct RTP streams. Any idea where I missed the turn? Cheers

4 10

core:drop_requests meaning
by Alex Balashov 23 Jul '14

23 Jul '14

Hi, I was wondering what precisely these stats mean: core:drop_replies = 0 core:drop_requests = 3 Assuming one does not explicitly drop() requests anywhere in script, would a continuous monotonic increase in drop_requests be considered a problem? Or are there some normal circumstances and ordinary behaviours (i.e. auth challenges) that would register a request drop? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 3

server for git repository is down
by Daniel-Constantin Mierla 22 Jul '14

22 Jul '14

Hello, a short note to say that the server for git repository on sip-router.org is down. The hosting provider hasn't reported any maintenance, I am digging further into it to see what happened. Meanwhile, the mirror on github can be used if anyone wants to fetch the source tree: - https://github.com/kamailio/kamailio Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

How install Kamailio with Asterisk
by ANTHONY HESNAUX 22 Jul '14

22 Jul '14

hello all, I would install kamailio with ASTERISK in realtime on CENTOS 6.5 x64. I followed the tutorial aspito: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but its not work. I have a problem with kamailio.cfg Its not possible to start kamailio When I loadmodule "uac.so" I want to make this configuration: SIP phone1 -> Register Kamailio -> ASTERISK Register SIP phone 2 -> Register Kamailio -> ASTERISK Register kamailio: 192.168.0.100 Port: 5060 ASTERISK: 192.168.0.100 Port 5080

3 6

Unknown caller gets online user's identity
by g.aloitus＠gmail.com 21 Jul '14

21 Jul '14

Hello, I'm using Kamailio version 4.1.4+precise (amd64). I have followed "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). One main difference in my setup compared to that one is that I continued use of Kamailio's database. The problem is as follows: I decided to put Kamailio and through it Asterisk reachable from internet. I have tried to configure Asterisk so that only calls of registered users would be possible, and they could only call to other registered users or conference rooms and echo test number. Then I took the following steps: I ensured that there was no online users with kamctl online. Then I launched MicroSIP (www.microsip.org), but I did not defined account, I simply set the protocol to tls and media encryption to mandatory, because I'm using these. I called to extension with xxx(a)my.public.ip.address (where xxx is extension) getting "unauthorized". And that was what I wanted. But if there is online users, calls go through, and incoming call is coming from Asterisk (in syslog I can find out that src_user=asterisk). Kamailio and Asterisk are listening the same IP address, but different port. I have refused connections to the Asterisk's port with iptables. I have defined my public IP address as domain in sip.conf. There is also other domain defined which corresponds to users' domain I am using in Kamailio's database. In kamailio.cfg there is if statement which prevents Kamailio not to be open relay: if (from_uri!=myself && uri!=myself) ... If I change this for example: if (from_uri!=myself || uri!=myself) I get what I want this time: no calls from outside, but I somewhat think that this is not a final solution. I have not found from log files such information which would have helped me. I have not yet investigated this problem so much that I could tell the logic behind the selection of online user's identity which is used. However, if I make a call to conference room I notice that Asterisk is thinking that one of online users has joined the conference. If I can recall correctly, I started with Kamailio version 3.2, and integrated it with Asterisk 11 (currently 11.10.2). Is there something which has changed in Kamailio, but what I have not changed in my setup which could explain this. Best, Teijo

6 18

kamcmd htable access problem, probably with int->string conversion
by Moritz Graf 21 Jul '14

21 Jul '14

Hi There, i implemented a mapping via htable. For debugging purpose I use kamcmd to see if the values are correctly filled. Doing so I discovered that there is a problem getting the values of the keys, where the name is a integer value. So it's the following behaviour: ##################################################################### root@kamailio-stage:~# kamcmd htable.dump maptodest { entry: 55 size: 1 slot: { item: { name: 1 value: 0000000000000000001 } } } { entry: 504 size: 1 slot: { item: { name: blub value: 0000000000000000001 } } } { entry: 686 size: 1 slot: { item: { name: 001 value: 0000000000000000001 } } } root@kamailio-stage:~# kamcmd htable.get maptodest blub { item: { name: blub value: 0000000000000000001 } } root@kamailio-stage:~# kamcmd htable.get maptodest 001 error: 500 - Not enough parameters (htable name and key name) ##################################################################### Is there a reasonable explanation why it works with blub, but not with 001? Why is he complaining about the number of parameters? Does anyone know a workarround? Thanks for any advise!! greeetz Moritz -- Moritz Graf, M.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg

2 3

pkg.stats
by Vitaliy Aleksandrov 21 Jul '14

21 Jul '14

Hi list, I have a problem with pkg.stats for tcp.main process on kamailio "4.1.2 73ea61-dirty". kamailio has 64M of private memory for each process (-M 64). pkg.stats for tcp.main process shows that is uses 75M and still has 62M free: { entry: 36 pid: 10032 rank: -4 used: 73109720 free: 65891192 real_used: 79146048 } root@webRTC:~# ps -p 10032 -o pid,tid,state,pcpu,pmem,rss,size,sz,vsz,args PID TID S %CPU RSS SIZE SZ VSZ COMMAND 10032 10032 S 0 26420 67508 76821 307284 /.../kamailio -f /.../kamailio.cfg -P /.../kamailio.pid -m 128 -M 64 -u kamailio -g kamailio This server handles 1000+ tcp/tls connections and works just fine for the last two months. That's why i think this is not a memory leak and just a pkg.stats problem. Does anybody else saw this or maybe somebody is already working to fix it ?

2 3

Msilo not delivering message (status 408)
by Peter Villeneuve 21 Jul '14

21 Jul '14

Hi, This is most likely a stupid mistake on my route config, but msilo correctly stores messages for offline users, but when they come back online, the message never arrives. Looking through the debug logs indeed there is an error, with status 408, which suggests kamaiio can't find a route to the user that just logged. Before I spend hours looking through all the routing blocks, perhaps someone more exprienced in kamailio can see the problem right away and offer a solution? Thanks DEBUG: msilo [msilo.c:1148]: m_dump(): msg [1-12] for: sip:1010@my.domain.com DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e3040, 9, -1, 0x10) fd_no=2 called DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f61ced1b1b0, state 1, fd=9, id=1 DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f61ced1c4f8 DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f61ced1b1b0, 1 from 0 DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89ec40, 26, 2, 0x7f61ced1b1b0), fd_no=21 DEBUG: <core> [tcp_main.c:3515]: handle_tcp_child(): handle_tcp_child: CONN_RELEASE 0x7f61ced1b1b0 refcnt= 1 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced1d868, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): *completed with status 408* [mid: 140058058335472/13] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <13> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:13 fl:8 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced5f098, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): completed with status 408 [mid: 140058058329184/12] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <12> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:12 fl:8

4 19

please help with Msilo config
by Miguel Rios 19 Jul '14

19 Jul '14

Hi list, I'm a newbie when it comes to kamailio, although I have a fair amount of SIP experience. I'm trying to setup a very basic kamailio install (4.1 on Wheezy) with Msilo support. I just used the default kamailio.cfg file (changing obviously the relevant parameters for my setup) and have not touched the routing blocks. I'm very confused about how routing works, and I don't have a background in C. I've read the wiki up and down and have looked at inumerous tutorials spread out on the internet (most of which just add to my confusion because they seem out of date), but I still don't get it. I managed to setup a working kamailio server where local users can call each other fine. Now when I try to add the msilo module, and specially the routing example from MSILO Module is when things get tricky. I'm sure my error is a fairly basic one and has to do with the routing logic and syntax. Could some kind soul please share a copy of the whole ####### Routing Logic ######## on downwards for a basic no frills kamailio with msilo setup? Thanks, Miguel MSILO Module 3. Parameters 3.1. db_url (string) Database URL. Default value is “mysql://kamailio:kamailiorw@localhost/kamailio”. Example 1.1. Set the “db_url” parameter View on kamailio.org Preview by Yahoo

3 3

Fwd: Msilo not delivering message (status 408)
by Peter Villeneuve 18 Jul '14

18 Jul '14

Thanks Daniel. UDP is the same story as TCP and TLS, unfortunately. Looking through the logs, I see strange errors in parsing the contact header when msilo prepares to send the notification back to 1000. This is for the notification message that should have been sent back to 1000 after he sent a message to 1010 (offline). I also see that kamailio reports that 1000's location was not found, even though he's obviously registered. The more I play around with this, the more confused I become. Do you think this contact parsing error could be the root of the problems? Cheers, Peter Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:1000@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK90dc.68116a97000000000000000000000000.0>; state=16 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:193]: receive_msg(): preparing to run routing scripts... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=10 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [28]; uri=[sip:1000@my.domain.com] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [< sip:1000@my.domain.com>#015#012] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=25 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-bb2e Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: *ERROR: <core> [parser/contact/contact.c:175]: skip_name(): skip_name(): Error in contact, scheme separator not found* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/contact.c:202]: parse_contacts(): parse_contacts(): Error while skipping name part Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: nathelper [nhelpr_funcs.c:300]: get_contact_uri(): failed to parse Contact body Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: !------------------------------------>WE HAVE A NEW MESSAGE Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: registrar [lookup.c:181]: *lookup(): contact for [1000] found by address* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==9 && [79.my.server.IP1] == [127.0.0.1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==14 && [79.my.server.IP1] == [79.my.server.IP1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: ------------------MESSAGE PROCESSING: *location not found*, storing offline message ID: (sip:1000@85.xx.xx.xx:59287;transport=TLS) Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:605]: m_store(): ------------ start ------------ Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:648]: m_store(): using user id [sip:1000@85.xx.xx.xx:59287;transport=TLS] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:732]: m_store(): 'From' header: <sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:849]: m_store(): message stored. T:<sip:1000@my.domain.com> F:< sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:858]: m_store(): sending info message. Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:899]: m_store(): contact header found On Thu, Jul 17, 2014 at 11:01 PM, Daniel-Constantin Mierla < miconda(a)gmail.com> wrote: > You should not call m_store() if src_ip==myself -- in this way you avoid > to store notification messages that cannot be delivered. > > Why notifications are not delivered has to be troubleshooted there. Have > you tried also with UDP? > > Cheers, > Daniel > >

1 0

[OT] LVUG, the London VoIP User Group
by Richard Brady 18 Jul '14

18 Jul '14

Hi folks LVUG is a monthly get-together in London to discuss open-source VoIP technology, including Kamailio. If you're based in or near London please join us this coming Tuesday for the third LVUG which will be held in Old Street. The format is a couple short talks followed by beers at the venue or in a nearby pub. Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/ Hope to see you there! Richard -- Richard Brady E: rnbrady(a)gmail.com

1 0

Kamailio presence (NAT, XCAP)
by Radu Vatav 18 Jul '14

18 Jul '14

Hello, I'm just trying to use SIP as a replacement for Skype, nothing more than that. All is working except of presence. Using Jitsi 2.4.4997 (Windows x64) from home behind NAT and kamailio 4.1.4+wheezy on a rented server somewhere. My biggest problem is that I don't know exactly how it is supposed to work. Ok, this is what I understood so far: the clients connect via SIP to the server and can call each other, but for some reason the server can't notify them via SIP of presence changes when they are behind NAT. So the clients also connect to the built in xcap-server, register themselves as presence "watchers" and the xcap-server notifies them via udp/tcp of presence changes of the other users which are watched. All is very nice but it's not working and I don't know where to look for the problem. Which are exactly the steps in communication and what data exactly is there sent which way (when working correctly, which it doesn't) - no clue. I somehow put the configuration together from the 2-3 examples I found on the internet, I don't know if it is ok to attach something to this email, so here it is: https://vatav.cc/kam/kam_cfg.7z Here is the network activity on the server: https://vatav.cc/kam/kam_ngrep.7z So me ("radu") and "alice" are home behind the NAT, one is online and wants to see how the other one goes online and offline. I don't know what exactly to look for, but it doesn't look to me like there is anything sent to one user when the other one goes on-/off-line (there is a lot of "proxy authentication required" which drives me nuts, again I don't know if this is correct or not; Jitsi seems to be connected to the XCAP server, if I type in a wrong address for "XCAP server" Jitsi complains it can't connect). Some other things: - "kamctl online" shows almost real-time who is logged-in, so this works - siremis shows a bunch of users in "watchers list", but none in "active watchers list" - actually when a user goes online it contains the contacts of that user, but only briefly, when I refresh the view, the list is empty again. I don't know if this is how it is supposed to work, anyway, Jitsi doesn't show any change of the status of the user who changed status at all. Any help would be appreciated. It's sad something this simple as I need (basic Skype replacement) is so difficult to get working. Maybe this is why there is relatively low interest in SIP and everybody stays with MS/Google/... because "there isn't something else". Best regards, Radu Vatav

2 1

db_postgres Error
by Rene Montilva 18 Jul '14

18 Jul '14

Hi list i'm config SQLOps Module with postgresql db module or is postgresql driver? this is my kamailio config: #!ifndef DBURL #!define DBURL "postgres://user:ouser@x.x.x.x/kamailio" #!endif sql_query("ca", "select * from xxxxx","ra"); if($dbr(ra=>rows)> 0){ xlog("L_INFO", "saldo:$dbr(ra=>[0,0]) y seg:$dbr(ra=>[0,1]) "); $var(credit) = $dbr(ra=>[0,0]); # 30$ of credit $var(cost_per_sec) = $dbr(ra=>[0,1]); # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second sql_result_free ("ra"); if($var(cost_per_sec) > $var(credit)) { sl_send_reply("402","payment required"); exit; } } i got this warning and would like why this happen: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [SSL SYSCALL error: Connection timed out#012] WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [SSL SYSCALL error: EOF detected#012]

2 1

how to m_store to database not just a "body" with MSILO?
by denis kremov 18 Jul '14

18 Jul '14

Hello All, Subj, need you help, How to store this type of message to MSILO database? Content-Type: message/external-body;access-type=URL;URL="https://xxx.com//download/539c28884_3e071c9db68044f687bf.jpg" I modified and recompiled msilo.c to resolve this error: ERROR: msilo [msilo.c:626]: m_store(): body of the message is empty! but after that - m_store just store this information to database: mysql> select * from silo; +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | id | src_addr | dst_addr | username | domain | inc_time | exp_time | snd_time | ctype | body | extra_hdrs | callid | status | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | 30 | sip:777351300@xxx.in | sip:777581288@xxx.in | 777581288 | xxx.in | 1402479947 | 1403479946 | 0 | message/external-body | | | | 0 | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ 1 row in set (0.00 sec) How can I store to database "access-type=URL" and and URL="xxxx" ? Thanks All! -- denis kremov

3 2

Groups in Kamailio vs Freeswitch
by Peter Villeneuve 18 Jul '14

18 Jul '14

Hi guys, I'm designing a new service for a client and I was wondering what your opinion is of the 2 options I'm considering to separate users on the server. Basically I want to create closed groups where each user can only call and receive calls from members of the same group (single domain install). Initially I was thinking of just forwarding all the invites to FS and using the dialplan to enforce this segregation, but now I realize there is a groups module in kamailio too. Which 2 courses of action do people recommend? I'm more familiar with FS than Kamailio, so it would be easier for me to just forward everything to FS, although that will be more expensive and probably unnecessary if I could do it all in Kamailio with Siremis. Would kamailio's group module do the trick? Are there any tutorials out there I can follow? Cheers, Peter

2 1

trying to conifgure telepresence client software
by Fred Hunter 18 Jul '14

18 Jul '14

Hi, I recently registered with IP tel for the purposes of carrying out SiP based teleconferencing. I am attempting to register my software client, Polycom m100 desktop but am having some difficulty. Can you tell me what I should be entering into the 'registrar server' field, 'domain' and 'registrar server'? [cid:image001.png@01CFA03E.F01888F0] ---------------------------------------------------------------------------- www.puntersouthallgroup.com Punter Southall Group is a trading name of Punter Southall Group Limited. Registered office: 11 Strand, London, WC2N 5HR. Registered in England and Wales No. 04096788. This message and any attachments are confidential. If you have received this message in error please notify the sender and delete this email immediately. Unauthorised dissemination, disclosure, copying or use is prohibited. This email has been scanned for viruses and other malicious items. We reserve the right to monitor the content of all emails sent and received by our systems. Please consider the environment before printing this email. ----------------------------------------------------------------------------

2 1

Re: [SR-Users] how to remove non-digits using dp_translate
by Daniel-Constantin Mierla 18 Jul '14

18 Jul '14

Hmm, I see there is none inside dialplan for the moment. The option would be to do it in config file with a function from textops. Also, check if regex module is a good alternative for you: - http://kamailio.org/docs/modules/stable/modules/regex.html Cheers, Daniel On 16/07/14 16:50, Henry Fernandes wrote: > Can anyone show me how to use the regexp global flag using dp_translate? > -H > > From: Henry Fernandes <henry(a)usinternet.com <mailto:henry@usinternet.com>> > Reply-To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Date: Friday, 11 July, 2014 4:18 PM > To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > I don't know how to apply the global flag in Kamailio. Can you give > me an example? > -H > > From: Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > Reply-To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio > (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Date: Thursday, 3 July, 2014 1:47 AM > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > Hello, > > you have to replace [^0-9] with empty, using the global flag. > > Cheers, > Daniel > > On 02/07/14 23:11, Henry Fernandes wrote: >> I'd like to use the dialplan module to remove all non-digits from a >> string. For example, "1-306-555-1234" becomes "13065551234" and >> "(306) 555-1234" becomes "3065551234". >> >> Unfortunately, I can't figure out the regular expressions for >> subst_exp and repl_exp. I can easily replace one instance of a >> non-digit, however I can't figure out how to replace all non-digits. >> >> -H >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.comhttp://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 > <mailto:sr-users@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 <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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Some of the calls drops after 15 minutes + some seconds
by Andras FOGARASI 18 Jul '14

18 Jul '14

Hi, I have a simple kamailio install (2 servers, using location service and a failover node with dispatcher, STUN, clients behind different NATs), without rtpproxy, only peer-to-peer RTP and TURN server if the connection is really messy (it's not relevant here). Signaling is over TLS. Both of the clients are behind NAT. Basically everything works, but some of the calls are dropped after 15 minutes and some seconds, for me it seems the RTP connection is dropped but not sure. I cannot find find out wether it's a kamailio problem or client/NAT problem. Has anyone idea what is going on? Thanks, Andras

3 8

Really need help on kamailio-snmpstats module
by Allen Zhang 17 Jul '14

17 Jul '14

Hi This is a re-post. I'm desperate. What I'm trying to do: Use rpmbuild to build kamailio rpms from source code. We made a lot of changes in the kamailio code. I can't use the rpms from the build service. The plan is to push these changes to kamailio but that's not done, yet. The problem is when I try to install kamailio-snmpstats module from the rpm built, it reports error: Error: Package: kamailio-snmpstats-4.0.0-0.el6.x86_64 (company_name) Requires: net-snmp-agent-libs I've installed net-snmp on the build machine and the machine that needs to install kamailio-snmpstats. I have tried to use the kamailio.spec file from kamailio build server but it didn't solve the problem. Can anyone help, please? Regards, Allen

2 6

Condition Statements - How many levels
by Joel White 16 Jul '14

16 Jul '14

I was wondering how many levels deep in conditional _if_ _else_ statements can go. I need to create a statement that has 4 possible outcomes and need to know if this is possible. I have two variables I want one thing to happen if only the first variable is present Another if only the second variable is present Yet another if they are both present And a fail over if None of the variables are present I do see that is can be done with 3 conditions using _if_ _if else_ _else_ how can I accomplish 4 conditional statements? Thank you in advance Joel

2 2

Re: [SR-Users] how to remove non-digits using dp_translate
by Henry Fernandes 16 Jul '14

16 Jul '14

Can anyone show me how to use the regexp global flag using dp_translate? -H From: Henry Fernandes <henry(a)usinternet.com> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Friday, 11 July, 2014 4:18 PM To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate I don't know how to apply the global flag in Kamailio. Can you give me an example? -H From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Thursday, 3 July, 2014 1:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate Hello, you have to replace [^0-9] with empty, using the global flag. Cheers, Daniel On 02/07/14 23:11, Henry Fernandes wrote: > > I'd like to use the dialplan module to remove all non-digits from a string. > For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" > becomes "3065551234". > > > > > Unfortunately, I can't figure out the regular expressions for subst_exp and > repl_exp. I can easily replace one instance of a non-digit, however I can't > figure out how to replace all non-digits. > > > > > -H > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listi > nfo/sr-users > -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/lis tinfo/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

rtpengine upgrade fails
by Juha Heinanen 16 Jul '14

16 Jul '14

i build debian packages for rtpengine branch 3.3.0.0+0~mr3.4.1.0. after that i did 'apt-get dist-upgrade' and it failed like this: Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Installing new version of config file /etc/init.d/ngcp-rtpengine-daemon ... Restarting RTP/media proxy: ngcp-rtpengine-daemonFATAL: Module xt_MEDIAPROXY not found. iptables: No chain/target/match by that name. ip6tables: No chain/target/match by that name. FAILED TO CREATE KERNEL TABLE 0, KERNEL FORWARDING DISABLED invoke-rc.d: initscript ngcp-rtpengine-daemon, action "restart" failed. dpkg: error processing ngcp-rtpengine-daemon (--configure): subprocess installed post-installation script returned error exit status 255 however, if i now run 'apt-get install ngcp-rtpengine-daemon', setting up ngcp-rtpengine-daemon succeeds and also packages that depended on it get installed fine: Reading package lists... Done Building dependency tree Reading state information... Done ngcp-rtpengine-daemon is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. 2 not fully installed or removed. After this operation, 0 B of additional disk space will be used. Do you want to continue [Y/n]? Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Restarting RTP/media proxy: ngcp-rtpengine-daemon. Starting RTP/media proxy: ngcp-rtpengine-daemon already running. Setting up ngcp-rtpengine (3.3.0.0+0~mr3.4.1.0) ... any ideas why setting up ngcp-rtpengine-daemon fails with dist-upgrade? -- juha

2 1

KAMILIO-4.1.4 REGISTRATION ISSUE
by Cibin Paul 16 Jul '14

16 Jul '14

I have kamailio 4.1.4 installed with DB Engine as mysql. I added users using the command kamctl add and I could see the entry in subscriber table. When I am trying to register, kamailio is accepting any combination of username and password not listed in the table. What could be wrong? I followed the INSTALLATION document provided in the source file. Please help me troubleshoot this. Kamailio is up and running and not throwing any errors. kamctl moni shows 1 user registered. But the registered user doesn’t even exist Server Details OS : Centos 5.5 Kamailio version - 4.1.4 MySQL version - 5.0.95 Cibin

2 4

Rtpproxy redundancy
by Dren Daka 16 Jul '14

16 Jul '14

Hi, I'm using kamailio with two network interfaces also running rtpproxy on the same interfaces so rtpproxy is load balancing, when one of the connections is down rtprpoxy still sends 50% of the calls in that interface. How can i make rtprpoxy to do failover from the non working connection to the working one instead of load balancing. Thanks.

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