sr-users April 2014

sr-users@lists.kamailio.org

100 participants
171 discussions

permanent 1.5 s timer interval in module pipelimit
by Julia 28 Apr '14

28 Apr '14

Hello all, Timer pl_timer in pipelimit.c is defined as MS_TO_TICKS(1500) instead of timer_interval (in versions 3.3 - 4.1). I found a patch that solves the same problem in the ratelimit module: Module: sip-router Branch: 3.1 Commit: f027ca95a2a1818cd40e4ed6d4358a76d18252ad URL: <http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=f027ca9 5a2a1818cd40e4ed6d4358a76d18252ad> http://git.sip-router.org/cgi-bin/gitweb.cgi/... Author: Marius Zbihlei Committer: Marius Zbihlei Date: Mon Apr 11 13:45:10 2011 +0300 modules/ratelimit Timer was ran at a 1.5 s interval instead of timer_interval, thus rendering TAILDROP alg unusable (cherry picked from commit 3a06d7d3561f889c84249eaa5ec7a5b79a2a23bd) Regards, Julia

2 1

kamctl not connect to dbserver
by Юрий Горличенко 28 Apr '14

28 Apr '14

Hello. I started configure Kamailio 3.3 server with mysql database, that running on the remote server. I customise settings at kamailio.cfg (for BDURL) and kamctlc file (for my db - changed user, password, dbhost and other settings, such as dbengine and etc.) Now, when I tried add new subscriber from kamctl util (kamctl add newuser@kamailioIP passwd). I see "access denied for <myuser>@<interface of kamailio server>", But as I aleady says I configured filew with other settings for db server. Where I must configure ip of dbhost to connect kamctl to my dbserver?

2 1

kamailio command line
by betul.cerit＠netgsm.com.tr 28 Apr '14

28 Apr '14

Hi to all, I’m using “kamailio” command to see alerts,notices and warnings but it also starts a new kamailio process.Is there a way to see command line without starting a new kamailio process? I need something just like asterisk –rvvvvvvvvvvv in asterisk. Thanks in Advance.

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No secure attributes from rtpengine in SRTP/RTP bridge mode
by Alexey Rybalko 28 Apr '14

28 Apr '14

Hello! I have been experimenting with drop-in replacement of old "rtpproxy-ng" module with new "rtpengine". Wondering what is wrong in my configuration: there are no security attributes in rtpengine answer on RTP/SAVPF offer. Neither "fingerprint" (DTLS) nor "crypto" (SDES). I used Firefox 29 during this test. 1. Here's original offer : INVITE sip:user5@..... SIP/2.0 > Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 > Max-Forwards: 69 > To: <sip:user5@......> > From: "user4" <sip:user4@......>;tag=dvf1co8urv > Call-ID: phmq9o62cv21timhfnpf > CSeq: 4233 INVITE > Proxy-Authorization: Digest algorithm=MD5, username="user4", > realm="......", nonce="U1o8H1NaOvP3wxegsYCKOJX7S7DV/r1N", uri="sip:user5@......", > response="44e7b16c55d3237f63e04b3c0b194f45" > Contact: <sip:user4@ > ......;gr=urn:uuid:c193bcd4-aa2e-47ef-a106-22e60f5fde9e;ob> > Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE > Content-Type: application/sdp > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.7 > Content-Length: 607 > > v=0 > o=Mozilla-SIPUA-29.0 15825 0 IN IP4 0.0.0.0 > s=SIP Call > t=0 0 > a=ice-ufrag:2102f082 > a=ice-pwd:8733e5248a7fb087b40ea45b3ca6f634 > *a=fingerprint:sha-256 > 32:AA:85:DB:D8:3C:E6:C3:46:07:11:9E:9F:54:B9:42:7F:5C:37:5F:9D:D1:AD:19:22:A3:AC:9C:6C:A5:A6:CD* > m=audio 62290 *RTP/SAVPF* 109 0 8 101 > c=IN IP4 ..... > a=rtpmap:109 opus/48000/2 > a=ptime:20 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > ..... > Here's the snippet for translation SRTP-RTP: rtpengine_offer("force trust-address symmetric replace-origin replace-session-connection ICE=force_relay *RTP/AVP*"); 2. Here's final answer (from rtpengine): SIP/2.0 200 OK > Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 > Record-Route: <sip:......;lr;nat=yes> > Record-Route: <sip:.....:15060;transport=udp;lr;ovid=3207d8cd> > Record-Route: <sip:95f6551e81@.......:10080;transport=ws;lr;ovid=3207d8cd> > Contact: <sip:user5@......1:49362> > To: <sip:user5@.....>;tag=4d436110 > From: "user4"<sip:user4@....>;tag=dvf1co8urv > Call-ID: phmq9o62cv21timhfnpf > CSeq: 4233 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces, eventlist > > User-Agent: X-Lite release 4.5.4 stamp 70864 > Content-Length: 432 > > v=0 > o=- 1398422264455879 3 IN IP4 ..... > s=X-Lite 4 release 4.5.4 stamp 70864 > c=IN IP4 ...... > t=0 0 > m=audio 30002 *RTP/SAVPF* 109 0 8 101 > a=rtpmap:109 opus/48000/2 > a=fmtp:109 useinbandfec=1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=rtcp:30003 > .... > Error from JsSIP: {name: "INTERNAL_ERROR", message: "Could not negotiate answer SDP; cause = > *NO_DTLS_FINGERPRINT*", __exposedProps__: Object} > Here's the snippet for translation RTP-SRTP: rtpengine_answer("force trust-address symmetric replace-origin replace-session-connection rtcp-mux-demux ICE=force *RTP/SAVPF* "); There was another test with Chrome 34 with the same result. Offer: INVITE sip:user5@.... SIP/2.0 Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 Max-Forwards: 69 To: <sip:user5@....> From: "user4" <sip:user4@....>;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: 957 INVITE Proxy-Authorization: Digest algorithm=MD5, username="user4", realm=".....", nonce="U1pKZ1NaSTtLo2vjK9TGyBu6Axb+EtyN", uri="sip:user5@...", response="98f6275d3636664b611ff1411af982af" Contact: <sip:user4@....;gr=urn:uuid:8cd4e797-7314-485f-b191-4a15a6581c42;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 1586 v=0 o=- 7510391807340598328 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS ErnvtgCDe9aR6LWxNRgT83r0mxtyAV87LUxT m=audio 51672 *RTP/SAVPF *111 103 104 0 8 106 105 13 126 c=IN IP4 10.61.2.151 a=rtcp:51672 IN IP4 .... ...... a=ice-ufrag:EMn7uHfSS7ulRGU2 a=ice-pwd:FskTdhj7qT6ELP7uTIb+gquQ a=ice-options:google-ice *a=fingerprint:sha-256 46:6E:E0:18:4A:C5:06:A8:26:85:ED:FE:16:C1:86:5E:8D:BC:4D:D9:F2:1A:75:81:A1:A7:CE:5A:79:4D:B7:22*a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux *a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:R6qaiU7Cm471zNF6f3Q87TyXbHjEt/VhLgUgY2ZZ a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lch+mfN/hi9QmseWu+ss1M2vA8mwRh8GQYChaJvc* a=rtpmap:111 opus/48000/2 .... Answer: SIP/2.0 200 OK Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 > Record-Route: <sip:...;lr;nat=yes> Record-Route: > <sip:....:15060;transport=udp;lr;ovid=3207d8cd> Record-Route: > <sip:712a450958@...:10080;transport=ws;lr;ovid=3207d8cd> Contact: < > sip:user5@10.61.2.151:49362> To: <sip:user5@....>;tag=b8c8ee57 From: > "user4"<sip:user4@...>;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: > 957 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, > MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, > eventlist User-Agent: X-Lite release 4.5.4 stamp 70864 Content-Length: 432 > v=0 o=- 1398425917116411 3 IN IP4 ... s=X-Lite 4 release 4.5.4 stamp 70864 > c=IN IP4 .... t=0 0 m=audio 30006 *RTP/SAVPF* 111 0 8 126 a=rtpmap:111 > opus/48000/2 a=fmtp:111 useinbandfec=1 a=rtpmap:126 telephone-event/8000 > a=fmtp:126 0-15 a=sendrecv a=rtcp:30007 .... > Error is JsSIP: Failed to set remote answer sdp: Called with a SDP without crypto enabled > Any help on this issue would be greatly appreciated! best regards, Alexey

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Double attempt to record-route
by Juha Heinanen 27 Apr '14

27 Apr '14

I tried to call record_route_preset() in branch route in order to add contact specific RR headers when request is forked to more than one contact. What happened was that I got error message: Sep 26 17:03:47 wheezy2 /usr/sbin/sip-proxy[2436]: ERROR: rr [rr_mod.c:256]: Duble attempt to record-route Is it really so that record routing is not a branch specific thing? Is there a way to solve this problem? -- Juha

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Kamailio+IMS+WebSocket
by mohammed alyaseen 26 Apr '14

26 Apr '14

Hi folks, I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF.. I did trace on the PCSCF and I found a strange signaling behaviour. the invite is negotiated between the IMS components as follows: INVITE from PCSCF to SCSCF INVITE from PCSCF to PCSCF PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF" SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235" The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000 Can anyone help me get a solution for this problem Thanks in advance Best regards, Medo Mohammed Al-khameri Technical University Berlin TUB

1 0

Re: [SR-Users] cnxcc
by Andrzej Ciupek 26 Apr '14

26 Apr '14

Hello I’am using default config with options from an example. It looks like event route isn’t execuded after hangup. Apr 26 16:26:14 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:15 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:16 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:17 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:18 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:19 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4611]: INFO: <script>: 1000 has 1 call(s) Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: Setting up money based credit control Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: Setting up channel based credit control Apr 26 16:26:20 kamailio /usr/local/sbin/kamailio[4604]: INFO: <script>: CNXCC ROUTE: 1000 has 2 call(s) kamailio.cfg … … # ----- prepaid call controll ----- #!define DLG_FLAG 28 #!define CC_FLAG 29 loadmodule "dialog.so" modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("rtimer", "timer", "name=ta;interval=1;mode=1;") modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT") loadmodule "cnxcc.so" modparam("cnxcc", "dlg_flag", CC_FLAG) modparam("cnxcc", "credit_check_period", 1) #check every 1 second … … … route { … … if (!allow_source_address("2") and !allow_source_address("1")){ route(CNXCC); route(PSTN); } … } route[CNXCC] { $var(client) = "1000"; xlog("L_INFO", "Setting up money based credit control"); $var(credit) = "10"; # 10$ of credit $var(cost_per_sec) = "1"; # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second # if only one call is established, that call should last 9 seconds. if (!cnxcc_set_max_credit("$var(client)", "$var(credit)", "$var(cost_per_sec)", "$var(i_pulse)", "$var(f_pulse)")) { xlog("Error setting up credit control"); return; } $var(count) = -1; if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { xlog("Error getting customer's channel count"); } xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)"); if ($var(retcode) < -1) { xlog("Too many channels for customer"); sl_send_reply(403, "Forbidden"); if (!cnxcc_terminate_all("$var(client)")) { xlog("Error terminating customer's calls"); } else { xlog("Terminating customer's call"); } exit; } } route[SHOW_CHANNEL_COUNT] { $var(count) = @cnxcc.channels["1000"].count; xlog("L_INFO", "1000 has $var(count) call(s)"); } event_route[dialog:failed] { xlog("dialog failed"); } event_route[cnxcc:call-shutdown] { xlog("L_INFO", "Shutting down channel by callcontroll"); xlog("L_INFO", "[$ci]: call killed"); $var(count) = $var(count)-1; if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { xlog("Error getting customer's channel count"); } xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)"); # perform some kind of notification, database update, email sending, etc } Greetings Andrzej Ciupek Od: Carlos Ruiz Díaz Wysłano: ‎sobota‎, ‎26‎ ‎kwietnia‎ ‎2014 ‎15‎:‎19 Do: Kamailio (SER) - Users Mailing List, Andrzej Ciupek Hi, sorry guys, I forgot about this. I will make my own tests and post the results. Andrzej, can you post the configuration file you are using? Regards, Carlos On Sat, Apr 26, 2014 at 4:15 AM, Kelvin Chua <kelchy(a)gmail.com> wrote: No, I have not yet found the solution for this issue on my setup. Maybe Carlos can help us out On Apr 26, 2014 5:10 PM, "Andrzej Ciupek" <andrzej.ciupek(a)hotmail.com> wrote: Hello Have You solved this issue ? I’am using Kamailio 4.1.3 and I’am getting the same results. Greetings Andrzej Ciupek _______________________________________________ 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 -- Carlos http://caruizdiaz.com http://ngvoice.com +52 55 3048 3303

2 1

Re: [SR-Users] cnxcc
by Kelvin Chua 26 Apr '14

26 Apr '14

No, I have not yet found the solution for this issue on my setup. Maybe Carlos can help us out On Apr 26, 2014 5:10 PM, "Andrzej Ciupek" <andrzej.ciupek(a)hotmail.com> wrote: > Hello > > Have You solved this issue ? > I’am using Kamailio 4.1.3 and I’am getting the same results. > > Greetings > Andrzej Ciupek >

2 1

Kamailio+IMS+WebSocket
by mohammed alyaseen 26 Apr '14

26 Apr '14

1 0

LCR from_uri help
by Geoffrey Mina 26 Apr '14

26 Apr '14

Greetings, I need a little help with the LCR module. I am running Kamailio 4.1.2. The LCR module is behaving in a way that is inconsistent with our old 1.5 versions. We have a customer that passes a certain caller-id. That from_uri should be routing over a specific trunk. Here are the relevant bits of the LCR data from kamcmd. I would expect gw_index: 7 to match before gw_index: 6 here, but that is not the case. Any help would be greatly appreciated. Please note, i Have stripped out a bunch of other routes that are not in play here... hence the reason gw_index starts at "6". kamcmd> lcr.dump_gws { lcr_id: 1 gw_id: 11 gw_index: 6 gw_name: Route A scheme: sip ip_addr: 1.2.3.4 hostname: port: 5060 params: transport: UDP strip: 0 prefix: tag: flags: 2 defunct_until: 0 } { lcr_id: 1 gw_id: 1 gw_index: 7 gw_name: Route P scheme: sip ip_addr: 4.3.2.1 hostname: port: 5060 params: transport: UDP strip: 0 prefix: DLR# tag: flags: 0 defunct_until: 0 } kamcmd> lcr.dump_rules { lcr_id: 1 rule_id: 31 prefix: +1 from_uri: 2089328663 request_uri: stopper: 0 } { gw_index: 6 priority: 1 weight: 1 } { lcr_id: 1 rule_id: 5 prefix: +1 from_uri: request_uri: stopper: 0 } { gw_index: 7 priority: 1 weight: 1 }

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sr-users April 2014