- sr-users - kamailio.org

Difference between modules_k and modules_s
by aawaise 16 Sep '14

16 Sep '14

In the source code of kamailio, I found some modules code in modules_k folder. Code of same modules is also given modules_s. Examples of such modules are registrar module and usrloc module. If we want to add somethings like log statements etc. Where the change is supposed to be done ? Thanks. Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Difference-between-modules-k-and-mo… Sent from the Users mailing list archive at Nabble.com.

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Re: [SR-Users] Dispatcher - failure_route to specific node
by Nicholas Gill 16 Sep '14

16 Sep '14

On 14/09/14 20:00, sr-users-request(a)lists.sip-router.org wrote: > In your script snippets it seems > that upon receiving the 302, in the failure route you set the $du and then > you pass the message to a RELAY route, which I suppose sends the 407 reply. This seems to be the case, yes. The 407 comes from kamailio on the already authenticated call. I presume this is done to trigger the caller to send another INVITE. > If you don't need for special reasons that the user performs a proxy > authentication, in the failure route you could just relay the INVITE to the > target FreeSWITCH after setting $du to the value of your custom header. This is exactly what I'm hoping to do, but I'm not sure of the correct configuration to achieve this. It was my intention to relay the INVITE to the target server by setting the $du and then trigger the RELAY route. I have also tried setting $du and then calling t_relay() directly, but I suspect I'm missing something minor but important. Cheers, -nick

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How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?
by Alex Villacís Lasso 15 Sep '14

15 Sep '14

I have a rtpproxy configuration that spawns several rtpproxy instances, using bridge mode. An example is shown below: /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 10000 -M 20000 Here, rtpproxy bridges between 192.168.2.18 and 127.0.0.1 . Now I want to migrate to rtpengine with the rtpproxy-ng module in kamailio. However, I do not find an equivalent to bridge mode in the rtpengine command-line parameters. I see the --ip=IP parameter, but the source code expects a single IP address, and cannot be specified more than once. The closest I see is the --advertised-ip=IP parameter, but I am not sure that it will do what I need.

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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

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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

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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

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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

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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

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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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