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37519 discussions

Store and add custom variables for sip registrations
by Markus Bönke 01 Nov '16

01 Nov '16

Hallo all, we want to use kamailio as registrar server in front of freeswitch. When a user registers we need to lookup and store some user variables. To every INVITE that is received from a registered user we want to add the variables as x sip headers before we relay them to freeswitch. The question is what is the best way to implement this? We tried to store the variables in a htable on register and then lookup and attach the variables on every INVITE from there. Is this the correct approach, one issue we see is that htable is not writing new inserts directly to the database. I’ve also seen that there is a table location_atts in the kamailio db which looks like that it is intended to store custom key value pairs for registrations, but the seems to be no module which writes to that table, or is there one? Thanks for any help or suggestions Markus

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TSILO module use failure
by Igor Olhovskiy 31 Oct '16

31 Oct '16

Hi! I’m trying to implement scheme much like Federico Cabiddu is described here https://www.youtube.com/watch?v=4XIrR9bwUkM Done a bit modification to his script, cause it’s not fully correct on 4.4 version So, parts of config looks like …. route[REGISTRAR] { … if (!save("location")) { sl_reply_error(); } else { route(PUSHJOIN); } exit; …. } route[LOCATION] { if (!lookup("location")) { …… if (is_method("INVITE")) { send_reply("100", "Trying"); route(SUSPEND); } } else { if (is_method("INVITE")) { if (t_newtran()) { ts_store("sip:$tU@$fd"); } $sht(vtp=>stored::$rU) = 1; } } route(RELAY); } ….. route[SUSPEND] { if(!t_suspend()) { send_reply("501", "Unknown destination"); exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); } # append branches or resume the transaction route[PUSHJOIN] { … #if was suspended - recover, if not - ts_append ….. ts_append(«location", "sip:$tU@$fd"); return; … t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } # lookup and relay after resuming transaction route[INVRESUME] { lookup("location"); t_relay(); ts_store("sip:$tU@$fd"); $sht(vtp=>stored::$rU) = 1; } Branches on ts_store are saved in format (sip:username@domain) and also in this format they are added on ts_append. All regarding to routes working ok, except on ts_append I got ERROR: tm [t_append_branches.c:168]: t_append_branches(): ERROR: t_append_branch: failure to add branches (-1) Also, it’s a mixed environment, like wss + udp + tcp. Any idea why? Thanks. -- Best regards, Igor

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Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 30 Oct '16

30 Oct '16

Hi all, I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great. Thanks, Serhat On 24 October 2016 at 21:00, Serhat Guler <srtguler(a)gmail.com> wrote: > Hi Daniel, > > I am using only record_route() without any parameters. I do not have a > proper computer atm to draw the network diagram, but I can tell you shortly > about the network setup. > > I have only enabled websockets for the pcscf to allow ws and wss > connections. In that case there is a ws connection that uses UDP protocol. > This is the ACK to complete the session setup. > > the sipml5 client is configured as follows: > WebSocket Server URL: ws://192.168.0.11:880 > SIP outbound Proxy URL: udp://192.168.0.11:4060 > > Mercuro IMS client: uses UDP port as well: 4060 > > The call is made from sipml5 client. The Mercuro phone rings, and when I > reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from > sipml5 doesn't pass the PCSCF as I explained in the previous message. > > A part of PCSCF cfg file: > > # Check for Subsequent requests: > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if ($route_uri =~ "sip:mo@.*") { > setflag(FLT_MO); > } > if(!isdsturiset()) { > handle_ruri_alias(); > } > # RTP-Relay, if necessary > route(RTPPROXY); > t_relay(); > } else { > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > t_relay(); > exit; > } else { > xlog("L_INFO", "ACK without matching transaction ... > ignore and discard!!!!!\n"); > # ACK without matching transaction ... ignore and > discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > > Cheers, > Serhat > > > > On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> I haven't noticed the log files, it's ok. >> >> From the Route header, I see that there is a proxy that uses WS: >> >> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >> 1nCMEI1mR0RztrB;did=e82.0c3> >> That is the address of the next hop and typically a proxy doesn't use >> websocket connection to another proxy. Can you show a diagram with the sip >> server nodes in your network and what protocols are used between them? >> >> Are you simply use record_route() function, or some other function or >> different parameters to it? >> >> Cheers, >> Daniel >> >> >> On 24/10/16 12:18, Serhat Guler wrote: >> >> Hi Daniel, >> >> Thanks for your reply. I actually attached a log file with debug level 3, >> consisting ACK related messages. If you would like to see more logs, I'll >> send a new log file in the evening. >> >> Cheers, >> Serhat >> >> On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> can you get all the log messages for ACK but with debug=3 in the >>> kamailio.cfg? >>> >>> Cheers, >>> Daniel >>> >>> On 23/10/16 22:04, Serhat Guler wrote: >>> >>> Hello, >>> >>> I finally managed to place a call from sipml5 webrtc client to Mercuro >>> IMS client. The phone rings, and when I answer it sends 200 OK to the >>> sipml5 where as sipml5 send back an ACK message which never passes the >>> originating PCSCF. The PCSCF says: >>> >>> 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS >>> connection (id: 0) for WebSocket could not be found >>> 8(3640) ERROR: <core> [msg_translator.c:1947]: >>> build_req_buf_from_sip_req(): could not create Via header >>> 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building >>> failed >>> >>> I doubt that the WebSocket connection is closed, cause when I terminate >>> the call from Mercuro client a bye request is being sent to the sipml5. >>> >>> The ACK package: >>> >>> ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. >>> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 >>> hG4bKvuly7bmxnN4aqM4zZTIS;rport >>> From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB >>> To: <sip:alice@net1.test>;tag=18823 >>> Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c >>> all=no;transport=ws>;+g.oma.sip-im;language="en,fr" >>> Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 >>> CSeq: 3887 ACK >>> Content-Length: >>> Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> >>> Max-Forwards: 69 >>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >>> 1nCMEI1mR0RztrB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz >>> trB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.1c3> >>> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 >>> Organization: Doubango Telecom >>> >>> Have been thinking for quite a while, but couldn't really find a reason >>> why it wouldn't add the v,a header. A debug 3 level log file is also >>> attached. >>> >>> Thanks in advance, >>> Serhat >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >>> >>> _______________________________________________ SIP Express Router >>> (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org http://lists.sip-router.org/cg >>> i-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >> >> >

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digest authentication based on two password columns
by Jayesh Nambiar 28 Oct '16

28 Oct '16

Hello, I was wondering if it is possible to do a failover in digest authentication. For eg: I get two possible valid password values. Can I do a pv_proxy_authenticate in such a way where if authentication fails, I do a pv_proxy_autenticate again on the different value to check if that succeeds? Is it even allowed in digest authentication mechanism. Just getting opinions. Thanks for any help. - Jayesh

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Re: [SR-Users] fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Daniel, No that exactly the one, Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 28 October, 2016 07:37:31 Subject: Re: [SR-Users] fetch contact Hello, the contact address is one for attributes in $ulc(...), iirc, it is the =>addr -- see the wiki for pv cookbook for more details. Or are you looking for something else? Cheers, Daniel On 27/10/16 21:43, Slava Bendersky wrote: Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. 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 -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ 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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Errors when parsing Diversion header with multiple comma-separated diversions
by Björn Bylander 28 Oct '16

28 Oct '16

Hello! Given a Diversion header like the following: Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Kamailio 4.4.3 (and probably earlier versions) emit error messages complaining about the comma character. According to RFC 6044 multiple comma-separated diversions in a single Diversion-header is allowed. Full INVITE as an example: INVITE sip:+1234567890@127.0.0.100:5060;user=phone;transport=tcp SIP/2.0 Via:SIP/2.0/TCP 127.0.0.101;branch=z9hG4bKBroadWorks.-1od4eg8-127.0.0.100V5060-0-82844591-13 54681939-1477634439005- From:<sip:+9876543210@127.0.0.101;user=phone>;tag=1354681939-1477634439005- To:<sip:+1234567890@127.0.0.100:5060;user=phone> Call-ID:BW0800390052810161562691307@127.0.0.101 CSeq:82844591 INVITE Contact:<sip:127.0.0.101:5060;transport=tcp> P-Asserted-Identity:<sip:+9876543210@127.0.0.101;user=phone> Privacy:none Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE X-BroadWorks-DNC:network-address="sip:+9876543210@127.0.0.101;user=phone" Accept:application/dtmf-relay,application/media_control+xml,application/sdp, application/x-broadworks-call-center+xml,multipart/mixed Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:267 v=0 o=BroadWorks 100442303 1 IN IP4 127.0.0.101 s=- c=IN IP4 127.0.0.102 t=0 0 m=audio 29778 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 Example log messages: Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:488]: parse_to_param(): ERROR parse_to_param : invalid character ',' in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered>> Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:254]: parse_to_param(): ERROR: parse_to_param :unexpected char ["] in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered,>> . Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:906]: parse_addr_spec(): ERROR: parse_to: invalid To - unexpected end of header in state 21 Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_diversion.c:69]: parse_diversion_header(): ERROR:parse_diversion_header: bad diversion header Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:345]: lval_pvar_assign(): non existing right pvar Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:405]: lval_assign(): assignment failed at pos: (562,21-562,35) The code on line 562 is $avp(div_user) = $(di{uri.user});. Can this be handled in any way by Kamailio without changing the parser? Best regards, Björn Bylander

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fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks

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logfile is empty
by Hermann Norpois 27 Oct '16

27 Oct '16

my logfile /var/log/kamailio/kamailio.log is empty. 1) /etc/rsyslog.d/kamailio.cfg exists with the entry: log_facility=LOG_LOCAL0 2) /etc/rsyslog.con has the entry: # # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log # local0.* -/var/log/kamailio/kamailio/kamailio.log Alternatively I edited /etc/rsyslog.d/kamailio.conf that should be used by rsylog.conf 3) /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log I am working with ubuntu on an amazon instance. Why is the logfile empty? Thanks Herman

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How do I see the full SDP?
by Martin Taschen 27 Oct '16

27 Oct '16

Hello, kamailio seems to work fine. After registration I would like to see the complete SDP of an user. . How is this done? Thanks Martin Taschen

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Re: [SR-Users] sr-users Digest, Vol 137, Issue 25
by surendra tiwari 27 Oct '16

27 Oct '16

i used the cnxcc module its successfully terminated the call. but if you want i can help you. On Thu, Oct 27, 2016 at 12:58 AM, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: cnxcc doesn't terminate calls (Igor Potjevlesch) > 2. logfile is empty (Hermann Norpois) > 3. Re: Out of memory in UB 210: OOM killed process 12261 > (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB > (Jurijs Ivolga) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2016 12:48:29 +0200 > From: "Igor Potjevlesch" <igor.potjevlesch(a)gmail.com> > To: "'Kamailio $SER$ - Users Mailing List'" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] cnxcc doesn't terminate calls > Message-ID: <009c01d22f76$7d0c9a30$7725ce90$(a)gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > > > Yep, the dialog is created. I put the dialog into DB and see it. > > Nothing appears in the logs when the timer is reached. > > > > I succeed to do the same behaviour with dialog module only and > dlg_set_timeout. Looks to do why I want. > > > > I'm a bit frustrating about cnxcc because I really don't understand why > nothing happen after max_time, but, anyway, it works with dialog only. > > > > Regards, > > > > Igor. > > > > De : sr-users [mailto:sr-users-bounces@lists.sip-router.org] De la part > de Arsen > Envoyé : mercredi 26 octobre 2016 11:15 > À : Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> > Objet : Re: [SR-Users] cnxcc doesn't terminate calls > > > > Hi Igor, > > > > Make sure that you create a dialog for this call and $dlg_var(key) is > available (This pseudo-variable will be available only for subsequential > requests after doing loose_route().) > > > > Also check kamailio logs maybe you try to terminate call for wrong > customer id. > > > > On Wed, Oct 26, 2016 at 10:52 AM, Igor Potjevlesch < > igor.potjevlesch(a)gmail.com <mailto:igor.potjevlesch@gmail.com> > wrote: > > Hi, > > > > I'm testing cnxcc module in order to make a simple call duration > limitation. > > My understanding is to use the function like this : > > > > $dlg_var(customer_id) = "customer-" + $fU; > > $var(max_time) = 5; > > > > if (!cnxcc_set_max_time("$dlg_var(customer_id)", > "$var(max_time)")) > > > > Unfortunately, after 5 seconds, nothing happen. I don't even see an error > from dialog module. > > > > Anyone has a better understanding of the way the module runs? Thank you! > > > > Regards, > > > > Igor. > > > _______________________________________________ > 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 > > > > > > > > -- > > Regards, > Arsen. > >

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