- sr-users - kamailio.org

event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Parse-error: $rm from $avp(inc_carrier)/n$mb/n"); }

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Does kamailio reloads the modules per sip message ?
by Rahul MathuR 29 Sep '14

29 Sep '14

Hello, We have observed a strange behavior in corex module that it gets loaded at every sip packet which arrives to kamailio. We put a static variable and saw that it gets re-initialized to 0 everytime any sip packet comes to it. Could you please tell me how to stop it and load it just once. Thanks in advance ! -- Warm Regds. MathuRahul

2 1

DBTEXT Engine & ALIAS_DB
by Marino Mileti 29 Sep '14

29 Sep '14

Hi to all, I've setup FS with dbtext module, and now I'm trying to make a alias_db_lookup(dbaliases) but it doesn't find any alias.Same conf with mysql db works great. Is it possible to have alias with dbtext engine? Marino

2 1

tm.t_uac_wait documentation?
by Juha Heinanen 29 Sep '14

29 Sep '14

i could not find any documentation about tm.t_uac_wait rpc command. is it documented somewhere? -- juha

2 3

tm.t_uac_wait does not use advertise address
by Juha Heinanen 29 Sep '14

29 Sep '14

based on more tests, it appears that tm.t_uac_wait does not use advertise address (if given on listen line) when it substitutes !! (SUBST_CHARs) in request headers. i'll open a bug report on it. -- juha

2 1

cnxcc serial fork
by Kelvin Chua 29 Sep '14

29 Sep '14

hi carlos, i have a serial forking scenario to several providers as failover. and i used cnxcc_set_max_credit to control the session credit/time. just wanted to make sure it does what i think it should do so i made a little experiment. the logic goes this way. 1. cnxcc_set_max_credit -> $100, cost per second -> $1 2. call will fail, try the next provider 3. cnxcc_set_max_credit -> $200, cost per second -> $2 what i am expecting to see is, if i do kamctl kamcmd cnxcc.active_clients i would see $200 as the max_amount. but in kamailio 4.1.5, i still see $100, should it be updated to $200? is there an override in cnxcc_set_max_credit when called twice or after failure_route? Kelvin Chua

2 3

rtpproxy & extra_id_pv
by Marino Mileti 26 Sep '14

26 Sep '14

Hi guys, I've seen that setting the parameter extra_id_pv, every branch should be hae different callid.. How can i set this parameter? I've tried with : modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") but in the INVITE message the callid is still the same for all branches. Any suggest?

2 2

Kamailio server crashing
by aawaise 26 Sep '14

26 Sep '14

Hello, I installed kamailio and replaced kamailio.cfg with a customized kamailio.cfg and replaced kamctlrc file also. And restarted the kamailio server. Server gets started ok and after a minute or two, I get the error, " '/usr/local/kamailio/sbin' doesnot belong to any package ". I am facing this problem for quite some time now. Any solution to this will be very helpful. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-server-crashing-tp130923.h… Sent from the Users mailing list archive at Nabble.com.

2 1

Bash Code Injection and 'exec' module
by Seudin Kasumovic 25 Sep '14

25 Sep '14

Hi kamailio users, we are witnesses of new discovered bug in bash: Bash Code Injection Vulnerability via Specially Crafted Environment Variables (CVE-2014-6271) https://access.redhat.com/node/1200223 As exec module exports all SIP headers in environment so it's was easy to push bash command. There is attached simple kamailio test config file. With sipp we sent header to output 123 into file /tmp/123 like this: User-Agent: () { :;}; echo 123 > /tmp/123 Debug output from kamailio is: 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTENT_LENGTH=135 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTENT_TYPE=application/sdp 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_ALLOW=INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH * 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_USER_AGENT=() { :;}; echo 123 > /tmp/123* 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_SUBJECT=Performance Test 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_MAX_FORWARDS=70 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTACT=< sip:T00157@198.51.100.2:5060> 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CSEQ=1 INVITE 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CALLID= 1-5394(a)198.51.100.2 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_TO=+442033998806 <sip:+442033998806@orange.voip> 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_FROM=+442033998833 <sip:T00157@orange.voip>;tag=5394SIPpTag001 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_VIA=SIP/2.0/UDP 198.51.100.2:5060;branch=z9hG4bK-5394-1-0 5(30147) DEBUG: exec [exec_mod.c:175]: w_exec_msg(): executing [/bin/true] ls /tmp shows new created file !!! I created simple patch to fix this issue in exec module based on suggestion from RedHat until you fix your bash what is recommended. -- Seudin Kasumovic

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Re: [SR-Users] Kamailio - Separated Servers
by Bruno Emer 24 Sep '14

24 Sep '14

Hello Guys! Good news about my tests here! I found some configurations relative to Outbound proxy and I get this working. Now, I can have two outbound proxies, one registrar and one database. I can register one user in the Outbound Proxy1 and one user in Outbound Proxy 2 and place a call between these users. But now I'm facing another problem! The ACK messages are not being delivered, so the session is being dropped after a few seconds. I think that the problem is in the Contact field. Trying to investigate it now! I'm not sure about how to change the message to deliver the ACK. Did someone had this problem before? In anycase, here is the routing configuration that I'm using in my Oubound proxy: request_route { 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; } route(REQINIT); if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } route(WITHINDLG); t_check_trans(); if (is_method("REGISTER")) { remove_hf("Route"); add_path(); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { if (is_method("INVITE|SUBSCRIBE|ACK")) record_route(); if (@via[2] == "") { # From client so route to registrar... if ($rU == $null) { sl_send_reply("484", "Address Incomplete"); exit; } remove_hf("Route"); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { # From registrar so route using "Route:" headers... if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: xlog("L_ERR", "in request_route\n"); sl_reply_error(); exit; } } t_on_failure("FAIL_OUTBOUND"); } } route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } route[WITHINDLG] { if (has_totag()) { if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: if (is_method("ACK")) { if ( t_check_trans() ) { record_route(); route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not Found"); } } else { if (is_method("NOTIFY")) { record_route(); } route(RELAY); } exit; } } onreply_route { if (($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (!t_check_trans()) { drop; } if ($rm == "REGISTER" && $rs >= 200 && $rs <= 299) { remove_hf("Flow-Timer"); if ($(hdr(Require)[*])=~"outbound") insert_hf("Flow-Timer: FLOW_TIMER\r\n", "Call-ID"); } } failure_route[FAIL_OUTBOUND] { if (t_branch_timeout() || !t_branch_replied()) { send_reply("430", "Flow Failed"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } Thanks again!!! *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2014-09-22 12:40 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hi, Daniel and Frank! > > Thanks for your support! > Now, I changed my scenario just a lil bit to make it easy to understand > and deploy. I'm doing the following: > > 1. Created a server using the "Outbound" module configuration example, > but add WebSocket support in it. (IP Address 192.168.1.16) > 2. Created a server running as Proxy and Registrar with WebSockets > support. (IP Address 192.168.1.15) > 3. Registered two users (bob and alice(a)enterprise.com) using WebRTC. > These users sent they Register requests to the Oubount server > (192.168.1.16). > 4. I'm also using an external postgres database to store the user > informations. In my proxy/registrar server, I've configurated > "modparam("usrloc", "db_mode", 3)", so it will not store data in memory. > > > After register the users in my scenario, I can see the following > information stored in my database (output of kamctl db show location): > > id | ruid | username | domain | > contact | received | path | > expires | q | callid | cseq | last_modified | > flags | cflags | user_agent | socket | methods | > instance | reg_id > > -----+----------------------+----------+----------------+------------------------------------------------+-----------------------+------+---------------------+----+------------------------+------+---------------------+-------+--------+-------------------+---------------------+---------+-------------------------------------------------+-------- > 277 | uloc-54203f71-10bd-2 | alice | enterprise.com | > sip:v4s0prah@o35ctmdnnruv.invalid;transport=ws | sip:192.168.1.16:5060 | > | 2014-09-22 15:31:21 | -1 | udb4shpe4dnm1jsl6016s5 | 82 | > 2014-09-22 15:26:21 | 0 | 0 | JsSIP 0.4.0-devel | udp: > 192.168.1.15:80 | 783 | > <urn:uuid:75c837d1-740a-4c62-baa0-56a06149699e> | 1 > 278 | uloc-54203f71-10bc-2 | bob | enterprise.com | > sip:lr78s270@0mkf6fgtprlh.invalid;transport=ws | sip:192.168.1.16:5060 | > | 2014-09-22 15:34:02 | -1 | 5mpl9q4bkuqsnpk8r316j5 | 82 | > 2014-09-22 15:29:02 | 0 | 0 | JsSIP 0.4.0-devel | udp: > 192.168.1.15:80 | 783 | > <urn:uuid:e92ef911-1ae4-45b1-bfc9-b5752b0f2fe1> | 1 > > > So, I can assume that both users are registered and ready to start > sessions. > > Now, when I try to start a new session between then (let's say bob calling > alice) what I get in the endpoint is the following message: > > SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) > Via: SIP/2.0/WS > 0mkf6fgtprlh.invalid;rport=33154;received=192.168.1.11;branch=z9hG4bK7711221 > To: <sip:alice@enterprise.com>;tag=94c1b96a0e6eb467885b5057e1e49f9e.1970 > From: <sip:bob@enterprise.com>;tag=on056iipql > Call-ID: 5e1pmj2fppbgg13doi4k > CSeq: 3989 INVITE > Server: kamailio (4.1.5 (x86_64/linux)) > Content-Length: 0 > > What I understood about this problem is that the Proxy server is trying to > forward the INVITE to the Outbound server and this server does not know > that to do with this message - like forward it to alice. > > I'm stucked in this step. > > Here you can find the configuration files that I'm using: > > Outbound Server: > https://mega.co.nz/#!Wlw2RL7S!G2YOP-1QbSCeNQddaAe3kyQMqz8yteZlZsAUpjZX4v4 > > Proxy Server: > https://mega.co.nz/#!qxAXTbbb!4zBjRtmADdl3rPCN2Yidd3xnZEjFahW3GSMb00w41xc > > > > Thanks again! > > Bruno Emer > > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2014-09-18 18:04 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > > Ok, Daniel! >> >> I am trying exactly what you said, but for some reason the proxy server >> can't forward the message back to the WebSocket server. I am getting >> "487 Unresolvable destination". Appears that my Proxy Server can't get data >> from usrloc. >> >> When I register an user using WebRTC I can see that the system stores the >> information in kamailio database and in the field "contact" it puts a >> string. The Proxy server is looking in database and just find this >> information. Actually, I'm trying to understand the entire process that >> envolves passing information from an Edge Server to a Proxy server that >> sends the Register information to another Registrar server (as I described >> before). The database is shared between all the servers, and they are in >> the same network. >> >> I just read the information in another topic, as you said, but I think >> that my case is a lil bit different from it. I you agree, I can post my >> configuration files here. Maybe this can help another users with the same >> problem! >> >> Thanks again!!! >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> <https://www.facebook.com/bruno.emer.5> >> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >> <https://twitter.com/brunoemer_> >> <https://www.flickr.com/photos/122070309@N03/> >> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >> >> 2014-09-17 9:13 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >> Hello! >>> >>> I am relatively new to Kamailio and I'm trying to create a new >>> enviroment using it in my company. I am thinking about use Amazon to host >>> the servers and use OpsWorks to automatically escalate then if necessary. >>> To accomplish this, my idea is to separate the servers, using one dedicated >>> server to run as WebSocket, one to run as proxy and one as a Registrar. >>> I'll be using just one database to store informations to all of my servers. >>> With this, if I need more resources later, I can just create new servers >>> with the specific roles (WebSockets, Proxy, Registrar). >>> >>> By now, the idea is clear, but the point is that I don't know how to >>> separate the WebSockets server from the proxy server. Actually, I can do >>> this, but when I have one agent using a regular softphone and one agent >>> using WebSockets (with JSSIP) they are not able to establish a session if >>> the softphone user starts it. Now, I want to know if is there a way to use >>> two websockets servers, register users using both of then and start >>> sessions between then, with a separated proxy and registrar. >>> >>> Has anyone done this before? Is possible to use kamailio like this? >>> >>> >>> Thanks. >>> >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >> >> >

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Group call
by marino.mileti＠alice.it 24 Sep '14

24 Sep '14

Hi guys, is possible to make a group call with kamailio? Example...i dial 1000 and kamailio make a parallel forking to 1001,1002,1003? In case of yes...every group member (1001,1002,1003) will be resolved by "location"? Because some member is natted so RTPPROXY should be engaged Is there any examples? BRMarino

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Re: [SR-Users] Sip Event Filter
by Raúl Alexis Betancor Santana 24 Sep '14

24 Sep '14

Hi, you are facing the problem the wrong way. You could do that easy on the UAC side. For example, with Asterisk as UAC, you only need to pass a 10s timeout as paran to the dial app. A sip-proxy its not for what you are trying to acomplish. What you are trying to do it's a UAC or B2BUA task. Best regards. El 24/09/2014 09:00, Delifisek Tux <delifisek.tux(a)gmail.com> escribió: > > Hello, > > Very newbie here. Even I may not know what I'm talking about. Please be patience, thank you :) > > I had auto dialer and I have problem with unreachable numbers. > > So I wish if I call busy, unavailable, ringing but not answering numbers, system will hangup before the voice mail active. > > After some googling I found a sip proxy may solve my problem. > > I setup kamailio and manage to work with an example edge proxy setup. > > Then what ? > > My wish is, after the number dialed, start to filter every sip message, then let say if I do not get 183 session in progress in 10 seconds, or get a 10 trying in 10 seconds hangup line. > > Which module should I look after. > > My Best Regards. > >

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Sip Event Filter
by Delifisek Tux 24 Sep '14

24 Sep '14

Hello, Very newbie here. Even I may not know what I'm talking about. Please be patience, thank you :) I had auto dialer and I have problem with unreachable numbers. So I wish if I call busy, unavailable, ringing but not answering numbers, system will hangup before the voice mail active. After some googling I found a sip proxy may solve my problem. I setup kamailio and manage to work with an example edge proxy setup. Then what ? My wish is, after the number dialed, start to filter every sip message, then let say if I do not get 183 session in progress in 10 seconds, or get a 10 trying in 10 seconds hangup line. Which module should I look after. My Best Regards.

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upgrade question to Kamailio 4, how to handle ruid in location/aliases
by Klaus Darilion 24 Sep '14

24 Sep '14

Hi! I have to upgrade a production server from 3.x to 4.x. 4.x added a unique constraint on ruid. I have plenty of entries in aliases and usrloc table. How do I update these entries to get the ruid field populated? Can I do an upgrade reusing the entries stored in the usrloc table or do I have to truncate the table and then wait until every client reREGISTERs? Thanks Klaus

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R: Re: RTPPROXY & BRANCH
by marino.mileti＠alice.it 24 Sep '14

24 Sep '14

Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) So at the moment this scenario is not possible? Is there any possible turnaround? ----Messaggio originale---- Da: rfuchs(a)sipwise.com Data: 24-set-2014 1.15 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] RTPPROXY &amp; BRANCH On 23/09/14 04:35 AM, marino.mileti(a)alice.it wrote: > I've a problem with rtpproxy during a parallel ring scenario. > I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I > try to call them in parallel mode (ringall) rtpproxy module sends in to > the INVITE the same RTP ports. > > Is it possible to manage rtpproxy in order to generate a couple of new > port for each branch? Not at the moment. The logic is that one endpoint advertised by the client is mapped to one endpoint advertised by rtpengine. Why would you like a separate set of ports? cheers _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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RTPPROXY & BRANCH
by marino.mileti＠alice.it 23 Sep '14

23 Sep '14

Hi I've a problem with rtpproxy during a parallel ring scenario. I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I try to call them in parallel mode (ringall) rtpproxy module sends in to the INVITE the same RTP ports. Is it possible to manage rtpproxy in order to generate a couple of new port for each branch? My rtpproxy string in the kamailio conf (in manage branch section): rtpproxy_manage("bFAIE","192.168.10.10"); Brief extract of 2 SIP INVITE message send by kamailio to natted clients: 192.168.10.10 --> 192.168.10.20 User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:1001@192.168.10.20:6090 SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1004 1476 1657 IN IP4 192.168.10.10 Session Name (s): Talk Connection Information (c): IN IP4 192.168.10.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 62910 RTP/AVP 111 110 0 8 101 Media Description, name and address (m): video 47160 RTP/AVP 102 Media Attribute (a): nortpproxy:yes 192.168.10.10 --> 192.168.10.50 User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:1002@192.168.10.50:6090 SIP/2.0 Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1004 1476 1657 IN IP4 192.168.10.10 Session Name (s): Talk Connection Information (c): IN IP4 192.168.10.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 62910 RTP/AVP 111 110 0 8 101 Media Description, name and address (m): video 47160 RTP/AVP 102 Media Attribute (a): nortpproxy:yes

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Wrong from/to header compose way in print_to and print_from
by dongwf 23 Sep '14

23 Sep '14

Hi kamailio dev: I am writing a module to implement some features, my version is 4.1.2, I am not sure if it is a known issue, I found: 1. When I get a INVITE request with from/to like from:"display"<aaa(a)sip.com>, then I call tm's new_dlg_uas with the sip req, now the dlg's rem_uri set as value: "display"<aaa(a)sip.com>; 2. After that, when I need use the dlg to send BYE with tm's req_within with the above saved dlg, the print_to function will check the rem_uri's first value if "<" or not, now my uri has display name, so the outgoing BYE's to header becomes to:<"display"<aaa(a)sip.com>>, is it wrong? Should we check the last char in the rem_uri as ">"? The detail code is : static inline char* print_to(char* w, dlg_t* dialog, struct cell* t) { t->to.s = w; t->to.len = TO_LEN + dialog->rem_uri.len + CRLF_LEN + ((dialog->rem_uri.s[0]!='<')?2:0); memapp(w, TO, TO_LEN); if(dialog->rem_uri.s[0]!='<') memapp(w, "<", 1); memapp(w, dialog->rem_uri.s, dialog->rem_uri.len); if(dialog->rem_uri.s[0]!='<') memapp(w, ">", 1); ....

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Kamailio fronting Freeswitch authentication
by Ashwin Jain 23 Sep '14

23 Sep '14

Hi all, I am using Kamailio to front my freeswitch servers and it is working great. I am getting some problems with authentication, sometimes users which are not registered with kamailio are also able to make calls. I have a feeling that I am doing something really silly with kamailio routing. Following is the AUTH route that I am using, please let me know I am missing something very obvious. route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } I have set "WITH_AUTH" and "WITH_FREESWITCH". Let me know, if any more details are required. System configuration: Ubuntu: 12.04 LTS Kamailio: 4.1.5 (x86_64/linux) Freeswitch: 1.4.7 stable release -- Thanks and Regards, Ashwin Jain

2 1

Siremis v4.1.0 Questions.
by Mahmoud Ramadan Ali 23 Sep '14

23 Sep '14

Hello everyone, Please i want more clarification about what i can do with the Siremis v4.1.0 Can i configure Kamailio to load balance between multiple Asterisk servers and RTP proxy using the Siresmis web interface or i have to interface with the Kamailio configuration file and do that manually...? Thanks in advance.

4 6

User definable Ring Duration before forwarding
by Tim Chubb 23 Sep '14

23 Sep '14

Hi I was wondering if the following scenario is possible and if it is how I would go about implementing it: I would like to be able read from a database a list of destinations associated with a given dialled number which are tried in order of priority with each destination having a user definable ring duration before the next destination is tried. i.e. A Call is received from the PSTN, the number the caller dialled is looked up, it has 3 destinations attached to it, the users deskphone which is configured to ring for 10 seconds before the call is forwarded to their Mobile phone with a ring duration of 15 seconds, which if unanswered would be redirected to the main switchboard IVR. I think im looking for the equivalent of the Asterisk Dial app, where you can specify a timeout as an argument, is this possible to do with kamailio? All examples I have seen use a global timer value, which is unsuitable for this requirement, as each user could define n failover routes each with a distinct timeout before the next route is tried. Currently I have got as far as querying the database to retrieve the data but im a bit stumped as to how to handle the timeout variable. Any ideas? Tim Chubb

2 1

Kamailio - Separated Servers
by Bruno Emer 22 Sep '14

22 Sep '14

Hello! I am relatively new to Kamailio and I'm trying to create a new enviroment using it in my company. I am thinking about use Amazon to host the servers and use OpsWorks to automatically escalate then if necessary. To accomplish this, my idea is to separate the servers, using one dedicated server to run as WebSocket, one to run as proxy and one as a Registrar. I'll be using just one database to store informations to all of my servers. With this, if I need more resources later, I can just create new servers with the specific roles (WebSockets, Proxy, Registrar). By now, the idea is clear, but the point is that I don't know how to separate the WebSockets server from the proxy server. Actually, I can do this, but when I have one agent using a regular softphone and one agent using WebSockets (with JSSIP) they are not able to establish a session if the softphone user starts it. Now, I want to know if is there a way to use two websockets servers, register users using both of then and start sessions between then, with a separated proxy and registrar. Has anyone done this before? Is possible to use kamailio like this? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

3 4

R: Re: Kamailio multihomed - BYE problem
by marino.mileti＠alice.it 22 Sep '14

22 Sep '14

This is SIP capure. Kamailio listen on 172.20.11.172 & 192.168.10.10Caller is 172.20.11.208Called user that pickup call is 192.168.10.20Other group member is 172.20.11.152 Callee (172.20.11.208) make a call to group number 5000172.20.11.208 --> 172.20.11.172--------------Session Initiation Protocol (INVITE)Request-Line: INVITE sip:5000@172.20.11.172 SIP/2.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rportFrom: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrContact: <sip:1004@172.20.11.208:6090>;+sip.instance="<urn:uuid:a6e51976-8d02-4f51-a065-0f836be30038>" Kamailio replies with 100 Trying172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (100)Status-Line: SIP/2.0 100 trying -- your call is important to usVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrServer: kamailio (4.1.5 (i386/linux))Content-Length: 0 Kamailio lookup db and send 1st INVITE to 1st group member (1000(a)172.20.11.152)172.20.11.172 --> 172.20.11.152--------------Session Initiation Protocol (INVITE)Request-Line: INVITE sip:1000@172.20.11.152:5090 SIP/2.0Record-Route: <sip:172.20.11.172;lr=on;nat=yes>Via: SIP/2.0/UDP 172.20.11.172;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrMax-Forwards: 69Contact: <sip:1004@172.20.11.208:6090>;+sip.instance="<urn:uuid:a6e51976-8d02-4f51-a065-0f836be30038>" Kamailio sends 2nd INVITE to 2nd group member (1001(a)192.168.10.20) using another network interface192.168.10.10 --> 192.168.10.20--------------Session Initiation Protocol (INVITE)Request-Line: INVITE sip:1001@192.168.10.20:6090 SIP/2.0Record-Route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-Route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-Route: <sip:172.20.11.172;lr=on;nat=yes>Via: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrContact: <sip:1004@172.20.11.172:5060>;+sip.instance="<urn:uuid:a6e51976-8d02-4f51-a065-0f836be30038>" Kamailio receive 100 trying from 172.20.11.152172.20.11.152 --> 172.20.11.172--------------Session Initiation Protocol (100)Status-Line: SIP/2.0 100 TryingVia: SIP/2.0/UDP 172.20.11.172;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172Call-ID: q0KGluu3SrCSeq: 20 INVITE Kamailio receive 180 ringing from 172.20.11.152172.20.11.152 --> 172.20.11.172--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 172.20.11.172;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=puL7wn8Call-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio sends 180 ringing to caller 172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=puL7wn8Call-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio receive 100 trying from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Session Initiation Protocol (100)Status-Line: SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172Call-ID: q0KGluu3SrCSeq: 20 INVITE Kamailio receive 180 ringing from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio sends 180 ringing to caller 172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio receive 200 OK from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Session Initiation Protocol (200)Status-Line: SIP/2.0 200 OkVia: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITEContact: <sip:1001@192.168.10.20:6090>;+sip.instance="<urn:uuid:6382d45f-042d-4be8-b6bd-1621911e88a9>"Record-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio sends 200 OK to caller172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (200)Status-Line: SIP/2.0 200 OkVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITEContact: <sip:1001@192.168.10.20:6090>;+sip.instance="<urn:uuid:6382d45f-042d-4be8-b6bd-1621911e88a9>"Record-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio CANCEL other leg (messages removed...not so useful)-------------- Kamailio receive BYE from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Internet Protocol Version 4, Src: 192.168.10.20 (192.168.10.20), Dst: 192.168.10.10 (192.168.10.10)User Datagram Protocol, Src Port: 6090 (6090), Dst Port: sip (5060)Session Initiation Protocol (BYE)Request-Line: BYE sip:1004@172.20.11.172:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.10.20:6090;branch=z9hG4bK.20aa9gXIY;rportFrom: <sip:5000@172.20.11.172>;tag=-iPr1gnTo: <sip:1004@172.20.11.172>;tag=TNzDNgaNNCSeq: 111 BYECall-ID: q0KGluu3SrRoute: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio replies with 404 Not Here192.168.10.10 --> 192.168.10.20--------------Internet Protocol Version 4, Src: 192.168.10.10 (192.168.10.10), Dst: 192.168.10.20 (192.168.10.20)User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090)Session Initiation Protocol (404)Status-Line: SIP/2.0 404 Not hereMessage HeaderVia: SIP/2.0/UDP 192.168.10.20:6090;branch=z9hG4bK.20aa9gXIY;rport=6090From: <sip:5000@172.20.11.172>;tag=-iPr1gnTo: <sip:1004@172.20.11.172>;tag=TNzDNgaNNCSeq: 111 BYECall-ID: q0KGluu3SrServer: kamailio (4.1.5 (i386/linux))Content-Length: 0 ----Messaggio originale---- Da: davy.van.de.moere(a)gmail.com Data: 22-set-2014 13.52 A: "marino.mileti(a)alice.it"<marino.mileti(a)alice.it>, "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] Kamailio multihomed - BYE problem Could you show us some SIP headers? 2014-09-22 13:50 GMT+02:00 davy van de moere <davy.van.de.moere(a)gmail.com>: A very long shot, look into handle_ruri_alias(). 2014-09-22 12:58 GMT+02:00 marino.mileti(a)alice.it <marino.mileti(a)alice.it>: Hi, I'm having a problem with routing of BYEs in my multi homed Kamailio. My setup is a Kamailio v4.1.5 with two NIC (172.20.11.0 & 192.168.10.0) and three phones. Two phones (1 for each LAN) are in a group, so using alias_db I can call them in parallel mode. Caller is on 172.20 network...When I answer from phone on 192.168.10.0, RTP is ok ... but on BYE Kamailio replies with 404 Not Here...and the RURI of BYE has the IP of Kamailio instead the IP of the callee.When I answer from phone on 172.20.11.0, RTP is still ok and also BYE is managed correctly from Kamailio and the RURI of BYE has the IP of the callee. How can i solve this issue?Many thanks _______________________________________________ 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

Kamailio multihomed - BYE problem
by marino.mileti＠alice.it 22 Sep '14

22 Sep '14

Hi, I'm having a problem with routing of BYEs in my multi homed Kamailio. My setup is a Kamailio v4.1.5 with two NIC (172.20.11.0 & 192.168.10.0) and three phones. Two phones (1 for each LAN) are in a group, so using alias_db I can call them in parallel mode. Caller is on 172.20 network...When I answer from phone on 192.168.10.0, RTP is ok ... but on BYE Kamailio replies with 404 Not Here...and the RURI of BYE has the IP of Kamailio instead the IP of the callee.When I answer from phone on 172.20.11.0, RTP is still ok and also BYE is managed correctly from Kamailio and the RURI of BYE has the IP of the callee. How can i solve this issue?Many thanks

2 2

Re: [SR-Users] [OT] Accounting Drouting Gateway Selection
by Dan Christian Bogos 22 Sep '14

22 Sep '14

Hi Spencer, We use a similar approach as yours and use 302 Redirects coming from Kamailio instead of proxying the call (in our case we do it via LCR module in Kamailio). In FreeSWITCH you can then manually route these redirects and get your CDR data out of it. Cheers, DanB On 22.09.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > On 18/09/14 17:34, Spencer Thomason wrote: >> >Hello all, >> >I have a need to determine which calls go to which gateway. The challenge is that CDRs are produced by a FreeSWITCH instance upstream of a provider side Kamailio proxy that handles route selection via drouting module. The topology is like this: >> > >> >customer proxy -> FS [1-N] (accounting) -> carrier proxy (route selection) >> > >> >Does anyone have any ideas how to make the FreeSWITCH instances aware of which gateway actually received the call?

1 0

Kamailio fronting Freeswitch authentication
by Ashwin Jain 22 Sep '14

22 Sep '14

Hi all, I am using Kamailio to front my freeswitch servers and it is working great. I am getting some problems with authentication, sometimes users which are not registered with kamailio are also able to make calls. I have a feeling that I am doing something really silly with kamailio routing. Following is the AUTH route that I am using, please let me know I am missing something very obvious. route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } I have set "WITH_AUTH" and "WITH_FREESWITCH". Let me know, if any more details are required. System configuration: Ubuntu: 12.04 LTS Kamailio: 4.1.5 (x86_64/linux) Freeswitch: 1.4.7 stable release -- Thanks and Regards, Ashwin Jain

1 0

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