- sr-users - kamailio.org

Possible conflict between t_newtran and setflag for ACC
by Patrick Wakano 01 Dec '19

01 Dec '19

Hello list, Hope you all doing well! I am using the ACC module and using the setflag() function as done in several examples. It works fine. However, I've added the t_newtran() function almost in the begging of the INVITE handler to help the retransmission detection and after that I noticed the ACC was not saving anything in DB. So after debugging I discovered that if I call the t_newtran() before setting the ACC flags, the module will not save the calls in DB, but if I call it after setting … [View More]

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tcp limit, too many open files
by David Villasmil 01 Dec '19

01 Dec '19

Hello guys, I'm facing an issue where, if i start kamailio as root, all works properly, but if i start it as kamailio, i get hundreds of: handle_new_connect(): error while accepting connection(24): Too many open files My limits is like: kamailio soft core unlimited kamailio soft data unlimited kamailio soft fsize unlimited kamailio soft memlock unlimited kamailio soft nofile unlimited kamailio … [View More]

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Kamailio as SBC for Microsoft Teams
by Henning Westerholt 29 Nov '19

29 Nov '19

Hello, I have created a "How-To" blog post on using Kamailio as "session border controller" for Microsoft Teams Direct Routing: https://skalatan.de/de/blog/kamailio-sbc-teams You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

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Adding "q" value for the main branch
by Maxim M 28 Nov '19

28 Nov '19

Maxim Malygin <maxim.malygin(a)gmail.com> 17:08 (5 минут назад) кому: sr-users Hello, I need to form list of Contact headers with "q" parameters for "302 Moved Temporary". I do something like this: ... # $xavp(contacts) - contains list of contact URIs (uri) and Q values (q) $var(i) = 0; $var(num) = $cnt($xavp(contacts)); while($var(i) < $var(num)) { if($var(i) == 0) { # How to add q value to this branch? $ru = $xavp(contacts[$var(i)]=>uri); } else { … [View More]

1 0

Get totally rid of H.264 SDP
by Kjeld Flarup 28 Nov '19

28 Nov '19

Hi I have a door telephone, with GSM fallback. Since telephone operators are going away from SS7, more and more traffic goes SIP. Thus we now have some numbers where we can see that doing an invite with H.264 causes them to fail. I have tried to remove coded 96 with sdp_remove_codecs_by_id, but it still leaves most of the video SDP in the invite. SDP received by kamailio Media Description, name and address (m): video 6100 RTP/AVP 96 Media Attribute (a): rtcp:6101 IN … [View More]

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Kamailio does not detect message
by Adesh Pandey 28 Nov '19

28 Nov '19

Hi, I have below scenario: User registers on Kamailio server, calls to a number which is forwarded to asterisk through dispatcher, asterisk server answers and sends a text message to the caller using sendText but first message is delivered to client with message "501 Not Implemented" but in the subsequent calls asterisk shows that the text was delivered but kamailio did not detect the message. Guys, any help would be appreciated, I am using the below mentioned config. https://github.com/… [View More]

1 0

Kamailio developers meeting - follow up remarks
by Daniel-Constantin Mierla 27 Nov '19

27 Nov '19

Hello, last week we had the 2nd annual Kamailio Developers Meeting hosted by Sipgate in Dusseldorf, Germany: * https://www.kamailio.org/w/developers-meeting/ 16 people were at the event in various roles. Giacomo Vacca published on his blog a good summary of what happened there: * https://www.giacomovacca.com/2019/11/my-notes-on-kamailio-developer-meeting… This year we had a lot of discussions, as well as work done on multiple planes, not only Kamailio code. So I am trying … [View More]to list here some of the conclusions for future development, the technical aspects of the meeting, so everyone is aware and can provide feedback. 1) Effective work was done on: * kamailio code * kamailio rpm packaging * kamailio tools (kamctl) * kamailio release process * kamailio project keys (to be used to sign the packages) 2) Documentation 2.a) Wiki * it was somehow a rough consensus to move the wiki content to github, along with changing the format from dokuwiki markdown to the standard/github markdown. This should enable people to make pull requests so developers or community members can review and aprove new content. It also makes it easier to contribute using existing github account, now the kamailio.org/wiki is requiring to make a dedicated account, which many prefer not to do it. * the presentation can be done either by using mkdocs to generate html files hosted on kamailio.org or using the github provided wiki portal. 2.b) Docs for variables and transformations * there was a proposal to move them in the documentation the modules that export them, there are pros and cons, needs more discussions. Now they are in the wiki, so this probably has to resume after deciding on 2.a). 3) Kamailio Modules 3.a) replication (dmq) - several participants discussed about negotiations between nodes to take active role on some cases (e.g., active dialogs) 3.b) api integration - quite some interest in JSON-based API routing, concluding in extending rtjson to cover more use cases 3.c) security - have options to restrict the use of TLS1.3 or newer 4) Kamailio Releases * v5.3.2 was released during the event, allowing to document the process * work to automatize the process is planned, then eventually assing teams for takeing cares of releases from specific branches 5) Kamailio Testing * existing docker-based testing framework should be extended and integrated in CD/CI pipeline 6) Kamailio packages 6.a) rpms * rpm.kamailio.org has been prepared and is expected to take over the opensuse build service for building rpms and hosting them. Expected to provide support for hosting many kamailio versions in the same release series so one can do downgrade to older releases. Also, there is work in progress to provide nightly builds. 6.b) debs * work is planned to offer many kamailio versions in the same release series 7) Kamailio tools * kamctl/kamdbctl should be obsoleted in favor of kamcli, which offers a better framework for input validation and output formatting, as well as better portability, no longer depending on shell interpreter 8) Various discussions * kemi exports from C point of view and how to combine the documentation for modules and their kemi exports * how to make kamaiio friendlier in virtualized environments (ended up in the need of making the use of advertised address a bit more dynamic) * project organizatoric topics - to be approached separately * next events - Fosdem - someone should submit a proposal to present about Kamailio 9) Long term goals We speak here more or less about Kamailio 6.0 ... * change the behaviour of the native config interpreter to be consistent with the other programming languages in terms of handling the response code (change what is now: the evaluation of negative value to false and positive value to true and the hidden return 0 to exit) * make the pool of processes more generic, so they can handle traffic from more sockets (being sip traffic or something else) -- this should make better use of resources, as some sockets might be less busy that others I hope I covered the important topics, if I remember something else, I will reply on this thread. Or maybe other participants can contribute missing topics. Should anyone have comments or suggestions on the above topics, or new ones, let's discuss on sr-users because it impacts the long term use of Kamailio (sr-dev is cc-ed now for notification purposes). Overall, there were 2 very intensive days, however in a friendly and relaxed environment offered by Sipgate. Extremely useful discussions, not only about Kamailio, but about RTC ecosystem. At the evening social network event event, a couple of external people joined, some of them presenting very interesting use cases of Kamailio. Many thanks to all participants that allocated time and resources to come to Dusseldorf, as well as to the companies that covered expenses for participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com [View Less]

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Kamailio xavp_contact - Segmentation Fault
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, Kamailio is crashing when i'm trying to set the parameter; modparam("tm|usrloc", "xavp_contact", "ulattrs") That's happening with Kamailio 5.3.1 and 5.2.5 too. Crash is happening when Im register 2 devices with the same extension. This is the core dump: [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /usr/local/etc/ka'. Program terminated … [View More]with signal 11, Segmentation fault. #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 288 core/xavp.c: No such file or directory. (gdb) bt #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 #1 0x0000000000581869 in xavp_insert (xavp=0x0, idx=0, list=0x0) at core/xavp.c:752 #2 0x00007f05be0ee3a1 in ki_t_next_contacts (msg=0x7f05cb76c218) at t_serial.c:552 #3 0x00007f05be0f09e8 in t_next_contacts (msg=0x7f05cb76c218, key=0x0, value=0x0) at t_serial.c:756 #4 0x0000000000434d41 in do_action (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1067 #5 0x00000000004418e8 in run_actions (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1572 #6 0x0000000000441fa9 in run_actions_safe (h=0x7ffe93c9de90, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1636 #7 0x000000000065734e in rval_get_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, i=0x7ffe93c9cf78, rv=0x7f05cb642570, cache=0x0) at core/rvalue.c:912 #8 0x000000000065b8fe in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9cf78, rve=0x7f05cb642568) at core/rvalue.c:1910 #9 0x000000000065bd51 in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9d42c, rve=0x7f05cb642c98) at core/rvalue.c:1918 #10 0x0000000000434807 in do_action (h=0x7ffe93c9de90, a=0x7f05cb645028, msg=0x7f05cb76c218) at core/action.c:1043 #11 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb637e48, msg=0x7f05cb76c218) at core/action.c:1572 #12 0x0000000000431767 in do_action (h=0x7ffe93c9de90, a=0x7f05cb583858, msg=0x7f05cb76c218) at core/action.c:691 #13 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb574750, msg=0x7f05cb76c218) at core/action.c:1572 #14 0x0000000000442071 in run_top_route (a=0x7f05cb574750, msg=0x7f05cb76c218, c=0x0) at core/action.c:1657 #15 0x00000000005874a9 in receive_msg (buf=0xa6ec00 <buf.6868> "OPTIONS sip:10@192.168.0.231:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.231;branch=z9hG4bKd8a9.1b3abe656532065414a2f35a8916008d.1\r\nVia: SIP/2.0/UDP 192.168.0.131:5060;received=192.168.0.131;branch=z9hG"..., len=694, rcv_info=0x7ffe93c9e4d0) at core/receive.c:341 #16 0x000000000047bf81 in udp_rcv_loop () at core/udp_server.c:541 #17 0x0000000000424f9d in main_loop () at main.c:1669 #18 0x000000000042c688 in main (argc=13, argv=0x7ffe93c9ea18) at main.c:2710 Thank you. [View Less]

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Kamailio Parrallel Forking
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, I'm trying to use parallel forking if one extension is registered from multiple devices. (using WebSockets) This is the related code from kamailio.cfg: ------------------------------------------------------------------------------------------------------------------------------------ if (!t_load_contacts()) { xlogl("L_WARN", "Error loading contacts for $rU\n"); sl_send_reply("500", "Server Internal Error (Code:$cfg(line))"); exit; } else { xlogl("L_INFO", "… [View More]

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Kamailio 5.2.3 segfaulting
by Daniel Greenwald 27 Nov '19

27 Nov '19

I'm seeing this in the log and kamailio is crashing every hour or so: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 33/var/log/kamailio/kamailio.log Any ideas? Should I emergency upgrade to 5.3.1?

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No Audio using RTP Proxy
by Sujit Roy 27 Nov '19

27 Nov '19

Hello I am facing a problem as below. Please suggest for the work around. My call flow is like this. SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy So when the call arrives at Kamalio+RTPProxy, i m getting below in log. Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@ 192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong Nov 26 23:25:31 rtpproxy[18508]: INFO:… [View More]

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Encode host ip in Contact header for semi-stateless setup
by Daniel Greenwald 27 Nov '19

27 Nov '19

At kamailio world 2017, Jan Lorenz of Pascom explained how they use kamailio in a semi-stateless manner and encode the asterisk's IP into the contact header on the way out so that it can be decoded and used to route properly on the way back without Record-Routing. I would like to create such a setup. I'm looking for pointers if there is a particular module/method that would be ideal to to use for this. Any other gotcha's to consider in such a setup. My plan is to keep the transaction on the … [View More]

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dialog module: Does dlg_manage() have to be called for all requests?
by George Diamantopoulos 27 Nov '19

27 Nov '19

Hello all, I've been reading through the dialog module docs again, and there seems to be a discrepancy in what's suggested in the docs. In the intro, it is stated that 'To create the dialog associated with an initial INVITE request, execute the function “dlg_manage()"'. Later on, in section "7.10" the following example code is provided: request_route { ... if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = "DLGTIMEOUT"; $dlg_ctx(timeout_bye) = 1;… [View More]

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Modify SIP Message Header and Message Body parameter
by Sujit Roy 26 Nov '19

26 Nov '19

Hello Please see below SIP Message SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.1681.101:5060;branch=z9hG4bK5a61798d7c850d39;rport=5060 From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567@ 192.1681.102> Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE *Server: kamailio (5.1.9 (x86_64/linux))* Content-Length: 0 *1. I want to Change the Server Parameter . How to do it ? * SIP/2.0 183 Session Progress Via: … [View More]

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Beginner Help: Relay vs Forward
by Michael Iedema 26 Nov '19

26 Nov '19

Hi all, I am trying to place Kamailio between several SIP services which are differentiated by port and by the type of message they handle. To be concrete, here are the services: Originator: - reachable on port 5060 - originates all traffic to the services Service A: - reachable on port 5061 - handles REGISTER Service B: - reachable on port 5062 - handles INVITE Service C: - reachable on port 5063 - handles MESSAGE The services, originator and Kamailio instance may all be run on … [View More]the same IP or may be split onto different addresses. Previously, the originator needed to know the address and port of each service. Now I want to make Kamailio responsible for that task so it can be done more intelligently. To start though I simply want to get traffic flowing between the services and this is where I’m struggling to climb the learning curve. In my test setup the originator and all services are on the same IP with only Kamailio running on a different IP. Address 1 Address 2 ======================== Originator <--> Kamailio:6060 | | Service A <-------> | Service B <-------> | Service C <-------> | My first attempt at the logic looks like this: # HOST DEFINITIONS #!substdef "/HOST_ORIGINATOR/192.168.86.110/" #!substdef "/HOST_REGISTER/192.168.86.110/“ #!substdef "/HOST_INVITE/192.168.86.110/" #!substdef "/HOST_MESSAGE/192.168.86.110/" # PORT DEFINITIONS #!substdef “/PORT_ORIGINATOR/5060/" #!substdef "/PORT_REGISTER/5061/“ #!substdef "/PORT_INVITE/5062/" #!substdef "/PORT_MESSAGE/5063/" # ROUTE LOGIC route { # ORIGINATOR sent us something if ( $(ct{nameaddr.uri}{uri.port}) == HOST_ORIGINATOR ) { xlog("L_INFO", "ORIGINATOR: sent $rm\n”); if (is_method("REGISTER")) { $du = "sip: HOST_REGISTER:PORT_REGISTER"; xlog("L_INFO", "ORIGINATOR -> REGISTER: Destination URI is now $du\n"); t_relay(); exit; } } … #### END SNIPPET #### My very newbie questions are: - Am I even on the right track here? I understand that t_relay() establishes a stateful dialog handler between the originator and register services. - Can I accomplish this bi-directional path statelessly with forward()? - Do I need to add logic to handle the messages from the REGISTER service back to ORIGINATOR? This config currently fails because of "no corresponding socket found” when the relay triggers. I believe this is because I’m running the Kamailio instance in a Docker container so my first step today is to set up a native install. I also looked at the dispatcher module which seems very close to what I want to do. Maybe that’s a more natural way to go? Anyway, apologies for the lengthy explanation. I hope it is at least clear what I intend to do. I will gladly research and do trial and error to solve this but need to know I’m going down roughly the right path. I’m not sure what I should even be googling even of the time :-) Thanks as always for your time and the great software! Regards, -Michael [View Less]

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Re-use TCP connections on different transactions
by Joel Serrano 26 Nov '19

26 Nov '19

Hello, I'm setting up a Kamailio instance behind a TCP load balancer (with proxy protocol and NAT routing: meaning Kam stays in the flow all the time). I've managed to get working almost everything we need for our service, except for one thing, and that is for Kam to use existing connections for subsequent transactions: Following this example: [image: image.png] EXT & INT represent the external and internal interface of a LB between the UAC and Kamailio, using TLS on both legs and … [View More]proxy protocol. Transaction 1: INVITE, 100, 180, 183, 200 OK UAC 1.1.1.1:1111 -> 2.2.2.2:443 (EXT) 3.3.3.3:3333 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 2: ACK UAC 1.1.1.1:1112 -> 2.2.2.2:443 (EXT) 4.4.4.4:4444 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 3: BYE Kam 7.7.7.7:5060 -> 3.3.3.3:3333 (INT) 2.2.2.2:443 (EXT) -> 1.1.1.1:1111 UAC My problem is with Transaction 3. In this case the BYE is originated by the callee, and Kam has to send it to the caller. As the TCP load balancer is between Kam and the UAC, Kam has to send it to the LB so then the LB can forward it back to the UAC. This works well for msgs that belong to the same transaction (INVITE, 100, 180, 183, 200 OK) but it fails when they don't belong to the same transaction. Thanks to the newly added $tcp(c_si) and $tcp(c_sp) pseudovars, I can save the internal IP:Port of the LB, so I can send stuff later to it, my problem is that Kam doesn't seem to allow this? On the original INVITE, I use the following to save where I have to reach the UAC: add_contact_alias("$tcp(c_si)", "$tcp(c_sp)", "tls"); Then, handle_ruri_alias() will take care of setting $du to the correct (internal LB) IP:Port so I can reach the UAC, this works. My problem is that Kamailio doesn't identify that there is a valid existing TLS connection still up (from the INVITE), and tries to create a new one (and this obviously doesn't gives all sorts of problems). So when I run handle_ruri_alias(), and $du is set to 3.3.3.3:3333 (from the example above), instead of using the existing connection, Kamailio tries to create a new one. I have a log statement right before with the result of tcp_conid_state(1) (the connid is 1 for this connection) and the $rc is 1 (Connection is OK), but when I tell Kamailio it has to use it I get this in the logs: DEBUG: {1 11726467 BYE gqR1qqNK8B} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} And then the problems begin... I have tried playing around with: tcp_reuse_port tcp_connection_match But no luck..! I also thought it could be a problem of the connection being created on one worker, and a different worker handling BYE transaction, so tested with children=1 and tcp_children=1, but still same problem. A more detailed log: In blue my log statement checking for the status of conid "1", in red Kam not being able to find it, although it exists (as validated by tcp_conid_state(), and even in netstat I can see the connection established). In this log, 35.191.0.66:60271 would be the equivalent of 3.3.3.3:3333 and 104.175.176.242:28157 would be 1.1.1.1:1111 from the example above. ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1657]: _tcpconn_find(): found connection by id: 1"} "message":" NOTICE: {1 11727734 BYE 5-LX4GdI9X} <script>: JOEL TEST New request - M=BYE TCP STATUS:1 ID=5-LX4GdI9X"} ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1144]: handle_ruri_alias(): setting dst_uri to <sip:35.191.0.66:60271;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1166]: handle_ruri_alias(): rewriting r-uri to <sip:linphone@104.175.176.242:28157;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:1328]: t_newtran(): msg (0x7f3c884259d0) id=534\/18664 global id=532\/18664 T start=0xffffffffffffffff"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=63496, isACK=0"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK808f.eee2444f92a02cb33e1b7a21f20bc6bb.0]"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:675]: t_lookup_request(): no transaction found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_hooks.c:336]: run_reqin_callbacks_internal(): trans=0x7f3c83b8c598, callback type 1, id 0 entered"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/crypto\/md5utils.c:67]: MD5StringArray(): MD5 calculated: 3071029feb05962b26b53a9664a14210"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1095]: check_route_param(): route params checking against [;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes] (orig: [r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes])"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1101]: check_route_param(): params are <;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/msg_translator.c:161]: check_via_address(): (198.1.54.228, 198.1.54.228, 0)"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 35.191.0.66"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1242]: tcpconn_new(): on port 60271, type 3"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1561]: tcpconn_add(): hashes: 1446:2350:0, 5"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization"} ... So time to seek help from the community, any suggestions/ideas/comments? Sorry if all this sounds confusing, I've tried my best to put in text the whole scenario in and "understandable" way... Is this even doable? Thanks, Joel. [View Less]

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How to enforce REGISTER with AKAv1 (IMS)
by Moshe A 26 Nov '19

26 Nov '19

Hello, How can we enforce REGISTER with AKAv1? Here our setup 1. kamailio version 4.3.4 for IMS 2. Boghe IMS client v1.0.58.550 3. scscf.cfg is configure with: #!define REG_AUTH_DEFAULT_ALG "AKAv1-MD5" 4. Set Authentication Schemes in Private User Identity - IMPI for user bob to: Digest-AKAv1 (3GPP) results: First REGISTER fails with 401 Unauthorized Second REGISTER 200 OK, but the REGISTER is MD5 and not AKAv1 Attached is tcpdump. By the way: in which DB should we find Authentication … [View More]

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call not going to end point
by Gaurav Bmotra 26 Nov '19

26 Nov '19

hi i m using kamailio 5.4 dev Docker image when i try to call form calientA to clientB i got (Temporarily Unavailable (code: 480)) i m using freeswitch as media server so the flow and structure should be like this ClientA---->Kamailio------>FS---->Kamailio>---->ClientB but issue is (what i m getting ) ClientA---->Kamailio------>FS---->Kamailio>-------->FS-------->kamailio--------->FS IT CREATE THE CALL LOOP b/w KAMAILIO AND FREESWITCH *MY CONFIG FILE IS … [View More]ATACHED * in logs i m getting -------------------------------------------------------- Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 38902: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "18_4 Uc0,8,101 9857c00a-8947-1238-c6a7-12590b724429 3.82.96.70 22082 jKrKymKNg480p;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 9857c00a-8947-1238-c6a7-12590b724429, tag jKrKymKNg480p;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 35888: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "20_4 Uc0,8,101 98589959-8947-1238-c6a7-12590b724429 3.82.96.70 29846 m5a51aNvapN6D;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 98589959-8947-1238-c6a7-12590b724429, tag m5a51aNvapN6D;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 37706: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "13_4 Uc0,8,101 985962c1-8947-1238-c6a7-12590b724429 3.82.96.70 29432 pQXp50p3471BN;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985962c1-8947-1238-c6a7-12590b724429, tag pQXp50p3471BN;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 45078: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "16_5 Uc0,8,101 985a3ce3-8947-1238-c6a7-12590b724429 3.82.96.70 25096 r9F88praZSeHc;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985a3ce3-8947-1238-c6a7-12590b724429, tag r9F88praZSeHc;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 58376: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- *here is my config file * #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.4 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To use RTPEngine (instead of RTPProxy) for nat traversal execute: # - define WITH_RTPENGINE # - install RTPEngine: https://github.com/sipwise/rtpengine # - start RTPEngine: # rtpengine --listen-ng=127.0.0.1:2223 ... # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!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 Local Config If Exists ######### import_file "kamailio-local.cfg" ####### 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 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:LOCAL_IP:5091 advertise PUBLIC_IP:5091 /* 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 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### 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 = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN 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 */ # mpath="/usr/lib/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.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 "acc.so" loadmodule "counters.so" loadmodule "dispatcher.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" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!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 # ----------------- setting module-specific parameters --------------- #-------------------dispatcher params------------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- 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) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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;") #!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) modparam("debugger", "log_level_name", "exec") #!endif ####### 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 { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # 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); route(DISPATCH); # user location service } # Wrapper for relaying requests route[RELAY] { # 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; } Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); $rcv_addr="sip:"+$si+":"+$sp; if(starts_with("$rcv_addr","$du")) { route(LOCATION); } else { route(RELAY); } exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # 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 statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } 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); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!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; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!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; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal 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; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif 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(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!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 (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } 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 # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\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_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { 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")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } -- *Regards:* Gaurav Kumar [View Less]

2 2

How to get the audio codec used for each established call.
by Abdoul Osséni 25 Nov '19

25 Nov '19

Hello all, I try to save in the CDR the audio codec used for each established call. Can you help me? Regards Abdoul.I

3 4

SDP Modification
by Sujit Roy 25 Nov '19

25 Nov '19

Hello How can i change c= and o= in SDP using RTPProxy ? Please advise Thanks -- Regards =================== Sujit Roy

2 1

Passing media thru RTP Proxy
by Sujit Roy 25 Nov '19

25 Nov '19

Hello I have installed and configured both Kamailio and RTP Proxy and calls are going. But i m unable to force media to go thru RTP Proxy. How can i enforce media to go thru RTP Proxy ? Thanks -- Regards =================== Sujit Roy

3 2

Kamailio stopped relaying SIP text message from asterisk
by Adesh Pandey 25 Nov '19

25 Nov '19

Hi, I have been working on kamailio with following scenario: 1. Webrtc SIP Users register to Kamailio 2. Users initiate outgoing calls to some other numbers 3. Kamailio dispatches outgoing calls to Asterisk 4. Asterisk answers the call and then send a text message to the caller using SendText, which was working fine a few days back but all of sudden it has stopped working 5. At Asterisk end it is showing that message was delivered successfully but when I check SIP user logs, there is no … [View More]

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Unable to connect to sip server via proxy
by Ashwini Kumar 24 Nov '19

24 Nov '19

Hi I have two network interfaces on my system in which one interface is connected to my office WAN/LAN and other is connected to SIP private network connected to PSTN. Now, I want to access sip server using the internal IP of our office LAN as proxy. *192.168.6.2 (Office LAN IP as proxy) -> 10.50.185.2 (SIP)* So on machine 192.168.6.2, I setup kamailio as SIP proxy server so that our office LAN machine can connect to SIP server directly via SIP proxy. I used config defined in - *https://… [View More]

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Avoid full table scan for dispatcher table on db_redis using ds_reload() from config script
by Joel Serrano 23 Nov '19

23 Nov '19

Hello, I'm trying out redis as db backend, and right now I only have dispatcher records there... Every so often, I do a ds_reload() from within Kam config script, and I see in logs: Nov 22 20:36:35 test COPS[25531]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query After enabling debug logs: Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1761]: db_redis_query(): querying prefix (table) '… [View More]

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Question about ims_ipsec_pcscf
by Woscek, Martin W. 22 Nov '19

22 Nov '19

Hi all, We are looking for information as to what the ims_ipsec_pcscf module was written for and tested against. What was the target UE/EPC/PCRF that Tsvetomir Dimitrov wrote the module towards? Has an emulated EPC been tested with this module? If so which one(s)? Thanks in advance, Martin

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