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37525 discussions

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

17 Jun '16

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

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

17 Jun '16

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

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Kamailio Problem carrying media
by Sirvan Paraste 16 Jun '16

16 Jun '16

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

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Kamailio on multiple ports
by huseyin kalyoncu 16 Jun '16

16 Jun '16

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

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

16 Jun '16

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

2 2

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

16 Jun '16

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

1 1

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

16 Jun '16

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

1 0

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

16 Jun '16

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

2 1

Question about testing RTPENGINE
by Dmitry 16 Jun '16

16 Jun '16

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

1 1

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

16 Jun '16

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

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