- sr-users - kamailio.org

app_perl persistent database connection
by Vik Killa 15 Sep '14

15 Sep '14

hello, does kamailio start/run the perl script defined by modparam("app_perl", "filename", "/usr/local/test/bin/sips.pl") at startup and constantly run in background? I want a persistent database connection rather than having to constantly re-connect to database on each transaction. Is this possible? I'm connecting to postgres with use DBI; my $dsn = "DBI:Pg:database=$database;host=$host;port=$port"; my $dbh = DBI->connect($dsn, $user, $pass) || die "Could not connect to database: $DBI::errstr";

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Foward REGISTER to all dispatcher members
by Kees Meijs - SIGNET B.V. 15 Sep '14

15 Sep '14

Dear list members, Currently I'm using the dispatcher module to distribute traffic towards media servers (being Asterisk, in my case). I'd like to forward REGISTER requests to all (active) dispatcher members, e.g. using a short and simple while loop. Could someone please give me some pointers towards the right direction? Is it for example possible to use the MI di_list functionality for iterating over all the members? Thanks in advance! Best regards, Kees

2 1

Extract domain of a user from location list table
by aawaise 15 Sep '14

15 Sep '14

I went through usrloc page on kamailio site. Link given below. http://www.kamailio.org/docs/modules/3.1.x/modules_s/usrloc.html It states that the module keeps a user location table and provides access to the table to other modules. The module exports no functions that could be used directly from scripts. How can one extract value of domain from Location List ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-domain-of-a-user-from-locat… Sent from the Users mailing list archive at Nabble.com.

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Dispatcher - failure_route to specific node
by Nicholas Gill 13 Sep '14

13 Sep '14

Hello sr-users, We're using Kamailio to proxy to a number of FreeSWITCH backend servers with the dispatcher module. In some specific scenarios it matters to which node the request is sent. In these cases we've configured FS to respond to the invite with 302 and provide a custom header which contains the uri of the needed backend server (Intending to emulate the behaviour of 305 Use Proxy). To attempt to handle this I've tried the following in kamailio: onreply_route[MANAGE_REPLY] { ... if (t_check_status("302") and is_present_hf("X-imx-redirect")) { $avp(imx-redirect) = $hdr(X-imx-redirect); xlog("SCRIPT: REDIRECT, Store avp for redirect: $avp(imx-redirect)\n"); } } Then in the dispatch failure route: if ($avp(imx-redirect) != $null) { $du = $avp(imx-redirect); xlog("SCRIPT: REDIRECT: going to <$ru> via <$du>\n"); route(RELAY); exit; } Which results in the following sip event sequence: // Normal invite sequence FS1 -> Kamailio SIP/2.0 302 Moved Temporarily (X-imx-redirect = sip:FS2...) Kamailio -> FS1 ACK Kamailio -> Caller SIP/2.0 407 Proxy Authentication Required Then the caller generates a new INVITE however I'm unsure how to route this INVITE to the specific node. The avp appears (intentionally) tied to the transaction so doesn't seem to be able to be used to correlate in this scenario. Advice on how to redirect from a backend server to another specific backend server behind Kamailio would be greatly appreciated. Cheers, -nick

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Need help in how to read INVITE packet SDP c line IP Address
by varun pratapsingh 12 Sep '14

12 Sep '14

Hi All, Can anybody help me by telling how to read INVITE packet SDP c line IP address in Kamailio.cfg. For example a INVITE packet comes and its SDP contains the line v=0 o=- 3619547086 3619547086 IN IP4 1.223.65.215 s=pjmedia c=IN IP4 1.223.65.215 t=0 0 a=X-nat:0 Now I want to read this SDP packet c line IP address and want to put the IP 1.223.65.215 in a variable. Can anybody please help me out in this. Regards Varun

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New..... / kickoff point to learn Kamailio
by Poekel 12 Sep '14

12 Sep '14

Hello, I am totally new/blank at kamailio and would like learn more but do not know where to start........ I would like to build a voicemail platform with kamailio and Asterisk. ( if i CAN use kamailio for this purpose) Someone creates a mailbox in Asterisk realtime for example through a website. I want to manage the voicemail boxes from one place (kamailio?) and route sip traffic to the voicemail boxes? Can i do that with kamailio ? How? can someone give me some advice on how and where to start? Any help appreciated! Poekel

2 1

received and rport via parameters
by Daniel-Constantin Mierla 12 Sep '14

12 Sep '14

Hello, opening this for discussion to get more eyes on the IETF specs to be sure I haven't missed any requirements. We are no adding received parameter to Via header as required by RFC 3261: 18.2.1 Receiving Requests When the server transport receives a request over any transport, it MUST examine the value of the "sent-by" parameter in the top Via header field value. If the host portion of the "sent-by" parameter contains a domain name, or if it contains an IP address that differs from the packet source address, the server MUST add a "received" parameter to that Via header field value. This parameter MUST contain the source address from which the packet was received. This is to assist the server transport layer in sending the response, since it must be sent to the source IP address from which the request came. But then RFC3581 says: 4. Server Behavior The server behavior specified here affects the transport processing defined in Section 18.2 of SIP [1]. When a server compliant to this specification (which can be a proxy or UAS) receives a request, it examines the topmost Via header field value. If this Via header field value contains an "rport" parameter with no value, it MUST set the value of the parameter to the source port of the request. This is analogous to the way in which a server will insert the "received" parameter into the topmost Via header field value. In fact, the server MUST insert a "received" parameter containing the source IP address that the request came from, even if it is identical to the value of the "sent-by" component. Note that this processing takes place independent of the transport protocol. I understand that if rport parameter is present in incoming Via, received has to be added always, even when matching the IP address in via. RFC3261 required to be added only when it source IP was different than the address in Via. On the other hand, the RFC3261 says received is needed for helping the server to route back the reply, so received shouldn't make much sense for upstream. However, apparently, some clients use it (or at least check it and don't work properly if not present after a rport -- one specific example is that some phone is not making calls after sending REGISTER with rport and not receiving received). RFC3581 says that server behavior affect what was specified in 3261, so my plan is to send received always if rport is present, but I wanted to see if anyone sees issues with that or there is another more recent rfc amending further the behaviour related to rport. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

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Dispatcher - failure_route to specific node
by Nicholas Gill 12 Sep '14

12 Sep '14

Hello sr-users, We're evaluating Kamailio to proxy to a number of FreeSWITCH backend servers with the dispatcher module. In some specific scenarios it matters to which node the request is sent. In these cases we've configured FS to respond to the invite with 302 and provide a custom header which contains the uri of the needed backend server (Intending to emulate the behaviour of 305 Use Proxy). To attempt to handle this I've tried the following in kamailio: onreply_route[MANAGE_REPLY] { ... if (t_check_status("302") and is_present_hf("X-imx-redirect")) { $avp(imx-redirect) = $hdr(X-imx-redirect); xlog("SCRIPT: REDIRECT, Store avp for redirect: $avp(imx-redirect)\n"); } } Then in the dispatch failure route: if ($avp(imx-redirect) != $null) { $du = $avp(imx-redirect); xlog("SCRIPT: REDIRECT: going to <$ru> via <$du>\n"); route(RELAY); exit; } Which results in the following sip event sequence: // Normal invite sequence FS1 -> Kamailio SIP/2.0 302 Moved Temporarily (X-imx-redirect = sip:FS2...) Kamailio -> FS1 ACK Kamailio -> Caller SIP/2.0 407 Proxy Authentication Required Then the caller generates a new INVITE however I'm unsure how to route this INVITE to the specific node. The avp appears (intentionally) tied to the transaction so doesn't seem to be able to be used to correlate in this scenario. Advice on how to redirect from a backend server to another specific backend server behind Kamailio would be greatly appreciated. Cheers, -nick

1 0

send_reply in app_java or app_mono?
by Tom Johnson 11 Sep '14

11 Sep '14

Is there a way to call the send_reply method of the route from either Java or csharp? As I have been having issues with the app_java, I have been looking at app_mono and it seems to have better access to the variables, etc. However, the SR.core.ModF only seems to take methods that do not take parameters. Or for that matter, is there a way to return something back to the route in the config file which it can use to call send_reply? I have a strange requirement to return 202 rather than 200 from subscription requests and a 603 if it fails. Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

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app_java issue
by Tom Johnson 11 Sep '14

11 Sep '14

Per my previous message, I determined that the module's makefile was not properly finding the jdk/jre directories needed to compile. So, I temporarily hardcoded the paths in the makefile. However, now when I use the module, Kamailio crashes. Below is the lines from /var/log/messages: Sep 9 15:13:10 MAIFREEPBX kamailio: INFO: <core> [tcp_main.c:4836]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected) Sep 9 15:13:10 MAIFREEPBX kamailio: WARNING: <core> [daemonize.c:352]: daemonize(): pid file contains old pid, replacing pid Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 512 Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:125]: mod_init(): Initializing Java VM with options: -Xdebug -verbose:gc,class,jni -Djava.compiler=NONE -Djava.class.path=/opt/microautomation/i3/I3Notifier.jar:/opt/kamailio/java/kamailio.jar:/opt/microautomation/i3/lib/log4j-1.2.16.jar Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:146]: mod_init(): app_java: Java VM initialization OK Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:198]: mod_init(): app_java: module initialization OK Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: : <core> [mem/q_malloc.c:159]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(0, abcdefed)[0x7f0941207190:0x7f09412071c0]! Sep 9 15:13:11 MAIFREEPBX abrt[12282]: Saved core dump of pid 12268 (/usr/local/sbin/kamailio) to /var/spool/abrt/ccpp-2014-09-09-15:13:10-12268 (183721984 bytes) The code dump mentioned does not exist. If I do not load the app_java module, Kamailio works. Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

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websocket:ws_current_connections vs. kamctl fifo ws.dump
by Luis F Urrea 11 Sep '14

11 Sep '14

Hello I have a question in regards monitoring websocket connections on a production box. We ususally trust whatever is indicated through the $ kamctl fifo get statistics We've seen that this is what the community contributed munin plugins use. However when trying to monitor web socket connections and using this output we've seen completely different results from the output of $ kamctl fifo get_statistics websocket:ws_current_connections Which seem to indicate current websocket connections and the output of $ kamctl fifo ws.dump Which seems to indicate current OPEN connections. Can anybody shed some light as to what are the differences between both if any, and advice which one is the proper one to monitor for possible file handle exhaustion. TIA

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Auto-response
by Mick Stevens 11 Sep '14

11 Sep '14

1 0

Re: [SR-Users] TLS Handshake failing with WSS
by Manuel Camarg 11 Sep '14

11 Sep '14

According to this article I mentioned in the beginning: http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket In http:request they do straight away the set_reply_close(); I don't know exactly if this kamailio.cfg implementation may be useful for working with WSS: 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; } } HTTP authentication is not being properly set up. Any indications on how to perform this sucessfully? Kind regards Manuel >Hello, > >from the logs you sent now, it appears that you have set_reply_close() >in config, therefore the connection is closed after sending the reply. > >Cheers, >Daniel

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How to increment dialog calee_cseq value when using t_uac_dlg via xmlrpc for generating INFO requests
by Drazen Rep 11 Sep '14

11 Sep '14

Hi, I am using Kamailio 4.1.1 and dlg_manage() function for processing SIP requests with dialog module. After call is established Kamailio is being used for generating in-dialog INFO request toward caller (using t_uac_dlg function via xmlrpc). INFO messages are correctly generated but dialog calee_cseq value is not being incremented with every message. That represents a problem since other SIP requests generated by callee side (like OPTIONS) are not in order, and there is no correct information about calee_cseq value in dialog memory records. The question is how can I increment dialog calee_cseq value when local INFO request toward caller is being generated using t_uac_dlg via xmlrpc? Thx, Drazen

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UAC module
by Yuriy Gorlichenko 11 Sep '14

11 Sep '14

Hello. I tried to use UAC module at my system for registration multiple providers, and call through it. When I implementing this at my cfg file I had some troubles: -- with Authentification at INVITE (using UAC_AUTH). IT does not send second INVITE packet with www_auth header for some trunks (not for all, only for some providers) -- with BYE recieving from PROVIDER ("No BYE from Called party" task at USers Mailing lists) After many days of tests and debugging I ask question from myself - does module correctly works? As I know this module breaks some philosophy about SIP proxy and development of this module is not question of first priority. After that all I need advice - what I must use for registtration my SIP trunks - 3d party registrant or this module, that works fine and I doing something wrong? Thanks for advice.

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Kamailio v4.2 - development frozen
by Daniel-Constantin Mierla 11 Sep '14

11 Sep '14

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 4.2.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 4.2 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to rich a stable state before releasing 4.2.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Help with updating the wiki page for migration from 4.1 to 4.2 as well as what is new in 4.2 is very appreciated. I will post updates about the very soon. Many thanks to everyone involved in development of 4.2 and the early testers that played with master branch. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

LCR base on call rate
by Satish Patel 11 Sep '14

11 Sep '14

I heard somewhere LCR can do routing based on call rate and call price, does it true? I haven't seen any config or doc which does call rate using LCR. It only does routing base on prefix scan. Am I missing something here? Sent from my iPhone

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WebRTC meeting Norfolk, 15 October 2014
by Daniel Pocock 10 Sep '14

10 Sep '14

I'll be in Norfolk, VA for xTupleCon in October On 15 October, there will be two events for WebRTC: 14:15 a talk about the xTuple WebRTC extension at xTupleCon - must register for xTupleCon to attend this 17:30 a technical / developer workshop at xTuple's offices - free, anybody welcome, even if not attending xTupleCon, RSVP through Eventbrite[1] Please see my blog[2] for more comments about all of this and feel free to email me in advance if you have questions about it or if you may like to meet up there. 1. http://www.eventbrite.com/e/browser-based-webrtc-telephony-for-web-apps-wor… 2. http://danielpocock.com/xtuplecon-webrtc-talk-schedule-change

1 0

TCP SYN only
by Laurent Schweizer 10 Sep '14

10 Sep '14

Dear all, Since few weeks I have more and more TCP issue, I can see that sometimes the client send a TCP SYN but never get any response. 106.123315 62.12.179.104 -> X.X.X.X TCP 39174 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294942209 TSER=0 WS=1 109.124384 62.12.179.104 -> X.X.X.X TCP 39174 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294942510 TSER=0 WS=1 111.202946 62.12.179.104 -> X.X.X.X TCP 4993 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294942717 TSER=0 WS=1 114.204707 62.12.179.104 -> X.X.X.X TCP 4993 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294943018 TSER=0 WS=1 I have found on internet many doc regarding TCP and max connection issue and I have already done change in sysctl (see below) to increase the number of tcp connection but seems still not ok. Any idea ? Best regards Laurent net.ipv4.ip_forward = 0 net.ipv4.conf.default.rp_filter = 1 net.ipv4.conf.default.accept_source_route = 0 kernel.sysrq = 0 kernel.core_uses_pid = 1 net.ipv4.tcp_syncookies = 0 net.bridge.bridge-nf-call-ip6tables = 0 net.bridge.bridge-nf-call-iptables = 0 net.bridge.bridge-nf-call-arptables = 0 kernel.msgmnb = 65536 kernel.msgmax = 65536 kernel.shmmax = 68719476736 kernel.shmall = 4294967296 net.ipv4.tcp_timestamps = 1 net.ipv4.tcp_fin_timeout = 30 net.ipv4.tcp_keepalive_time = 1200 net.ipv4.tcp_max_syn_backlog = 8192 net.ipv4.tcp_max_tw_buckets = 5000 fs.file-max = 2097152 net.ipv4.ip_local_port_range = 2000 65535 net.ipv4.tcp_max_tw_buckets = 1440000 net.ipv4.tcp_tw_recycle = 1 net.ipv4.tcp_tw_reuse = 1 net.core.somaxconn = 65536 net.core.netdev_max_backlog = 65536

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Kamailio Installaton compatibility with ARM9 Architecture
by Nandini madhu 10 Sep '14

10 Sep '14

Dear all, Greetings. We need to install Kamailio server in ARM9 architecture based system (Example: Xilinx XC7Z045-2FFG676I FPGA board). Pleasse suggest me that, is there any possibility of installing Kamailio in ARM9 architecture based systems. Please respond Regards sermj2012

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New developer: Federico Cabiddu
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, I would like to announce that Federico Cabiddu has now write access to Kamailio git repository. He is going to push a new module named tsilo, which facilitates adding new branches to active transactions -- the typical case implemented so far is to track invites per callee and add a new branch if the callee registers while invite transaction is active, quite useful for services with mobile clients that are getting push notifications to wake up. Federico is one of the guys using Kamailio for many years, currently working at Orange, France. His commit user id is: fcabiddu My warm welcome and thanks for joining the effort to develop Kamailio! Looking forward to future contribution to the project, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

new module db_kazoo
by Luis Azedo 10 Sep '14

10 Sep '14

Hello Awesome Kamailio Community, We are writing on behalf of 2600hz, where we have been using Kamailio for some time now and are very pleased with our results! So, seems time to commit something back, dontchya think? We would like to present to you a new module, to hopefully be included in master. We call it db_kazoo (although a new name is fine too). db_kazoo is a general purpose AMQP connector (connects to our rabbitmq-server). It exposes publish/consume capabilities into Kamailio. Why is this amazing, you ask? Well even if you didn’t ask, we will explain… >From a high-level, the purpose of the module might be for things like: - Integrate to an AMQP application to make real-time routing decisions (instead of using, say, a SQL database) - Provide a real-time integration into your program, instead of your database, so you can overlay additional logic in your preferred language while also utilizing a message bus - Utilize messaging to have a distributed messaging layer, such that machines processing requests/responses/events can go up/down or share the workload and your Kamailio node will still be happy With this module, someone can: 1 - publish json payloads to rabbitmq 2 - publish json payloads to rabbitmq and wait for correlated response message 3 - subscribe to an exchange with a routing key The module works with a main forked process that does the communication with rabbitmq for issuing publishes, waiting for replies and consuming messages. When it consumes a message it defers the process to a worker process so that it doesn't block this main process. The worker process issues an event-route where we can act on the received payload. The name of the event-route is composed by values extracted from the payload. Consumed messages have the option of being acknowledge in two ways: 1 - immediately when received 2 - after processing by the worker One unique feature of our implementation revolves around failover of the message bus itself. In our design, the module supports multiple RabbitMQ servers and will fallback from one to the next in a list of RabbitMQ servers if the connection fails to the current connected server. When using acknowledge in db_kazoo with clustering in RabbitMQ, we have simulated and experienced full reconnects while in the middle of processing pending messages. In this way, when we get disconnected from one server we proved that, even at high speeds, we are able to connect to the next in the list and continue interacting with our application. We have run a variety of sipp load tests on this module and believe it is ready for prime time. We monitored memory, response accuracy and overall stability and it seemed OK. But we would, of course, love for others to help us find what we have missed, or contribute more features, or overall just use the work we slaved over for so many hours. Or just make comments and suggestions! If nothing else, we hope we’ve provided something useful to the Kamailio community, as you all have provided useful items to us. Thanks to everyone for all the work on Kamailio and related products. Look forward to hearing from you all. This is our first formal / large contribution to Kamailio so if we’ve done something wrong process, code or otherwise please let us know!

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Location of Encryption Algorithm in TLS Module
by aawaise 10 Sep '14

10 Sep '14

Hello, I have downloaded and unzipped kamailio package available online. I want to know about TLS module, My question is that where is the encryption module located in TLS module ?? Is it in some .c file of TLS module or does TLS module use AES encryption available in LINUX or UNIX systems?? This question arises because I want to customize AES algorithm ?? Can it be done in TLS module or not ?? Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Location-of-Encryption-Algorithm-in… Sent from the Users mailing list archive at Nabble.com.

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Freezing development for v4.2 at the end of today
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, the development of new features for next major release, v4.2.0, is going to be frozen by end of today. As we have developers all over the world, we are going to allow everyone to benefit of the full day, so technically is like end of Wednesday, September 10, in Hawaii ( :-) a good fit, because it is the home region for the name of the project. If there is a patch with new features that you want to push, but needs review, we are going to take care of them in the next days, but upload them to the tracker or send them to sr-dev mailing list by today. Then follows about one month of testing and bug fixing for the existing code, with the target to have the 4.2.0 release out around mid of October. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

New developer: Luis Azedo
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, I would like to announce that a new developer has now write access to Kamailo git repository: Luis Azedo. His commit username is: lazedo He is going to push soon the 'kazoo' module that was discussed these days on the mailing list - a module connector for the Kazoo - the open source, distributed VoIP platform using Kamailio for core SIP routing. He also provided many patches to other components so far. Luis works for 2600hz, an USA company, and will be accompanied by Darren Schreiber and Karl Andreson in maintenance of the new module. My warm welcome and thanks for joining the effort of developing Kamailio further! Looking forward to future contributions to the project! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

No BYE from Called party
by Yuriy Gorlichenko 09 Sep '14

09 Sep '14

Hello All. I have kamailio with provider connection (trunk) When I call to external number through my provider call extablished Ok. But when i try hangup call from external number no BYE sended to me. When I hangup call from my kamailio (internal num) I send by to exteral number and it respond me Ok so session if fully complete. I guess that BYE from external number not recieves to me because I have wrong routing header fields at my INVITe or ACK messages, but can not find any information what what header must recieve info to external number where send BYE at hangup or thomething like this. This is my little dump for situation wherer I hangup from internal number and BYE finished successfully: IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599 E....< .@.'. ...6........G.RINVITE sip:12345678900@my.provider.com:5060 SIP/2.0 Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Max-Forwards: 70 From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068> Contact:<TrunkNum@my.kamailio.com:5068> Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 04 Sep 2014 21:53:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com", nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri=" sip:12345678900@my.provider.com:5060", qop=auth, nc=00000001, cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee", algorithm=MD5 v=0 o=root 1022912010 1022912010 IN IP4 my.kamailio.com s=Asterisk PBX 12.5.0 c=IN IP4 my.kamailio.com t=0 0 a=ice-lite m=audio 30032 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30033 a=ice-ufrag:3o8JrqkF a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2 a=candidate:TgT1dfTnI3kBgWQ IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882 E..... ./...6... ........b..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE Contact: <sip:12345678900@externail.number.end.ip:5060;transport=udp> User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 746 X-provider agentOutboundGateway: sip:5574012345678900@62.93.147.149 X-provider agentOutboundCarrierID: 23705946361020 X-provider agentCarrierRate: 0.20180 X-provider agentCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@my.provider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip s=FreeSWITCH c=IN IP4 externail.number.end.ip t=0 0 a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP m=audio 23216 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=ssrc:2990874569 cname:pRs5xP IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596 E..p.@..@.K\ ...6........\`.ACK sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617 E....D..@.KC ...6........q.ZBYE sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: Asterisk PBX 12.5.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568 E..T....-.676... ........@..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 Thanks for help.

2 4

Using app_java
by Tom Johnson 09 Sep '14

09 Sep '14

Any tips on compiling the app_java module? I am on CentOS 6. And I have libgcj-4.4.7 installed, but the build process complains about missing jni.h and other files.

2 1

Creating array out of hashtable
by aft 09 Sep '14

09 Sep '14

[1]Is it possible to iterate a htable without using "keys" like an array? [2] Is it possible to search for "keys" using regxp? -- -Cheers -Arif

2 2

Call connection from one server to another
by aawaise 09 Sep '14

09 Sep '14

Hello, I want to deploy two kamailio servers, both using same MYSQL database placed on Server 2008. Both are using different static IP's. Hopefully, I can add two A records in DNS server, both against same domain name, pointing to two different IP's to which two servers are attached. My question is that in load balancing scenario if user 1 is attached to server 1, and user 2 is attached to server 2. How can user1 call user2. Because request will go to server 1 to which user 1 is attached. But it wont be able to locate user 2 because it is not connected to server1 (but connected to server 2). I am using same domain name for two servers. What can be done in kamailio to solve this issue ?? Regards, aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Call-connection-from-one-server-to-… Sent from the Users mailing list archive at Nabble.com.

3 3

TLS Handshake failing with WSS
by Manuel Camarg 08 Sep '14

08 Sep '14

I'm trying to implement WSS with Kamailio Thing is that WS works fine, I've followed: http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket modparam("tls", "config", "webrtc/tls.cfg") In a tls.cfg file I have : [server:default] method = SSLv23 verify_certificate = no require_certificate = no private_key = webrtc/private.key certificate = webrtc/ssl.pem ca_list = webrtc/ca_list.pem In the log file: /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=5524 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8e0040, 11, 2, 0x7f7513516958), fd_no=1 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:712]: sr_ssl_ctx_info_callback(): SSL handshake done /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL disable renegotiation /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 123.123.123.123:63300 using TLSv1/SSLv3 AES256-SHA 256 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 124.124.124.124:10443 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=282 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f7513516958, FD 11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1293]: tcp_read_req(): tcp_read_req: EOF /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e0040, 11, -1, 0x10) fd_no=2 called /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f7513516958, state -1, fd=11, id=2 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f7513510a88 /usr/local/sbin/kamailio[4029]: DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f7513516958, -1 from 1 /usr/local/sbin/kamailio[4029]: DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7f7513510a88 In sipml5 the error: *Disconnected: Failed to connect to the server* In the Chrome console: *__tsip_transport_ws_onerror * *__tsip_transport_ws_onclose * SSL certificates seem to be ok: # openssl verify -CAfile ca_list.pem ssl.pem ssl.pem: OK Can't figure out a solution :( Any ideas? *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

2 3

Change caller name in kamailio as the extension remains same.
by Talha Bukhari 08 Sep '14

08 Sep '14

Hi, I am new to kamailio. i configure kmailio server and make call as local extensions like 1001 and etc. Now i want to know how to change the header name appear on softphone using script in kamailo as the extension remain same as 1001. what are the files i used to alter and where are to locate them. Hope you don't mind. Regards, Talha

2 4

out of memory ; needs 455
by Arun Kumar 08 Sep '14

08 Sep '14

Hi Server : kamailio 4.1.1 Configuration :Outbound with engage_media_proxy (See the attachment ) Shared Mem : 1024 MB Problem : after 1 month kamailio hang up with out of memory error but we have enough free shared memory , it might be memory pool size problem, i dont know how and where to set memory pool size

2 1

Kamailio - voicemail platform - asterisk
by Henny 08 Sep '14

08 Sep '14

Hello, I am new to kamailio...... I have an idea for a community voice-mail platform (with Asterisk)and was wondering if Kamailio could be of any use in this plan. Does somebody have any advice ? Thanx!

2 1

dialog match on loop back
by Kelvin Chua 06 Sep '14

06 Sep '14

hi guys, i'm running out of ideas i have a situation where I need to generate 2 sets of entries INVITE,BYE in acc (acc_db_request won't cut it) it's when a DID should be redirected to a user outside my domain, I need to generate a cdr for DID and another CDR for the outbound call. what i am trying to do is loop it back to itself before routing it out. the idea is kamailio assigns the DID call to a dialog. then the outbound call will be assigned to a separate dialog after it receives the loopback. this is not happening since the from_tag, to_tag and the callid are all the same. kamailio treats this call as just 1, not 2. is there any ideas out there to work around this scenario? Kelvin Chua

2 1

Adding IPs from which 503 messages originates to a database
by aft 06 Sep '14

06 Sep '14

I'm trying to configure HOMER sip capture server to do some accounting of the cases when "503 Service Unavailable" message arrives from capture agents. I've implemented the following to record the source IPs from which 503 message originates. onreply_route { ...................... ...................... ...................... else if(status == "503") { if($sht(a=>alarm::503) == $null) { $sht(a=>alarm::503) = 0; $sht(a=>alarm::$si::503) = 0; } xlog("Got 503"); xlog("($sht(a=>alarm::$si::503))"); $sht(a=>alarm::$si::503) = $sht(a=>alarm::$si::503) + 1; } ................... ................... ................... The following route supposed run periodically to update the databases. } route[TIMER_STATS] { ..................... ..................... xlog("($sht(a=>alarm::$si::503))"); if($sht(a=>alarm::$si::503) > 0) { sql_query("cb", "INSERT INTO alarm_data (create_date, type, total, description, source_ip) VALUES(NOW(), 'Too Many 503', $sht(a=>alarm::$si::503), 'service unavailable', $si)"); xlog("adding stuff to db"); } $sht(a=>alarm::503) = 0; $sht(a=>alarm::$si::503) = 0; } But the xlog("adding stuff to db") never executed. But in the first block of code, the log message to print the value $sht(a=>alarm::$si::503), which is shown to be non zero. Don't understand why its not working. -- -Cheers -Arif

2 4

Add custom parameter to To: header, and then recover it
by Alex Villacís Lasso 05 Sep '14

05 Sep '14

What functions exist (if any), to add custom parameters to the From: and To: headers?

2 1

Kamailio as a front end?
by Tom Johnson 05 Sep '14

05 Sep '14

I am looking for something that would act as front end to various SIP PBX systems. I am using Asterisk, but compatibility with others might be necessary. What I want to do is pass subscribe requests with certain content types to an external program (written in Java) and the rest onto the SIP PBX. Can Kamailio do this and if so, is there some example config files somewhere that I can look at to get started? Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

2 1

mt_match avp behavior
by Juha Heinanen 05 Sep '14

05 Sep '14

looking at the code, i appears that mt_match does not initialize pv_values pv when it stores matched values thus keeping possible existing values from previous calls. is that intended behavior? readme uses word "store". -- juha

2 5

mtree README missing params
by Juha Heinanen 05 Sep '14

05 Sep '14

looks like 4.1 and master mtree README is missing these parameters: {"pv_dstid", STR_PARAM, &dstid_param.s}, {"pv_weight", STR_PARAM, &weight_param.s}, {"pv_count", STR_PARAM, &count_param.s}, i didn'd find in the code any use for the last. -- juha

2 1

has "defined" been documented somewhere?
by Juha Heinanen 05 Sep '14

05 Sep '14

defined($avp(foo)) can be used to test if avp $avp(foo) has been defined. i looked through kamailio wiki, but didn't find defined function documented anywhere. -- juha

2 1

ldap returning attributes list
by Virmantas Variakojis 05 Sep '14

05 Sep '14

Hi, I'm trying to fetch attribute list with kamailio ldap_search: ldap_search("ldap://agents/ou=Xxxxx,dc=xxxx,dc=yyyyyy,dc=zz?memberOf?sub?(sAMAccountName=$fU)"); ldap_result("memberOf/$avp(s:ldapmemberof1)"); xlog("L_INFO", "LDAP $fU memberOf = $avp(s:ldapmemberof1)"); if (ldap_result_next()) { xlog("L_INFO", "if loop ldap_result"); ldap_result("memberOf/$avp(s:ldapmemberof2)"); xlog("L_INFO", "LDAP $fU memberOf = $avp(s:ldapmemberof2)"); } There's must be seven returning attributes list of memberOf, but I can only retrieve only one of them (looks like last of seven). First xlog output log into my log file, but if loop didn't start. What I'm doing wrong?

3 2

create loop for REGISTER SEND in Kamailio.cfg
by ANTHONY HESNAUX 05 Sep '14

05 Sep '14

hello everybody, I would like loop in kamailio.cfg because I want register soft phone on all asterisk system so I have this configuration but also if I want add another asterisk system my Kamailio.cfg #!define WITH_ASTERISK #!define WITH_CLUST_AST #!ifdef WITH_ASTERISK asterisk.bindip = "asterisk.home.fr" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = « kamailio.home.fr" desc "kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif #!ifdef WITH_CLUST_AST #!define HOST 2 asterisk1.bindip = "192.168.0.10" desc « Asterisk - 1 - IP Address" asterisk1.bindport = "5080" desc "Asterisk Port" asterisk2.bindip = "192.168.0.20" desc « Asterisk - 2 - IP Address" asterisk2.bindport = "5080" desc "Asterisk Port" #!endif # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk1.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk1.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $var(rip) = $sel(cfg_get.asterisk2.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk2.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif thank you for your help anthony

2 1

Dispatcher algorithm 9 ignores weight=xx attribute
by Alejandro Recarey 05 Sep '14

05 Sep '14

Hi, I have a kamailio with the dispatcher module setup. I use: ds_select_dst("1", "9"); to load balance between 3 gateways. My database looks like this: id setid destination flags priority attrs 5 1 sip:1.1.1.1 0 0 weight=10 8 1 sip:1.1.1.2 0 0 weight=45 9 1 sip:1.1.1.3 0 0 weight=45 As I understand it, most of the calls should be split evenly between gateways 8 and 9, and some calls (around 10%) going to gateway 5. Well, gateway 8 gets 99% of the calls. I am using kamailio 4.1 and I really cannot understand why this is happening. Running ds_list shows me that all gateays and weights are correctly parsed into kamailio. This is the result of my ds_list: SET:: 1 URI:: sip:1.1.1.1 flags=AP priority=0 attrs=weight=10 URI:: sip:1.1.1.2 flags=AP priority=0 attrs=weight=45 URI:: sip:1.1.1.3 flags=AP priority=0 attrs=weight=45 I have tried changing the priority to 100, or to 1 in every gateway, and the result is still the same. I have tried adding ; after the weight parameter, the colon is gone in the ds_list once kamailio parses it. If, keeping the same database, I use ds_select_dst("1", "0") the calls are split evenly across all 3 gateways. This is great, but not what I want, but it at least proves that something is working. Is there anything else I can try? Thanks a lot for the help! Alex

2 1

OT: RTCP logging
by Daniel Tryba 04 Sep '14

04 Sep '14

Slightly offtopic, but for QA reasons I'm interested in logging RTCP reports from peers that send them. It looks like rtpproxy-ng might be used to do this (I'm already using rtpproxy anyway). Is someone already doing this and willing to share some pointers?

3 4

Looking for some help
by Sharan Harkisoon 04 Sep '14

04 Sep '14

I am in need of a Kamailio expert that has some availability for consulting services (remote is fine). Feel free to contact me for details. Thanks, Sharan Harkisoon

2 1

Redirect server vs CANCEL
by Mihaly Zachar 04 Sep '14

04 Sep '14

Hi Gents, We have created a small redirect server using the Lua API to get the destinations. Lets say that it takes some time while the script generates the answer, during this time we got a CANCEL. How we can handle this properly ? As far as I could see the TM module always sends back the answer (302) and after that I get the "CANCEL" in my script. Here is my config: http://pastebin.com/7PyKBJyD Here is my test Lua script: http://pastebin.com/6VXn5y8m Could you please help me what is the best way to handle the CANCEL ? Is it possible to do it properly ? Thanks, Misi

2 1

nat_traversal vs nathelper for mediaproxy
by Satish Patel 04 Sep '14

04 Sep '14

I am planing to implement mediaproxy so which NAT module will be good for media proxy and why? should i use nat_traversal ro nathelper ? I haven't see any example people using nat_traversa. most of example i found on google are based on nathelper so just want to make sure which one i should use and why?

2 1

Shared memory
by Morten Tryfoss 04 Sep '14

04 Sep '14

Hi, I wonder about the difference between “used” and “real used” shared memory in Kamailio. Best regards, Morten Tryfoss

2 1

How to determine if a 4xx message came from dispatcher or a client?
by Olli Heiskanen 04 Sep '14

04 Sep '14

Hello, A question on Kamailio variables and using dispatcher: When in failure_route I want to know if the request message was going to a dispatcher ip or a sip client ip (as in any other than dispatcher ip), how do I make an if statement for that? If I use ds_is_from_list(), I get wrong results because in both cases the request was sent by Kamailio. Also, I don't seem to be able to get the source ip of the 4xx message (src_ip shows the source ip of the request message). cheers, Olli

2 1

Can log in from SIP
by Alexis Walhovd 04 Sep '14

04 Sep '14

Hi I just created an account with iptel.org. But I can not log in to my account through my SIP software from OSX. Please see below for details: I type “Alexis82” as password. All the best, Alexis

2 1

need help - Insert_hf when Route: missing
by Satish Patel 04 Sep '14

04 Sep '14

I have post many question on topology hiding any not get any reply back from people and developers now I don't have any option except some goofy hack When I use topology hiding it removes Route: and because of that callee doesn't able to send BYE back to opensips. I want use insert_hf to inject Route: so I get clean BYE message. Do you guys have any suggestion? Sent from my iPhone

2 1

Asterisk cluster behind kamailio natted to pubic IP, presenting internal ip addresses in From tag
by Tim Chubb 04 Sep '14

04 Sep '14

Hi All I'm scratching my head on this one, I have set everything up in the following configuration: Public ip natted to DMZ via Cisco ASA 5510 <= DMZ => Kamailio & RTP Proxy <= Voice DMZ => Asterisk My kamailio configuration is based heavily on the one found here: http://saevolgo.blogspot.co.uk/2013/08/rtpproxy-revisited-kamailio-40.html & http://saevolgo.blogspot.co.uk/2011/11/how-to-increasing-voip-services.html So far everything is working fine, I can register via the public IP address, IM & presence is working, and as does audio, however when I dial an extension the caller id comes up like this 12345@172.16.15.123:5080<mailto:12345@172.16.15.123:5080> which is the ip addresses of the asterisk server that the dispatcher has assigned to the call, I have tried setting the P-Asserted-Identity, P-Preferred-Identity & Remote-Party-ID headers to no effect. What I would like to achieve is that the public IP or domain name comes up when I call an extension. From examining the sip traffic traversing the kamailio box, it seems that the From and Contact headers sent by asterisk are the source of the internal implementation information reaching the end-user. How can I prevent implementation information like this leaking out? I consider this to be a security risk as I don't want anyone to be able to enumerate the number of servers that comprise a given cluster as easily as placing a few calls, ideally I only want data relating to the public side of the network getting out. But I would be happy if it takes digging into the sip traffic to glean any details initially. I'm getting the feeling that this could well be an asterisk configuration issue, so apologies if this is a little off topic, however I have been unable to find a similar configuration to try and achieve my desired results. Tim.

3 3

Support for TLS server_name extension (aka SNI=server name indication)
by Barry Flanagan 04 Sep '14

04 Sep '14

Hi, Back in Kamailio 1.5, the release notes state: "support for TLS server_name extension (aka SNI=server name indication)" However, I cannot find any indication of this in the current TLS docs, and trying to set tls_server_name or server_name in tls.cfg fails with "unsupported option". Is this actually supported? Thanks. -Barry Flanagan

4 6

Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)
by Alex Villacís Lasso 04 Sep '14

04 Sep '14

Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170: IP 198.58.101.75.5060 > 201.234.196.170.5060 INVITE sip:*43@201.234.196.170:5060 SIP/2.0 Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport Max-Forwards: 70 From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060> Contact: <sip:9002@198.58.101.75:5060> Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Fri, 29 Aug 2014 18:23:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 299 v=0 o=root 521741684 521741684 IN IP4 198.58.101.75 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 198.58.101.75 t=0 0 m=audio 16426 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv IP 201.234.196.170.5060 > 198.58.101.75.5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060 From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060> Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE Server: kamailio (4.1.5 (x86_64/linux)) Content-Length: 0 IP 201.234.196.170.5060 > 198.58.101.75.5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes> From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060>;tag=as2798a3b9 Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:*43@127.0.0.1:5080;alias=127.0.0.1~5080~1> Content-Type: application/sdp Require: timer Content-Length: 305 v=0 o=root 159029581 159029581 IN IP4 201.234.196.170 s=Asterisk PBX 11.12.0 c=IN IP4 201.234.196.170 t=0 0 m=audio 18446 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes According to a strict interpretation of the SIP RFC, which address should the machine at 198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract said address? I am trying to understand an issue of a missing ACK between 201.234.196.17x and a different public IP, with the only difference that the other IP is not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I want to know whether the packets correctly indicate the address for the ACK, or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.

3 7

want to know about best book or best way to learn kamailio.
by Talha Bukhari 04 Sep '14

04 Sep '14

please guide me, how i learn kamalio as i am new to kamailio. suggest me some book or resources. waiting.

2 1

Re: [SR-Users] How to add route header (Senthil K)
by Surendra 03 Sep '14

03 Sep '14

Hi Senthil, You can use record_route function to add Record_route header in INVITE message( use rr.so module). This record_route header will become Route header while sending BYE message from client. Suppose if you want add directly Route header, then you can use Insert_hf function. Please go thru the textops modules for the same. Regards surendra

1 0

How to uniquely identify SIP WS / WSS endpoint
by Muhammad Shahzad 03 Sep '14

03 Sep '14

Sorry for putting this question on both dev and user mailing lists, as it is a rather theoretical question and i hope some SIP guru on either mail list will answer. For non-WS endpoints which use TCP or UDP for SIP transport, each upstream request has top most VIA header pointing to the previous hop which forwarded the request to current hop while each downstream request has top most VIA header pointing to next hop to which it will be forwarded from current hop. But for WS endpoints, the top most VIA has dummy static value, so there is no way to identify who sent this request or to whom the reply is going to. Please note that i am not specifically interested in network address of remote endpoint (though VIA header is suppose to provide it), i only need to match requests and responses from / to a specific device using SIP v2.0 standard. Any help is highly appreciated. Thank you.

3 7

Use Kamailio as (SIP-TLS) Gateway to connect to (non TLS) fritz.box for fixed-line calls
by Lukas Wygasch 02 Sep '14

02 Sep '14

Dear Kamailio-Community, i would be very pleased when someone could help me with my private non-profit project. I do not want other people to do my work, but i'm not as good with SIP as i should to solve this problem by myself. So here is my Goal: Use Kamailio as (TLS-SIP-)Gateway to my (non-TLS)-FritzBox to do (local-country) fixed-line Calls from all over the World. In Detail: Connect to (Portforwarded / NAT behind Fritzbox-Router) Kamalio-Server with User from "Subsriber-List" by TLS-SIP Now I want Kamailio to connect to local fritz.box (by SIP) with one fritzbox-user to handle all invites and calls. ------------------------------- Scenario (fictional): Fritzbox (Router and VOIP-Server) Internal-IP: 192.168.178.1 or fritz.box External-IP: dynip.de VOIP-User: 610 VOIP-Passwort: test Kamailio (running on RaspberryPI and Siremis) Internal-IP: 192.168.178.2 Network Gateway: 192.168.178.1 Connection from Outside (NAT): dynip.de:5061 User: euser Pass: epass ------------------------------- Questions: Is there a Module i could use to do this Job? (configurable through Siremis e.g. Dispatcher List?) How can i ensure that my user "euser" from subsriber-list connects with 610:test to fritz.box? Is there a simple configuration to solve this problem? I'm sorry to ask you these my non-skilled questions, since it is not my business to use SIP, but this scenario could be a useful tutorial for many many fritz.box users over the world. Roaming prices for mobile-calls could be safed with sitting into next McDonalds and do a SIP-Call through free Wifi. Thank you very much! I would be very grateful to get some help. Sincerly yours Lukatz

2 1

Regrading Kamailio
by Saurabh Srivastava 02 Sep '14

02 Sep '14

Hello Kamailio Team, I am using kamailio server successfully for some time now. I have a specific set of problem. We have a Kamailio Server setup in Amazon EC2 Cloud, the users are able to register on the server and make use of this. *Now the problem:* If I am trying to register multiple SIP clients from same Public_IP:Private_IP (i.e. alice and bob from same computer) on a kamailio server which is sitting behind Amazon EC2 Cloud Based Server, then only the first user (who sends REGISTER packet first) is able to register successfully on the server and all the other users after the first one are not able to register on the server. Am I missing something in the configuration file such that this is happening. Please help in this regard. Waiting in anticipation Thanks & Best Regards Saurabh Srivastava

2 1

Roadmap to next major release - v4.2.0
by Daniel-Constantin Mierla 02 Sep '14

02 Sep '14

Hello, during the last devel meeting on IRC, done before the summer, we set the 4.2.0 release time frame for this autumn. As planned by that moment, we have to decide the development freezing date and more accurate milestones at the end of summer, therefore it is about the time. My proposal is to freeze development by end of September 10, 2014, have about one month of testing and release sometime around mid of October. If anyone has constraints to meet this date, just send here another proposal and we can decide together which one fits most of us. This one gives about 2 weeks and a half for new features development. Therefore, if you plan to include a new module (many were announced at the IRC devel meeting, hopefully some are ready), hurry up. It is also a good time to start filling the wiki page with what is new in upcoming release: - http://www.kamailio.org/wiki/features/new-in-devel Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 1

How to add route header
by Senthil K 02 Sep '14

02 Sep '14

Hi All, Kindly check my issue and please provide me your suggestions. In my requirement, I want to add a route header to the INVITE message which is sent out from the Kamailio server, Basically I will be making a call and that INVITE will reach the Kamailio, then the Kamailio will send the INVITE to another node, here at this situation when the Kamailio is sending the INVITE to the next node, I want to add a route header in INVITE. Please let me know how can I hard code a route header to the outgoing INVITE from Kamailio server, Kindly help me Kind Regards SenthilK

2 1

PhonerLite cant login
by Donovan 02 Sep '14

02 Sep '14

Hi, I hope you can help me. My login details are correct but it keeps saying timeout. It just suddenly stopped working. I have made no changes? https://url.odesk.com/_01tJ3v5df4REETK4PV-B3hQtMyc7vTyOQO Best, Don --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

2 1

To mark user as unreachable with nathelper.
by Igor Pavlov 02 Sep '14

02 Sep '14

Hi, colleagues. Is it possible to mark user as unreachable (ex. by setting flag in location table) with nathelper-module when it pinging user? I need something like asterisk's qualifying. ----- Best regards, Igor Pavlov

2 1

Adding a media attribute to SDP
by aft 01 Sep '14

01 Sep '14

Hi, I want to add this media attribute to SDP. a=ptime:<packet time> [1] Its not covered in SDPops module [2] its not added when i use rtpproxy_manage("z<packet size>") to repacketize the rtp stream. So how should i do it? -- -Cheers -Arif

2 4

Wrong ACK to Provider
by Yuriy Gorlichenko 01 Sep '14

01 Sep '14

Hello. I try to provide call scheme: internal client -> asterisk -> Kamailio -> provider -> external endpoint call when I make call I see this: asterisk kamailio provider invite --> invite --> <-- 407 ACK --> invite w/Auth --> <-- 100 <-- 100 <-- 180 <-- 180 <-- 183 <-- 183 <-- 200 <-- 200 ACK --> ACK --> My problem with last ACK, that I send to provider. Provider ignores it, and sends me some OK packets. As resultI can notend session ( answer to BYE 481 - transaction does not exists). I think it is wrong ACK but can not undrtand where I do mistake. Please help me to find it: My invite (with Auth creditans): IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 1606 E...]. .@..R ...6........N0TINVITE sip:12345678900@my.provider.ip:5060 SIP/2.0 Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Max-Forwards: 70 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068> Contact:<provider_username@my.external.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Wed, 27 Aug 2014 22:02:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="provider_username", realm="my.provider.ip", nonce="U/5Wv1P+VZNjFBLf6fwPizgd6iLto5St", uri="sip:12345678900@my.provider.ip:5060", qop=auth, nc=00000001, cnonce="2888860875", response="9f23110471fe9ff751cd55466e70ded2", algorithm=MD5 v=0 o=root 1370647246 1370647246 IN IP4 12.34.56.78 s=Asterisk PBX 12.5.0 c=IN IP4 12.34.56.78 t=0 0 a=ice-lite m=audio 30296 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30297 a=ice-ufrag:p5k92ynl a=ice-pwd:FIOYKt96NlBfEqKsQipUuadUev1g a=candidate:vV3V06Tv Provider trying IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 500 E.........PX6... ..........ySIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1;rport=5068;received=12.34.56.78 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 provider ringing IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1098 E..f......M.6... ........RV.SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip >;party=calling;privacy=off;screen=no provider seesion in progress IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1887 E..... ...,.6... ........g.DSIP/2.0 183 Session Progress Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 742 Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 s=FreeSWITCH c=IN IP4 67.192.253.160 t=0 0 a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD m=audio 27180 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=ssrc:326362635 cnam provider OK IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 2026 E..... ...,.6... ...........SIP/2.0 200 OK Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> Fл2rom: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE SupлЛ o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 s=FreeSWITCH c=л2IN IP4 67.192.253.160 t=0 0 a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD m=audio 27180 RTP/AVP 0 my ACK IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 614 E...]...@... ...6........n.hACK sip:12345678900@my.provider.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.4250e4d315c4aa6697b6d7f70e861b62.0 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK4d28fc11;rport=50600 Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Max-Forwards: 70 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Contact:<provider_username@my.external.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 So after this ACK provider still sends me 200 OK and my server still sends ACK.... tags and call-id always one. Thanks

3 11

Re: [SR-Users] Presence - How to force Kamailio to use one subscription
by Daniel-Constantin Mierla 01 Sep '14

01 Sep '14

Hello, if you have a patch and want to get it in the main repository, send it and if all looks ok, we can commit it -- that will ensure the feature is there with new releases of kamailio. Cheers, Daniel On 01/09/14 10:28, Surendra wrote: > Hi Veerabhadra, > > Kamailio doesn't support this. We had done this by changing the code > logic. > > Regards > surendra > > > > > > > _______________________________________________ > 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 Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

[OT] vim auto-complete plugin for kamailio.cfg
by Daniel-Constantin Mierla 01 Sep '14

01 Sep '14

Hello, I published a vim plugin offering basic autocomplete for kamailio configuration file. It is available via githup at: * https://github.com/miconda/vim-extensions More details about the plugin (including how to install and use) can be read at: * https://github.com/miconda/vim-extensions/blob/master/plugin/kamailio/READM… It was written more as a proof of concept so far, hopefully interested people will join development to bring it to a 'smarter auto-complete' plugin. Right now it relies on custom autocomplete feature from vim -- once activated, as you type a word, by pressing CTRL+x CTRL+u a pop up box presents possible completion by matching the typed part of the word. The dictionaries include module functions and parameters, script variables, core functions and parameters (they were generated with some scripts grepping various files, thus they might not be complete). It is not (yet) aware of context, so it displays all matches (e.g., even when in a route block, it displays core parameters), so again, everyone is welcome to contribute/fork/pull requests to develop the plugin further. The readme lists some other goals as well. Hopefully it is going to be useful at least for some people in the community! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Re: [SR-Users] Presence - How to force Kamailio to use one subscription
by Surendra 01 Sep '14

01 Sep '14

Hi Veerabhadra, Kamailio doesn't support this. We had done this by changing the code logic. Regards surendra

1 0

Presence - How to force Kamailio to use one subscription per user?
by Veerabhara Gundu 30 Aug '14

30 Aug '14

Hello Friends, I have a scenario, where clients are going away and re-registering with new IP and Port numbers. Presence server keep sending NOTIFY messages for both old Subscription and new Subscription for same clients till the old subscription expires. It's unnecessary traffic and processing on both client and server. I am looking for a way to force Kamailio to use one subscription per user similar to mode:04 of location table. If it's not possible, looking for a way to remove subscription from config file when I detect this error.We have an option to remove automatically Subscription when NOTIFY failed, but this is not working for TCP based clients. In TCP mode, when Kamailio not event trying to send NOTIFY if Kamailio already knows that TCP connection is broken and replying with 500 error. Any alternative suggestion also welcome. Thanks, Veera Thanks, VG

1 3

What does "::" operator do in statements related to hashtable?
by aft 30 Aug '14

30 Aug '14

Hi, >From kamailio documentation, the usage of hashtable is given as : modparam("htable", "htable", "a=>size=4;") ... $sht(a=>test) = 1; $sht(a=>$ci::srcip) = $si; I get it that in the first statement, a is the hashtable, a new key-value pair is added to one of its empty bucket (test,1). What is happening in the second statement? -- -Cheers -Arif

3 3

Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS
by Alex Villacís Lasso 29 Aug '14

29 Aug '14

As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration. In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost): REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad>";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0 Asterisk answers this through UDP, and Kamailio forwards it through the websocket: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0 Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket: NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws> Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Since I have not implemented handling of voicemail indications, the browser answers this: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;tag=ggu5etber9 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0 After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws . I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?

2 5

How to use the ip parameter in rtpproxy_manage()
by Sebastian Damm 29 Aug '14

29 Aug '14

Hi, I have the following setup: UAC ---> LB ---> Proxy ---> GW In NAT scenarios the loadbalancer detects it, but the proxy communicates with the RTP proxy. I want to send the original caller IP as detected by the loadbalancer (and transported to the proxy) to the RTP proxy. As far as I understand the documentation of the rtpproxy module, I could call rtpproxy_manage with a second parameter, indicating which IP address should be sent to the RTP proxy. I tried sending a pseudovariable, both within quotes or not, or even a static string. But when I look at those messages sent to the RTP proxy, there's always the IP from where the SIP packet was received (the loadbalancer IP) in the request. Previously we used the "r" parameter, sending the original IP from the SDP to the proxy. But due to strange behavior of some UACs, we want to get rid of that. Do I misunderstand the second parameter? This is what the documentation says: * ip_address* - new SDP IP address. Unfortunately, I couldn't find any example using this. Best Regards, Sebastian

2 3

Help analysing segmentation fault
by Charles Chance 29 Aug '14

29 Aug '14

Hi All, I wonder if some one could help me to diagnose a recurring issue? It happens at random times/intervals and under varying load. But always, just before the time of crash, I see the same critical error in log: CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0xb0632134 [1367:5814] with clid ' 0695dd7a346188dd24e7520e6c01092c(a)sip.sipcentric.com' and tags 'as77c89620' '' Analysing the core dump reveals: Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 128 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 0x081a737c in parse_uri (buf=0x3a70006e <Address 0x3a70006e out of bounds>, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389 389 scheme=buf[0]+(buf[1]<<8)+(buf[2]<<16)+(buf[3]<<24); (gdb) frame 1 #1 0x008fe5dd in dialog_publish (state=0x903f37 "Trying", ruri=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275 275 if (parse_uri(ruri->s, ruri->len, &ruri_uri) < 0) { (gdb) p *ruri $1 = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275} (gdb) up #2 0x008ff277 in dialog_publish_multi (state=0x903f37 "Trying", ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:387 387 dialog_publish(state,&(ruris->s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck); (gdb) p *ruris $2 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} (gdb) up #3 0x0090187a in __dialog_created (dlg=0xb0632134, type=2, _params=0x6db064) at pua_dialoginfo.c:470 470 dialog_publish_multi("Trying", dlginfo->pubruris_caller, &(dlg->from_uri), (include_req_uri)?&(dlg->req_uri):&(dlg->to_uri), &(dlg->callid), 1, dlginfo->lifetime, 0, 0, 0, 0, send_publish_flag==-1?1:0); (gdb) p *dlginfo->pubruris_caller $3 = {s = {s = 0x31590014 <Address 0x31590014 out of bounds>, len = 275}, next = 0xb0b5fd00} (gdb) p *dlginfo->pubruris_caller->next $4 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} In config, for pua_dialoginfo we are enabling the option "use_pubruri_avps" and setting "pubruri_caller_avp" and "pubruri_callee_avp" accordingly. Therefore, in pua_dialoginfo.c it is using get_str_list() function to set dlginfo->pubruris_caller from the avp. Could this be some race condition or something completely different? Thanks in advance, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

3 7

Add websocket header to SIP
by Kamrul Khan 28 Aug '14

28 Aug '14

Hi, I want to append the origin header of my incoming websockets to the SIP INVITES. I tried something like the below but didnt work: if (ws_handle_handshake()) { if(method=="INVITE"){ insert_hf("Origin: $hdr(Origin)"); } exit; } Is there any other way to do it ?

2 3

[Kamailio-4.1.5] [/etc/init.d/kamailio/start] Error
by Djamel Bahamid 28 Aug '14

28 Aug '14

Hello, When I try to execute the /etc/init.d/kamailio start, I've this error : [root@gtk:~][Mon 18/08/2014 16:21:58] # /etc/init.d/kamailio start Not starting Kamailio SIP server: invalid configuration file! failed failed ERROR: loading config file(/usr/local/etc/kamaiilio/kamailio.cfg): No such file or directory failed failed Thank you for your help, Best Regards, Djamel BAHAMID.

3 5

rfc: distributing dialog profiles
by Daniel-Constantin Mierla 28 Aug '14

28 Aug '14

Hello, if you are on sr-dev list, you should have seen the last commits to dialog module, which add support for remote profiles. As most of you know (or at least the users of dialog module), profiles are used to group active calls, their main purposes being counting dialogs belonging to a profile for implementing active call limits (e.g., all calls in the system, call from/to a specif user/address). It was working on a single Kamailio instance, but one could use other meanings to interact with other kamailio instance to get the number of calls across multiple instances. The purpose of remote profiles is to make the whole process easier and more transparent for config file, simplifying the maintenance. A remote profile is an item stored in local profile list not bound to a local dialog. At this moment, dialog module in master branch exports a function dlg_remote_profile() that allows to add/remove a remote profile. The next step is to build the notification system between kamailio instances. There are couple of options, as well as questions, that I want to discuss before going to implement the mechanism. There are two major aspects to care of: A) Distribution, I thought of two options for it: 1) using dmq module to publish the operations done with local profiles (add/remove) -- this seems to be the natural choice right now, Charles Chance is also willing to put effort in this direction This will be coded inside the module, so config won't be affected much (eventually some extra parameters of dialog flags). 2) add event routes for adding/removing a local profile. The attributes of the profile can be made available via some PV, then in config use what so ever option (e.g., http query, json rpc, etc) for distributing the information. This will require changes in config file and might not be trivial for new users B) Persistence, with the options: 1) store remote profiles in local database. It is good for restart, but if there is big downtime, then some notifications can be lost and the list of remote profiles might be inaccurate 2) keep in memory only and at startup send a 'request-for-update' to the other instances -- these instances will send the list of their local profiles. It can take a bit between request-for-update and receiving data from the other servers -- during this interval, the checks of overall call limits might not give the desired answer Of course, there can be enhancements/combinations, but it has to be analysed if the benefit worth the complexity. For 1), store also the IP of each remote profile and then interrogate for each profile to see if its still active. For 2), add a mechanism to reject new calls until the request-for-update is completed. Anyone having other ideas or comments to the above? Note that each profile got a unique id, so if you upgrade your dialog module, profiles for existing active calls will not be loaded. Also, the code was barely tested, report here any issue you have with dialog from master branch. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

5 9

Calls to devices behind NAT not working
by Adnan 28 Aug '14

28 Aug '14

Hi all, I'm trying to configure a Kamailio server, and am having trouble getting devices behind NAT receive calls. Looking at a SIP trace, nothing is being received by the end point, even though Kamailio seems to be sending out a response. I've opened up my firewall rules, so that shouldn't be an issue. If I call another endpoint which is behind the same NAT, it works (although no audio), and looking at the trace, the SIP messaging is coming from the Kamailio server (ie. not the other endpoint). I've even tried using the default kamailio-advanced.cfg & kamailio-basic.cfg both of which didn't work either. I've attached the current config I'm working on, if someone could please have a look through it, and see where I'm going wrong I'd highly appreciate it. Thanks Adnan

1 0

Minimal configuration without database support
by Eugene Prokopiev 27 Aug '14

27 Aug '14

Hi, Is it possible to run kamailio without database support (even dbtext)? Where can I find minimal configuration example with minimal modules set only for forwarding any requests/responses from one ip address to another one? -- WBR, Eugene Prokopiev

5 23

Understanding hashtable usage
by aft 27 Aug '14

27 Aug '14

Hi, I've got a piece of config code which should solve my problem, but i'm not getting the code. It involves hashtable module. Here it goes : modparam("htable", "htable", "a=>size=8;autoexpire=400") modparam("htable", "htable", "b=>size=8;autoexpire=31") Ok we have two hashtable , a and b, both has a "size" of 8. Then, else if(status == "503") { if($sht(a=>alarm::503) == $null) $sht(a=>alarm::503)=0; $sht(a=>alarm::$si::503) = $sht(a=>alarm::$si::503) + 1; } Here i get lost. what's "a=>alarm::503" and what's "a=>alarm::$si::503"? What's there difference? Next the code goes like :(its supposed be pseduocode) $elem = the size of the sht array. while($i < $elem) { $src_ip = $sht(source_ip_array)[$i]; $i = $i +1; sql_query("cb", "INSERT INTO alarm_data (create_date, type, total, description, source_ip) VALUES(NOW(), 'Too Many 503', $sht(a=>alarm::$src_ip::503), 'service unavailable', '$src_ip')"); } Now i have couple of problems with this code [1] what should be the value of $elem? [2] $sht(source_ip_array)[$i]? what should be the value of source_ip_array? The thing i'm trying to do is, when a 503 (Service unavailable) comes to kamailio, I want to insert it into a database which records this. -- -Cheers -Arif

1 0

match_dialog not handling BYE
by Satish Patel 26 Aug '14

26 Aug '14

I really want to use top hide using dialog module but having very difficult time.. [UA]------[Opensips]-----[Asterisk] I want to hide UA so Asterisk doesn't know its IP. I have following config, Please correct me if anything wrong in my config, Problem is if callee whoever connected to asterisk hangup call it doesn't not know where to send BYE because of top hiding remove all return routes.. i am surprise how it is working with other folks? something must be wrong in my config then.. if (loose_route() || match_dialog()) { if ($DLG_status==NULL) { xlog(" cannot match request to a dialog \n"); # something wrong - might want to drop such requests } if (is_method("BYE") { setflag(ACC_DO); # do accounting ... setflag(ACC_FAILED); # ... even if the transaction fails acc_aaa_request("Internally generated BYE"); end_media_session(); } else if (is_method("INVITE")) { # even if in most of the cases is useless, do RR for # re-INVITEs alos, as some buggy clients do change route set # during the dialog. record_route(); } .... .... if ( uri=~"^sip:[0-9]*@.*") { uac_replace_from("sip:4545@65.xxx.xxx.xxx"); t_on_failure("3"); resetflag(7); topology_hiding(); t_relay( "udp:65.xxx.xxx.xxx" );

1 0

Not able to start tls on port 5061
by Debojit Kakoti 26 Aug '14

26 Aug '14

I am getting error INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... -- Debojit Kakoti Mobile : +91 9650654870 Email : debojkakoti88(a)gmail.com Skype : deboj1988

2 1

Re: [SR-Users] sr-users Digest, Vol 111, Issue 21
by Rakesh Sharma 23 Aug '14

23 Aug '14

How to configure 3 kamailio on three different node, and they work as a single kamailio proxy. which module or method need to implement. On Fri, Aug 22, 2014 at 3:30 PM, <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. force_send_socket in event route tm:local-request > (Kristian F. H?gh) > 2. Re: SIP Trunk (Satish Patel) > 3. Re: SIP Trunk (Yuriy Gorlichenko) > 4. Re: CNXCC curious case (Carlos Ruiz D?az) > 5. Kamailio script editor (Grant Bagdasarian) > 6. Re: Kamailio script editor (Narsay, Deep) > 7. Re: Kamailio script editor (Grant Bagdasarian) > 8. Re: Kamailio script editor (Alex Balashov) > 9. Re: force_send_socket in event route tm:local-request > (Daniel-Constantin Mierla) > 10. Re: Kamailio script editor (Daniel-Constantin Mierla) > 11. switch kamailio to domain (Mihai Marin) > 12. Re: SIP Trunk (Satish Patel) > 13. Re: Kamailio script editor (M?ns Nilsson) > 14. Re: force_send_socket in event route tm:local-request > (Kristian Frederik H?gh) > 15. How to uniquely identify SIP WS / WSS endpoint (Muhammad Shahzad) > 16. nat_traversal vs nathelper for mediaproxy (Satish Patel) > 17. Re: switch kamailio to domain (Daniel-Constantin Mierla) > 18. Re: switch kamailio to domain (Mihai Marin) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 21 Aug 2014 12:00:49 +0200 > From: Kristian F. H?gh <kfh(a)uni-tel.dk> > To: <sr-users(a)lists.sip-router.org> > Subject: [SR-Users] force_send_socket in event route tm:local-request > Message-ID: <25156162.LDl4apoH3L@kfh-laptop> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Our kamailio 4.0.6 proxy uses pua/pua_dialoginfo and an external presence > server. > The PUBLISH requests generated by pua* appear in tm:local-request event > route. > > Before sending, we call set_advertised_address and force_send_socket. > Still the requests have the IP address from the first "listen=" as source > and Via address. > The requests are send out using the correct interface and no errors are > logged. > > When forwarding SUBSCRIBE requests received from clients, > force_send_socket before t_relay_to_udp do work as expected. > > mhomed=1 solves the issue, but I prefer force_send_socket. > > Regards, > Kristian H?gh > Uni-tel > > > > > ------------------------------ > > Message: 2 > Date: Thu, 21 Aug 2014 06:43:15 -0400 > From: Satish Patel <satish.txt(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Cc: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] SIP Trunk > Message-ID: <A3BD87DA-AE2A-4E52-B70A-031A875DF111(a)gmail.com> > Content-Type: text/plain; charset=us-ascii > > I will give it a try again today, can you please make sure my t_relay() > syntax is correct? > > So t_relay will rewrite my host past right and send call to trunk. > > While ago I was using rewritehost() function but I think it's not working > with UAC registrant module. > > > Sent from my iPhone > > On Aug 21, 2014, at 3:53 AM, Daniel Tryba <daniel(a)pocos.nl> wrote: > > > On Thursday 21 August 2014 05:56:46 Satish Patel wrote: > >> if ( $rU =~ "sip:1[0-9]@*") { > > > > Try $ru instead, $rU only contains the dialled "number". So > > $ru =~ "sip:1[0-9]@*" > > or > > $rU =~ "1[0-9]" > > > > But note the regexp, that only matches the exact numbers 10 to 19, if > you are > > trying to match prefixes you are doing it wrong (tm). > > > > _______________________________________________ > > 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 > > > > ------------------------------ > > Message: 3 > Date: Thu, 21 Aug 2014 15:37:38 +0400 > From: Yuriy Gorlichenko <ovoshlook(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] SIP Trunk > Message-ID: > <CABSP_VczfN-giH5ghuT2A9Gnqyk_+VNa= > XFtX3tAaWDqM4kZTg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > rewritehost() sucessfully work with UAC. But As I know > 1) It statless function > 2) It read only string argumetns, and do not read variables > > > 2014-08-21 14:43 GMT+04:00 Satish Patel <satish.txt(a)gmail.com>: > > > I will give it a try again today, can you please make sure my t_relay() > > syntax is correct? > > > > So t_relay will rewrite my host past right and send call to trunk. > > > > While ago I was using rewritehost() function but I think it's not working > > with UAC registrant module. > > > > > > Sent from my iPhone > > > > On Aug 21, 2014, at 3:53 AM, Daniel Tryba <daniel(a)pocos.nl> wrote: > > > > > On Thursday 21 August 2014 05:56:46 Satish Patel wrote: > > >> if ( $rU =~ "sip:1[0-9]@*") { > > > > > > Try $ru instead, $rU only contains the dialled "number". So > > > $ru =~ "sip:1[0-9]@*" > > > or > > > $rU =~ "1[0-9]" > > > > > > But note the regexp, that only matches the exact numbers 10 to 19, if > > you are > > > trying to match prefixes you are doing it wrong (tm). > > > > > > _______________________________________________ > > > 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 > > > > _______________________________________________ > > 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

switch kamailio to domain
by Mihai Marin 22 Aug '14

22 Aug '14

Hello Sirs, The RLS module is eating my life. I just managed to make it work 2 months ago and now is not working anymore after I tried to put the kamailio under domain (until now it was simple ip). I don't understand why is not working but I saw a big difference that I don't understand: My environment: - the domain "domain.com" is pointing to the web server - kamailio is hosted under sip.domain.com - DNS has domain.com A record for web server's ip, sip.domain.com A record for kamailio's ip and SRV record for _sip._tcp.domain.com poting to sip.domain.dom Problem: kamailio presence is not working anymore and I think, the error is: grep_sock_info - checking if host==us: 11==14 && [domain.com] == [xx.xx.xx.xx witch is the sip.domain.com IP] If, I change domain.com in DNS to point on the same ip as sip.domain.com RLS is working. This, I can't understand. Can anyone explain me why is not working using this configuration? Also, the RLS is working if i change my domain to be sip.domain.com (ex: sip:user@sip.domain.com). Thank you. Best regards, Mihai M

2 3

force_send_socket in event route tm:local-request
by Kristian F. Høgh 21 Aug '14

21 Aug '14

Hi, Our kamailio 4.0.6 proxy uses pua/pua_dialoginfo and an external presence server. The PUBLISH requests generated by pua* appear in tm:local-request event route. Before sending, we call set_advertised_address and force_send_socket. Still the requests have the IP address from the first "listen=" as source and Via address. The requests are send out using the correct interface and no errors are logged. When forwarding SUBSCRIBE requests received from clients, force_send_socket before t_relay_to_udp do work as expected. mhomed=1 solves the issue, but I prefer force_send_socket. Regards, Kristian Høgh Uni-tel

3 2

Kamailio script editor
by Grant Bagdasarian 21 Aug '14

21 Aug '14

Hello, What editor do most of you use for editing the kamailio config script? I'm using notepad, which works fine, but sometimes I wish I had something like text coloring and other useful stuff found in most scripting editors. Are there perhaps any plans for developing an editor for the kamailio config script? With code completion, etc. Regards, Grant

5 5

SIP Trunk
by Satish Patel 21 Aug '14

21 Aug '14

We have setup Kamailio front and SIP Proxy, Now i want to Trunk it with other SIP provide they gave me IP, Username/Password. How do i configure username/password on Kamailio SIP Proxy?

4 13

CNXCC curious case
by Kelvin Chua 21 Aug '14

21 Aug '14

I am using cnxcc_set_max_channels and set the max channels to 1 after the 1st call goes through and hangs up, the 2nd one would be blocked even if there are no more active dialogs. could it be a dialog state issue? Kelvin Chua

2 3

Re: [SR-Users] automatic string to int conversion for rtpengine flags failed - rtpproxy_offer() does not change sdp when called in branch_route
by Olli Heiskanen 20 Aug '14

20 Aug '14

Hi, The source for this string to int conversion error was found, it was just a minor glitch in an if statement! Man, I feel stupid... Anyways, the problem about calls not going through still persists. This I located to the rtpengine_offer() call in a branch route. The sdp is not changed and this results in Kamailio forking away until it can't. In my test case there is a webrtc client calling another webrtc client. Here I use Kamailio 4.1.5, rtpengine and rtpproxy-ng module. My branch route looks like this: branch_route[MANAGE_RTPENGINE_BRANCH] { xlog("L_INFO", "MANAGE_RTPENGINE_BRANCH: new branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "MANAGE_RTPENGINE_BRANCH: rtpengine_offer_flags = $avp(rtpengine_offer_flags) and mline = $avp(mline)"); if ($avp(rtpengine_offer_flags) != "") { xlog("L_INFO","MANAGE_RTPENGINE_BRANCH: rtpengine_offer with $avp(rtpengine_offer_flags)"); rtpproxy_offer($avp(rtpengine_offer_flags)); t_on_reply("RTPPROXY_REPLY"); } else { xlog("L_ERR","MANAGE_RTPENGINE_BRANCH but rtp_offer_flags is EMPTY! "); } } And this is what I get in the kamailio log: Aug 20 21:47:58 u363id562 kamailio[28376]: ALERT: <script>: UA_FAILURE: SAVPF found, setting flags using RTP/AVP in offer and RTP/SAVPF in answer Aug 20 21:47:58 u363id562 kamailio[28376]: INFO: <script>: UA_FAILURE: hooked to RTPPROXY_REPLY reply_route and MANAGE_RTPENGINE_BRANCH branch_route, offer flags: froc-sp Aug 20 21:47:58 u363id562 kamailio[28376]: INFO: <script>: RELAY: hooked to manage_failure, now relaying... Aug 20 21:47:58 u363id562 kamailio[28376]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: new branch [1] to sip:661@testers.com Aug 20 21:47:58 u363id562 kamailio[28376]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: rtpengine_offer_flags = froc-sp and mline = m=audio 49466 RTP/SAVPF 111 103 104 0 8 106 105 13 126 Aug 20 21:47:58 u363id562 kamailio[28376]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: rtpengine_offer with froc-sp Aug 20 21:47:58 u363id562 kamailio[28379]: ALERT: <script>: UA_FAILURE: SAVPF found, setting flags using RTP/AVP in offer and RTP/SAVPF in answer Aug 20 21:47:58 u363id562 kamailio[28379]: INFO: <script>: UA_FAILURE: hooked to RTPPROXY_REPLY reply_route and MANAGE_RTPENGINE_BRANCH branch_route, offer flags: froc-sp Aug 20 21:47:58 u363id562 kamailio[28379]: INFO: <script>: RELAY: hooked to manage_failure, now relaying... Aug 20 21:47:58 u363id562 kamailio[28379]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: new branch [2] to sip:661@testers.com Aug 20 21:47:58 u363id562 kamailio[28379]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: rtpengine_offer_flags = froc-sp and mline = m=audio 49466 RTP/SAVPF 111 103 104 0 8 106 105 13 126 Aug 20 21:47:58 u363id562 kamailio[28379]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: rtpengine_offer with froc-sp ... and so on And this is what's happening in syslog between catching a 488, the rtpengine_offer call and the next 488: Here 1.1.1.1 is the public ip of my Kamailio machine and 2.2.2.2 is the ip behind which my clients reside. I haven't been able to spot the problem, maybe You will. Aug 20 22:01:08 u363id562 kamailio[28610]: ALERT: <script>: UA_FAILURE: SAVPF found, setting flags using RTP/AVP in offer and RTP/SAVPF in answer Aug 20 22:01:08 u363id562 kamailio[28610]: INFO: <script>: UA_FAILURE: hooked to RTPPROXY_REPLY reply_route and MANAGE_RTPENGINE_BRANCH branch_route, offer flags: froc-sp Aug 20 22:01:08 u363id562 kamailio[28610]: INFO: <script>: RELAY: hooked to manage_failure, now relaying... Aug 20 22:01:08 u363id562 kamailio[28610]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: new branch [1] to sip:661@testers.com Aug 20 22:01:08 u363id562 kamailio[28610]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: rtpengine_offer_flags = froc-sp and mline = m=audio 55155 RTP/SAVPF 111 103 104 0 8 106 105 13 126 Aug 20 22:01:08 u363id562 kamailio[28610]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: rtpengine_offer with froc-sp Aug 20 22:01:08 u363id562 rtpengine[32233]: Got valid command from 127.0.0.1:57017: offer - { "sdp": "v=0#015#012o=- 3706977575786663946 2 IN IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-semantic: WMS C4FrYLUUCeWV8DP6OQlzG3XOBCZrnmVlTWuz#015#012m=audio 55155 RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 2.2.2.2#015#012a=rtcp:55155 IN IP4 2.2.2.2#015#012a=candidate:2999745851 1 udp 2122260223 192.168.56.1 55154 typ host generation 0#015#012a=candidate:2999745851 2 udp 2122260223 192.168.56.1 55154 typ host generation 0#015#012a=candidate:3350409123 ... Aug 20 22:01:08 u363id562 rtpengine[32233]: ... 1 udp 2122194687 192.168.0.101 55155 typ host generation 0#015#012a=candidate:3350409123 2 udp 2122194687 192.168.0.101 55155 typ host generation 0#015#012a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0#015#012a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0#015#012a=candidate:2301678419 1 tcp 1518214911 192.168.0.101 0 typ host generation 0#015#012a=candidate:2301678419 2 tcp 1518214911 192.168.0.101 0 typ host generation 0#015#012a=candidate:1190865175 1 udp 1685987071 91. ... Aug 20 22:01:08 u363id562 rtpengine[32233]: ... 145.67.22 55155 typ srflx raddr 192.168.0.101 rport 55155 generation 0#015#012a=candidate:1190865175 2 udp 1685987071 2.2.2.2 55155 typ srflx raddr 192.168.0.101 rport 55155 generation 0#015#012a=ice-ufrag:A/jaPavOgPBuvQLE#015#012a=ice-pwd:cDlvtQt87wG8uuOhrEmhg9dX#015#012a=ice-options:google-ice#015#012a=fingerprint:sha-256 CF:30:A7:7F:71:11:D4:5E:B0:E7:E3:F9:D8:C2:F4:60:2A:D0:76:46:F8:3A:97:01:C9:0C:5A:F7:B8:7D:C1:43#015#012a=setup:actpass#015#012a=mid:audio#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=extmap:3 http: ... Aug 20 22:01:08 u363id562 rtpengine[32233]: ... // www.webrtc.org/experiments/rtp-hdrext/abs-send-time#015#012a=sendrecv#015#0… opus/48000/2#015#012a=fmtp:111 minptime=10#015#012a=rtpmap:103 ISAC/16000#015#012a=rtpmap:104 ISAC/32000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:106 CN/32000#015#012a=rtpmap:105 CN/16000#015#012a=rtpmap:13 CN/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=maxptime:60#015#012a=ssrc:772277070 cname:BwsjKefqgs/g3RSF#015#012a=ssrc:772277070 msid:C4FrYLUUCeWV8DP6OQlzG3XOBCZrnmVlTWuz b7ec3c91-c3a2-4ede-adbf-ada31cb9504f#015#012a=ssrc:772277070 msla ... Aug 20 22:01:08 u363id562 rtpengine[32233]: ... bel:C4FrYLUUCeWV8DP6OQlzG3XOBCZrnmVlTWuz#015#012a=ssrc:772277070 label:b7ec3c91-c3a2-4ede-adbf-ada31cb9504f#015#012", "ICE": "remove", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "ug2htprt2erjsat1e6rc", "received-from": [ "IP4", "2.2.2.2" ], "from-tag": "vf9jdb7362", "command": "offer" } Aug 20 22:01:08 u363id562 rtpengine[32233]: Unknown flag encountered: 'force' Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] Creating new call Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] Returning to SIP proxy: d3:sdp762:v=0#015#012o=- 3706977575786663946 2 IN IP4 1.1.1.1#015#012s=-#015#012t=0 0#015#012a=msid-semantic: WMS C4FrYLUUCeWV8DP6OQlzG3XOBCZrnmVlTWuz#015#012m=audio 10624 RTP/AVP 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 1.1.1.1#015#012a=rtpmap:111 opus/48000/2#015#012a=fmtp:111 minptime=10#015#012a=rtpmap:103 ISAC/16000#015#012a=rtpmap:104 ISAC/32000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:106 CN/32000#015#012a=rtpmap:105 CN/16000#015#012a=rtpmap:13 CN/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=maxptime:60#015#012a=ss ... Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] ... rc:772277070 cname:BwsjKefqgs/g3RSF#015#012a=ssrc:772277070 msid:C4FrYLUUCeWV8DP6OQlzG3XOBCZrnmVlTWuz b7ec3c91-c3a2-4ede-adbf-ada31cb9504f#015#012a=ssrc:772277070 mslabel:C4FrYLUUCeWV8DP6OQlzG3XOBCZrnmVlTWuz#015#012a=ssrc:772277070 label:b7ec3c91-c3a2-4ede-adbf-ada31cb9504f#015#012a=sendrecv#015#012a=rtcp:10625#015#012a=rtcp-mux#015#0126:result2:oke Aug 20 22:01:08 u363id562 rtpengine[32233]: Got valid command from 127.0.0.1:49781: delete - { "call-id": "ug2htprt2erjsat1e6rc", "received-from": [ "IP4", "2.2.2.2" ], "from-tag": "vf9jdb7362", "command": "delete" } Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] Deleting full call Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] Final packet stats: Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] --- Tag '', created 0:00 ago, in dialogue with '(none)' Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] ------ Media #1, port 10626 <> 2.2.2.2:55155, 0 p, 0 b, 0 e Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] --- Tag 'vf9jdb7362', created 0:00 ago, in dialogue with '' Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] ------ Media #1, port 10624 <> [::]:0 , 0 p, 0 b, 0 e Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] Returning to SIP proxy: d7:createdi1408557668e11:last signali1408557668e4:tagsd10:vf9jdb7362d3:tag10:vf9jdb73627:createdi1408557668e16:in dialogue with0:6:mediasld5:indexi1e4:type5:audio8:protocol7:RTP/AVP7:streamsld10:local porti10624e8:endpointd6:family4:IPv67:address2:::4:porti0ee19:advertised endpointd6:family4:IPv67:address2:::4:porti0ee11:last packeti1408557668e5:flagsl3:RTP4:RTCPe5:statsd7:packetsi0e5:bytesi0e6:errorsi0eeed10:local porti10625e8:endpointd6:family4:IPv67:address2:::4:porti0 ... Aug 20 22:01:08 u363id562 rtpengine[32233]: [ug2htprt2erjsat1e6rc] ... ee19:advertised endpointd6:family4:IPv67:address2:::4:porti0ee11:last packeti1408557668e5:flagsl4:RTCP13:fallback RTCPe5:statsd7:packetsi0e5:bytesi0e6:errorsi0eeee5:flagsl8:rtcp-muxeeeee6:totalsd3:RTPd7:packetsi0e5:bytesi0e6:errorsi0ee4:RTCPd7:packetsi0e5:bytesi0e6:errorsi0eee6:result2:oke Aug 20 22:01:08 u363id562 kamailio[28613]: ALERT: <script>: UA_FAILURE: SAVPF found, setting flags using RTP/AVP in offer and RTP/SAVPF in answer Can you guys spot the problem from these logs? I've been stuck with this for a while, I must be doing something wrong but don't know what... cheers, Olli 2014-08-19 11:13 GMT+03:00 Olli Heiskanen <ohjelmistoarkkitehti(a)gmail.com>: > Hi, > > A little follow-up on this: > > The problem only happens when I call rtpengine_offer() inside a > branch_route. If I call rtpengine_offer() in the failure_route (after 488) > this conversion error does not happen, but then I get the double sdp issue > indicated here: > http://lists.sip-router.org/pipermail/sr-dev/2014-July/024507.html > > The same problem occurs with Kamailio 4.1.5 and rtpproxy-ng. My current > setup uses Kamailio 4.2 (devel) and rtpengine. > > If anyone has a way around this, I'd appreciate very much to hear it! This > prevents any calls from being setup. > > cheers, > Olli > > > > 2014-08-15 18:40 GMT+03:00 Olli Heiskanen <ohjelmistoarkkitehti(a)gmail.com> > : > > Hello, >> >> As outcome to my earlier sdp/rtp challenges I've upgraded my Asterisk >> version to 11.11.0 and still use a realtime integration with Kamailio. Now >> I face a somewhat different problem. With my setup I also changed from >> jssip client to a sip.js client in my websocket implementation. I cloned >> the latest rtpengine from git today. >> >> I had to revert my Asterisk settings a bit, Asterisk was taking over with >> sdp handling, I don't know if this is relevant but that's why I got calls >> seemingly working before. >> >> When getting the 488 Not Acceptable, I arm a branch route and call >> rtpengine_offer there. When trying to call rtpengine_offer, I get the >> following log: >> (first I print the rtpengine_offer_flags to make sure what is passed to >> the function. >> >> Aug 15 15:04:16 u363id562 kamailio[32178]: INFO: <script>: >> MANAGE_RTPENGINE_BRANCH: rtpengine_offer_flags = rtcp-mux-demux >> trust-address replace-origin replace-session-connection ICE=remove RTP/AVP >> Aug 15 15:04:16 u363id562 kamailio[32178]: WARNING: <core> >> [rvalue.c:1016]: rval_get_int(): automatic string to int conversion for >> "rtcp-mux-demux trust-address replace-origin replace-session-connection >> ICE=remove RTP/AVP" failed >> Aug 15 15:04:16 u363id562 kamailio[32178]: WARNING: <core> >> [rvalue.c:1920]: rval_expr_eval_int(): rval expression conversion to int >> failed (1128,32-1128,32) >> Aug 15 15:04:16 u363id562 rtpengine[32159]: Got valid command from >> 127.0.0.1:44292: delete - { "call-id": "k7bft3u75p5e42ib039r", >> "received-from": [ "IP4", "client_public_address" ], "from-tag": >> "74dovi97bi", "command": "delete" } >> Aug 15 15:04:16 u363id562 rtpengine[32159]: [k7bft3u75p5e42ib039r] >> Call-ID to delete not found >> Aug 15 15:04:16 u363id562 rtpengine[32159]: Returning to SIP proxy: >> d7:warning38:Call-ID not found or tags didn't match6:result2:oke >> >> After this I see another 488 and the loop swirls on until Kamailio runs >> out of forking capacity. Can You guys explain why is this happening? >> >> I set the variable like this: >> $avp(rtpengine_offer_flags) = "rtcp-mux-demux trust-address >> replace-origin replace-session-connection ICE=remove RTP/AVP"; >> >> cheers, >> Olli >> > >

1 0

Generate SIP MESSAGE in Kamailio
by Grant Bagdasarian 20 Aug '14

20 Aug '14

Hello, Is it possible for Kamailio to generate a SIP MESSAGE message when receiving a HTTP request? Take the content of the HTTP message and form it into a SIP MESSAGE and route it to a certain destination? I can't find a module that does this, or I'm just blind.. :) Regards, Grant

2 2

UAC "credential" username, password and realm from DB
by Yuriy Gorlichenko 20 Aug '14

20 Aug '14

Hello. I suceesfully authenticate some accounts of providers from UAc using DB. When I try to Call to any provider it responds me 407 or 401 reply. To Send to It new INVITE with auth Creditans I use UAC_AUTH() with configured modparams: modparam("uac","credential","username:domain:password") modparam("uac","auth_realm_avp","$avp(i:10)") modparam("uac","auth_username_avp","$avp(i:11)") modparam("uac","auth_password_avp","$avp(i:12)") I think that avp`s used to get varibles from DB, but at test it is not write. So my question- how to get this varibles from DB (uacreg table) Thanks

2 2

407 response from provider as failure route
by Yuriy Gorlichenko 19 Aug '14

19 Aug '14

Hi, I want to realise scheme: Asterisk->Kamailo->provider I register providers accounts at Kamailio using UAC module. When Generates call form Asterisk through Kamailio to provider, provider sends back 407 response. This reponse mut be handled by KAmailio through failure_route using uac_auth(). I try to catch this response at onreply_route by status and route this response to failure route (t_in_failure("MY_FAILURE_ROUTE"))but Kamailio ignores t_on_failure and forvard this pacjet to asterisk. So My question is: How to handle this response at kamailio and not forward to asterisk? This is invite that forwards to provider: INVITE sip:11234567890@my.provider.com:5060 SIP/2.0 Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as57a2d34c;lr=on> Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK8544.480e25d0e6a328f5f455233d808cda80.0 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK778521b0;rport=50600 Max-Forwards: 70 From: "John" <sip:my_provider_acc@my.provider.com:50600>;tag=as57a2d34c To: <sip:11234567890@my.provider.com:5068> Contact:<my_provider_acc@my.kamailio.com:5068> Call-ID: 3d8d6c357d69cdd51633b1c125729f44@10.0.1.6:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.4.0 Date: Sat, 16 Aug 2014 13:18:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 540 v=0 o=root 1941643043 1941643043 IN my.kamailio.com s=Asterisk PBX 12.4.0 c=IN IP4 my.kamailio.com t=0 0 a=ice-lite m=audio 30108 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30109 a=ice-ufrag:jOv9GFqq a=ice-pwd:wuXGbJ3ZI7MfwM6kwto78s8reEyU a=candidate:lzqT8la5i9wkzQzB 1 UDP 2130706431 my.kamailio.com 30108 typ host a=candidate:lzqT8la5i9wkzQzB 2 UDP 2130706430 my.kamailio.com 30109 typ host

2 2

Does rtpengine properly handle FQDN in SDP Origin?
by Anthony Messina 19 Aug '14

19 Aug '14

I've begun playing with rtpengine (git e0957d1) a little and in my testing: CSipSimple <-> Kamailio/rtpengine <-> Asterisk 12/13 I see the following error when parsing SDP from the Asterisk side, possibly related to the use of the FQDN in the Origin: Got valid command from 127.0.0.1:44407: answer - { "sdp": "v=0 o=- 3617462201 3617462204 IN IP4 hostname.example.com s=Asterisk <snip> Error parsing SDP at offset 5: Error parsing o= line Protocol error in packet from 127.0.0.1:44407: Failed to parse SDP [d3:sdp353:v=0 o=- 3617462201 3617462204 IN IP4 hostname.example.com s=Asterisk <snip> It was brought up here, https://issues.asterisk.org/jira/browse/ASTERISK-23994 and it appears that a FQDN should be supported in the SDP Origin. Is this the case, or might it be something else? -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 2

event_route xhttp:request
by Kamrul Khan 19 Aug '14

19 Aug '14

i am trying to connect to my kamailio through websocket. I added the below code in my config file: event_route[xhttp:request] { xlog("websocket connection request"); set_reply_close(); set_reply_no_connect(); if (ws_handle_handshake()) { exit; } xhttp_reply("404", "Not found", "", "");} However, when I try to connect from my web phone, It shows : Aug 15 15:22:31 ubuntu /usr/local/kamailio-devel//sbin/kamailio[17202]: WARNING: xhttp [xhttp_mod.c:133]: mod_init(): event_route[xhttp:request] is empty Can someone please tell me what wrong im doing ?

2 9

rtpengine symmetric RTP behavior
by Narsay, Deep 19 Aug '14

19 Aug '14

Hello, Is there any way rtpengine can be configured to use same UDP port to receive and transmit RTP packets? The set up I'm trying is SIP_Client --> Kamailio/rtpengine --> Freeswitch Internet Internet LAN LAN 192.168.1.10 I tried *rtpengine_offer * in kamailio.cfg, but rtpengine seems to always use different ports to Tx/Rx the packets. (Causing Freeswitch to auto adjust its ports, which mutes the audio channel one way). *rtpengine_offer("force symmetric media-address=192.168.1.10 replace-origin replace-session-connection ICE=remove");* *Thank you,* * - Deep N*

4 5

automatic string to int conversion for rtpengine flags failed
by Olli Heiskanen 19 Aug '14

19 Aug '14

Hello, As outcome to my earlier sdp/rtp challenges I've upgraded my Asterisk version to 11.11.0 and still use a realtime integration with Kamailio. Now I face a somewhat different problem. With my setup I also changed from jssip client to a sip.js client in my websocket implementation. I cloned the latest rtpengine from git today. I had to revert my Asterisk settings a bit, Asterisk was taking over with sdp handling, I don't know if this is relevant but that's why I got calls seemingly working before. When getting the 488 Not Acceptable, I arm a branch route and call rtpengine_offer there. When trying to call rtpengine_offer, I get the following log: (first I print the rtpengine_offer_flags to make sure what is passed to the function. Aug 15 15:04:16 u363id562 kamailio[32178]: INFO: <script>: MANAGE_RTPENGINE_BRANCH: rtpengine_offer_flags = rtcp-mux-demux trust-address replace-origin replace-session-connection ICE=remove RTP/AVP Aug 15 15:04:16 u363id562 kamailio[32178]: WARNING: <core> [rvalue.c:1016]: rval_get_int(): automatic string to int conversion for "rtcp-mux-demux trust-address replace-origin replace-session-connection ICE=remove RTP/AVP" failed Aug 15 15:04:16 u363id562 kamailio[32178]: WARNING: <core> [rvalue.c:1920]: rval_expr_eval_int(): rval expression conversion to int failed (1128,32-1128,32) Aug 15 15:04:16 u363id562 rtpengine[32159]: Got valid command from 127.0.0.1:44292: delete - { "call-id": "k7bft3u75p5e42ib039r", "received-from": [ "IP4", "client_public_address" ], "from-tag": "74dovi97bi", "command": "delete" } Aug 15 15:04:16 u363id562 rtpengine[32159]: [k7bft3u75p5e42ib039r] Call-ID to delete not found Aug 15 15:04:16 u363id562 rtpengine[32159]: Returning to SIP proxy: d7:warning38:Call-ID not found or tags didn't match6:result2:oke After this I see another 488 and the loop swirls on until Kamailio runs out of forking capacity. Can You guys explain why is this happening? I set the variable like this: $avp(rtpengine_offer_flags) = "rtcp-mux-demux trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"; cheers, Olli

1 1

Kamailio 4.1.(5-3) Dialog Module can't create hash if 2 same Invite comes.
by Yasin Caner 18 Aug '14

18 Aug '14

Hello i realize that if two same Invite comes to kamailio sequentially for a call, dialog module cannot create hash data and dont'give any error about it. After get ACK or BYE for this call , it gives warning like 'unable to find dialog for ACK with route param 'f3f1.0524' [7999:16976]' . So i cannot get any data call is hook up from dialog event_route. For prepaid account system , dialog module is weak or i made some mistakes. is there any better solution for accounting for high volume calls. I just need that call is hook up and closed or failed. which module is best option for it? (IMS ,SEAS, RADIUS) Thanks

4 7

XMLRPC problem
by Kelvin Chua 18 Aug '14

18 Aug '14

i am currently having problems on xmlrpc i am using node-xmlrpc and it fails when htable.dump returns more than 1 row. i raised this issue on the node-xmlrpc group and this is the explanation i got. Each <value> node inside <param> is supposed to only have one child node, so this looks like an invalid response. The<value> should (probably, i don't know your use case) contain an <array> with <data> wrapped around the structs. Seehttp://xmlrpc.scripting.com/spec.html does this mean kamailio is not following standards? here is the working response: <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> </value> </param> </params> </methodResponse> here is the non-working response: <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> <struct><member><name>entry</name><value><int>11</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>3::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> </value> </param> </params> </methodResponse> Kelvin Chua

4 6

uac module - remote registration updates
by Daniel-Constantin Mierla 18 Aug '14

18 Aug '14

Hello, lately I pushed several commits to enhance the handling of remote registrations with uac module. Among them: - uac.reg_info - rpc command to get the details of a remote registration record using filter on uuid, local/remote/auth username - uac.reg_enable and uac.reg_disable - rpc commands to enable/disable registration of a record from memory at runtime, without restarting kamailio - uac.reg_reload - rpc command to reload remote registration records from database (avoid restarting when adding new records in database) - uac.reg_refresh - rpc command to load a record from database based on l_uuid. If record is already in memory, then the password field is updated, otherwise a new full record is added in memory I barely had resources to test it properly, therefore I am asking people using this feature to give it a try and report the issues -- you have to use the git master branch. You can read more details about the new features in the readme of the uac module: - http://kamailio.org/docs/modules/devel/modules/uac.html Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Show domain name of particular online user in logs
by aawaise 17 Aug '14

17 Aug '14

Hello, I want to show the domain name of a callee (by which it is registered to the domain at the time of incoming invite packet) in the logs. I have tried adding a few lines to usrloc module. But didnot work. I am using LM_DBG command to show the domain name in the logs. How can I add some variable which holds the domain name and then print it in logs using LM_DBG or LM_ERR command ?? Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-domain-name-of-particular-onli… Sent from the Users mailing list archive at Nabble.com.

1 0

location of log file
by Kamrul Khan 16 Aug '14

16 Aug '14

Hi, Im new to kamailio. Can someone tell me where to find the logs (from xlog() or log() ) ? thanks in advanced

2 1

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