- sr-users - kamailio.org

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/>

3 2

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

3 3

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.

2 1

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

2 3

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

2 1

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.

2 1

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

unknown command <acc_rad_request>, missing loadmodule?
by Satish Patel 16 Aug '14

16 Aug '14

What is this? I have acc.so module loaded but still getting this error. I am running opensip 1.11 root@sip:~# opensips -c -f /etc/opensips/opensips.cfg Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 252, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 259, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 266, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 273, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 280, column 31-32: unknown command <acc_rad_request>, missing loadmodule? Aug 14 10:42:49 [7180] ERROR:core:main: bad config file (5 errors)

2 1

Text Operations on a Message
by AliReza Khoshgoftar Monfared 15 Aug '14

15 Aug '14

Hi list, How can I extract a specific part of my message body? I know that "textops" has the function "search_body(re)" but it returns True/False upon finding "re" in the body of the text, how can I return the actual text that regular expression matches? As an example, my message body is rate is 172#012 I want to extract the number "172" out of this. Thanks Alireza

3 11

Handling 407 reply
by Yuriy Gorlichenko 15 Aug '14

15 Aug '14

Hello I use UAC module to register trunks to my providers. I've try to call to my provider. I send INVITE to my porvider and then It recieve me 407 reply "proxy authentication required". Offcourse I can handle this reply manualy (by adding headers throuch header_add) and add md5 summ, but I think it`s not write and harder for Kamailio. I find module or function at some module that helps me send write INVITE with auth parameters as a reply ar 407 response of provider. Thanks for help.

2 2

Rate Limiting with Kamailio
by AliReza Khoshgoftar Monfared 15 Aug '14

15 Aug '14

Hi list, A quick question, What is the best way to limit the rate of the kamailio, Say I want it to process only 200 INVITEs per second I know this can be done using ratelimit module, but what if I want to change this rate dynamically? say, in an overload control scenario, I am getting a recommended rate piggybacked in my replies, and I read it, then how can I change the messge processing rate of kamailio in runtime inside my script? Is there a way to change the rate using ratelimit but inside script as messages come? The RPC function (rl.set_pipe) does it, but can I call it inside my script? Thanks, Alireza

3 3

Kamamilio ignores RTP engine returned string
by Yuriy Gorlichenko 15 Aug '14

15 Aug '14

Hello Rtpengine (rtpproxy-ng module) works fine with kamailio till today. Without any changes at kamailio or rtpengine kamailio ignores changed by rtpengine SDP content. To check this I use sdp_get() and after tying to call I print avp from this function and returned sdp body by rtpengine != this content My log: Jul 16 17:17:27 test rtpengine[5521]: Got valid command from 127.0.0.1:36484: offer - { "sdp": "v=0#015#012o=- 8380958077563183406 2 IN IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-semantic: WMS o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012m=audio 56984 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 192.168.56.1#015#012a=rtcp:56984 IN IP4 192.168.56.1#015#012a=candidate:2999745851 1 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2999745851 2 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2448668656 1 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:2448668656 2 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:1206525147 1 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:1206525147 2 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:3374335715 1 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:3374335715 2 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:2602400937 1 udp 2121998079 192.168.214.1 56988 typ host generation 0#015#012a=candidate:2602400937 2 udp 2121998079 192.168.214.1 56988 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:3748678400 1 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:3748678400 2 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:157933611 1 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:157933611 2 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:2275654163 1 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:2275654163 2 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:3583934553 1 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=candidate:3583934553 2 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=ice-ufrag:Oglxt4kDKBr8sfXY#015#012a=ice-pwd:iZAp/WE5TCqIoPA0hIpxztqq#015#012a=ice-options:google-ice#015#012a=fingerprint:sha-256 E3:E7:04:57:80:03:27:12:D6:88:F6:E7:3E:64:4A:55:DA:AD:3F:DA:4E:41:6E:6F:F0:D0:69:A6:AB:AE:79:4F#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://www.webrtc.org/experiments/rtp-hdrext/abs-send-time#015#012a=sendrec… 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:3151729659 cname:wvwdxd7V5hVETGHP#015#012a=ssrc:3151729659 msid:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=ssrc:3151729659 mslabel:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012a=ssrc:3151729659 label:bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012", "ICE": "remove", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "60m6hpu9sclqnjvhql2q", "received-from": [ "IP4", "213.108.168.163" ], "from-tag": "6oo4g9fr4c", "command": "offer" } Jul 16 17:17:27 test rtpengine[5521]: Unknown flag encountered: 'force' Jul 16 17:17:27 test rtpengine[5521]: [60m6hpu9sclqnjvhql2q] Creating new call Jul 16 17:17:27 test rtpengine[5521]: [60m6hpu9sclqnjvhql2q] Returning to SIP proxy: d3:sdp770:v=0#015#012o=- 8380958077563183406 2 IN IP4 123.33.24.45#015#012s=-#015#012t=0 0#015#012a=msid-semantic: WMS o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012m=audio 30008 RTP/AVP 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 123.33.24.45#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=ssrc:3151729659 cname:wvwdxd7V5hVETGHP#015#012a=ssrc:3151729659 msid:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=ssrc:3151729659 mslabel:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012a=ssrc:3151729659 label:bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=sendrecv#015#012a=rtcp:30009#015#012a=rtcp-mux#015#0126:result2:oke Jul 16 17:17:27 test /usr/sbin/kamailio[6969]: INFO: <script>: Faltured {INVITE} Jul 16 17:17:27 test /usr/sbin/kamailio[6969]: INFO: <script>: This is not falture {INVITE} Jul 16 17:17:27 test /usr/sbin/kamailio[6969]: INFO: <script>: SDP is {v=0#015#012o=- 8380958077563183406 2 IN IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-semantic: WMS o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012m=audio 56984 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 192.168.56.1#015#012a=rtcp:56984 IN IP4 192.168.56.1#015#012a=candidate:2999745851 1 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2999745851 2 udp 2122260223 192.168.56.1 56984 typ host generation 0#015#012a=candidate:2448668656 1 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:2448668656 2 udp 2122194687 192.168.142.1 56985 typ host generation 0#015#012a=candidate:1206525147 1 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:1206525147 2 udp 2122129151 192.168.2.120 56986 typ host generation 0#015#012a=candidate:3374335715 1 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:3374335715 2 udp 2122063615 10.34.8.7 56987 typ host generation 0#015#012a=candidate:2602400937 1 udp 2121998079 192.168.214.1 56988 typ host generation 0#015#012a=candidate:2602400937 2 udp 2121998079 192.168.214.1 56988 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:3748678400 1 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:3748678400 2 tcp 1518214911 192.168.142.1 0 typ host generation 0#015#012a=candidate:157933611 1 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:157933611 2 tcp 1518149375 192.168.2.120 0 typ host generation 0#015#012a=candidate:2275654163 1 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:2275654163 2 tcp 1518083839 10.34.8.7 0 typ host generation 0#015#012a=candidate:3583934553 1 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=candidate:3583934553 2 tcp 1518018303 192.168.214.1 0 typ host generation 0#015#012a=ice-ufrag:Oglxt4kDKBr8sfXY#015#012a=ice-pwd:iZAp/WE5TCqIoPA0hIpxztqq#015#012a=ice-options:google-ice#015#012a=fingerprint:sha-256 E3:E7:04:57:80:03:27:12:D6:88:F6:E7:3E:64:4A:55:DA:AD:3F:DA:4E:41:6E:6F:F0:D0:69:A6:AB:AE:79:4F#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://www.webrtc.org/experiments/rtp-hdrext/abs-send-time#015#012a=sendrec… 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:3151729659 cname:wvwdxd7V5hVETGHP#015#012a=ssrc:3151729659 msid:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012a=ssrc:3151729659 mslabel:o2PCyrMBSdtfFMt6oYXoSeBdfUskxKN6dWXE#015#012a=ssrc:3151729659 label:bdfb98b7-2eed-4d85-bdbb-b097df5c09f2#015#012}

3 3

can you define an array and loop in kamailio.cfg
by ANTHONY HESNAUX 15 Aug '14

15 Aug '14

hello all, Can you define an array and a loop in kamailio.cfg not work !!!! $var(i) = 0; tab[0]="192.168.0.2"; tab[1]="192.168.0.3 »; while($var(i)<2) { $var(rip) = $tab[$var(i)]; $uac_req(method)="REGISTER »; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.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(); }

2 1

sipcapture and haproxy
by Grant Bagdasarian 15 Aug '14

15 Aug '14

Hello, I just discovered HAproxy, and I was wondering if it is possible to use it in combination with the sipcapture module to store SIP messages on multiple database servers? Has anyone done this before? Regards, Grant

2 3

New Kamailio module Obfuscate
by Muhammad Shahzad 15 Aug '14

15 Aug '14

Hi, As already discussed in detail in following email thread, https://www.mail-archive.com/sr-users@lists.sip-router.org/msg19922.html The new Kamailio module obfuscate is ready for testing and can be downloaded at, http://webrtc.voip-demos.com/obfuscate.tbz2 It contains full code, with documentation and usage scenarios etc. Adding Kamailio dev team, can you guys review the code and add this module to Kamailio trunk if possible? I will be responsible for its maintenance and bug fixes etc. if any. Thank you.

2 23

what language is using in kamailio.cfg
by ANTHONY HESNAUX 15 Aug '14

15 Aug '14

hello every body, I would like to know what language is using in Kamailio.cfg because I don’t arrive to create a loop thank you anthony

3 2

RPMs - yum repositories
by Daniel-Constantin Mierla 15 Aug '14

15 Aug '14

Hello, courtesy lod.com (sponsoring the hardware and bandwidth) and Fred Posner helping with that, we got a server located in USA to use for the project. Main purpose for now is using it as a RPM repository, available as rpm.kamailio.org . The server takes the packages from open suse build service, making a local mirror. It is not yet automated, but should happen in the future (anyone that can help here with some script, is more than welcome to join the team) -- anyhow, we generate the rpms only at stable releases. The mirror should be usable as an yum repository or individual package download via http, more details at: - http://rpm.kamailio.org I am not an user of a distro using RPMs, therefore I am asking interested people in the community to give it a try and report here eventual issues. Also, if you build packages for other distributions, we can mirror them on the server -- get in touch to setup the process. 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

Send socket and sequential requests
by Alex Balashov 14 Aug '14

14 Aug '14

Hi, My proxy is bound to a.b.c.d:5060 and secondarily to e.f.g.h:5060. Most traffic comes into a.b.c.d:5060 and is relayed out that interface. Some initial INVITEs are routed out of e.f.g.h:5060, forced via $fs. The problem is that the Record-Route header still reflects a.b.c.d:5060 and that sequential messages from both ends end up entering and exiting a.b.c.d:5060. Many endpoints don't like that; for instance, e2e ACKs from the caller that come from a.b.c.d:5060 while the initial INVITE came from e.f.g.h:5060 can be a problem in that they are not matched by some UASs. What is the best way to approach this? Is this something covered by Kamailio's 'mhomed' core param? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 8

INVITE to provider
by Yuriy Gorlichenko 14 Aug '14

14 Aug '14

Hello. I Use UAC module for register to my porviders (I have several porviders at my server and). I successfully ring from provider to my asterisk servers and route calls to users that registers at Kamailio. So now I need route calls from users to my providers, But can not understand how to send INVITE to provider? Doesanyone has working example for it?

1 0

Fwd: Call disconnects within 30 seconds
by Rahul MathuR 14 Aug '14

14 Aug '14

Hello Davy, My constraint is that I cannot use TCP for this solution. I am attaching my kamailio.cfg file, please please help me to resolve this issue as soon as possible (it has already chewed many nights :( ) On Wed, Aug 13, 2014 at 8:50 PM, davy van de moere < davy.van.de.moere(a)gmail.com> wrote: > Hey Rahul, > > thank you for your pcap file. Your astpp is indeed not succeeding in > getting the OK delivered correctly to your pjsua. Also those 400 Bad > Session Description are no good. > > I do see as well fragmented packets passing by, perhaps an easy test would > be to have everything be sent over TCP. Big sip packets tend to suffer over > UDP networks. From your sip packets your setup looks quite standard, > although with a lot of headers (which is fine), so I'ld give it a try with > sending everything over tcp, t_relay_to_tcp should help you out! > > Good luck! > > > --- In reply to --- > I guess the attachment size was greater than 60Kb thereby getting > quarentined. > Re-sending it to you so that this issue could be resolved. > > > 2014-08-12 14:42 GMT+02:00 davy <davy.van.de.moere(a)gmail.com>: > > If your Kamailio setup is close to vanilla, it should do it by default. >> But Kamailio is a very powerfull tool, it can easily be setup to stop >> passing over ACKs :) >> >> To attempt to answer what explicitly sends along the ACKs, that will most >> likely be the t_relay function, together with the logic which went before >> it… >> >> A good old tcpdump will most likely enlighten us. >> >> Op 12-aug.-2014, om 14:39 heeft Rahul MathuR <rahul.ultimate(a)gmail.com> >> het volgende geschreven: >> >> Hello Davy, >> >> Thanks for writing back.. >> >> Tonight I'll take the tcpdump on Kamailio box and share the file. >> >> Please note that Kamailio and Freeswitch are both on public IP & at >> Freeswitch param, enable_timer=false is set. >> >> Is there any explicit way wherein ACKs can be transmitted to FS ? >> >> >> Thanks in advance ! >> >> >> On Tue, Aug 12, 2014 at 1:15 PM, davy van de moere < >> davy.van.de.moere(a)gmail.com> wrote: >> >>> Are you sure you're getting the ACK correctly to FS? >>> >>> FS typically has this behavior when it did not correctly receive a >>> confirmation of an answer, and after 30 seconds disconnects, as for FS the >>> call has failed. >>> >>> Do you have a trace of the packets? >>> >>> grtz, >>> Davy Van De Moere >>> >>> >>> 2014-08-12 13:37 GMT+02:00 Rahul MathuR <rahul.ultimate(a)gmail.com>: >>> >>>> Hello, >>>> >>>> I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and >>>> Sip server FreeSwitch. >>>> Whenever I call directly from UAC to Sip server, the call gets >>>> established for as long as I want, however when I use the proxy in between, >>>> it gets disconnected within 30 seconds. It seems that FS sends a BYE within >>>> 30 seconds. >>>> >>>> I would really appreciate if anybody please guide me where I am going >>>> wrong in this case ? >>>> >>>> -- >>>> Warm Regds. >>>> MathuRahul >>>> >>>> _______________________________________________ >>>> 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 >>> >>> >> >> >> -- >> Warm Regds. >> MathuRahul >> _______________________________________________ >> 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 > > -- Warm Regds. MathuRahul -- Warm Regds. MathuRahul

1 1

Keep Number of Messages per Source
by AliReza Khoshgoftar Monfared 13 Aug '14

13 Aug '14

Hi List, I am trying to keep a counter for number of messages received per source ip in my kamailio script. The basic challenge is to keep this data in a neat structure. Ideally, I want some array with keys named after source IP addresses, sth like number_of_messages["10.10.10.10"] = 23 Of course, this data structure shall be script-persistent (as opposed to transaction persistent) Is there any suggestion to do this? (keep a record of # of messages received from each source IP address and log them)?

3 7

How do I make subst_hf() use variables in search (not replacement)?
by Alex Villacís Lasso 13 Aug '14

13 Aug '14

Consider the following snippet: if (is_present_hf("Contact")) { xlog("L_ALERT", "===== reply to SUBSCRIBE has Contact: $ct\n"); xlog("L_ALERT", "===== want to replace with $td\n"); xlog("L_ALERT", "===== regexp to use is /$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asterisk.bindport)/$td/\n"); if (subst_hf("Contact", "/127.0.0.1:5080/$td/", "a")) { xlog("L_ALERT", "===== reply had Contact modified\n"); } } If I use the hardcoded regexp "/127.0.0.1:5080/$td/", subst_hf() replaces the Contact value correctly. However, if I use "/$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asterisk.bindport)/$td/" (where asterisk.bindip and asterisk.bindport are the supposed values 127.0.0.1 and 5080), subst_hf() fails to replace the Contact value. How do I make use of the existing configuration variables in order to substitute in the Contact header?

2 6

Kamailio + FreeRadius + MySQL Accounting STOP not updating
by Satish Patel 13 Aug '14

13 Aug '14

I have install Kamailio 4.1 with MySQL database so all users stored in MySQL (authentication), Later I have installed FreeRadius for accounting only, Just accounting (Invite/Bye). In short MySQL for authentication and FreeRadius for accounting ( CDRTool). Problems: If i register user using SIP phone and call other SIP extension then freeradius start accounting and insert entry in MySQL/radaccts tables if i end call then it doesn't update END accounting ( Bye ). I have google it everywhere but didn't get single clue so i hope i get something here from you guys -- kamailio.cfg ( default file i just add following radius section) loadmodule "acc.so" loadmodule "acc_radius.so" ... ... # ------- FreeRadius Accounting ------- modparam("acc_radius", "radius_flag", 1) modparam("acc_radius", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc_radius", "service_type", 15) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc_radius", "radius_extra", "User-Name=$Au") -- /etc/freeradius/sql.conf sql { driver = "rlm_sql_mysql" server = "localhost" login = "radius" password = "radius" radius_db = "radius" sqltrace = no sqltracefile = ${logdir}/sqltrace-%Y%m%d.log num_sql_socks = 25 connect_failure_retry_delay = 60 accounting_start_query = "\ CALL insert_radacct_record( \ 'radius', \ '%{Acct-Session-Id}', \ '%{Acct-Unique-Session-Id}', \ '%{Billing-Party}', \ '%{Billing-Party}', \ '%{SIP-Proxy-IP}', \ '%{NAS-Port}', \ FROM_UNIXTIME('%S'), \ '0', \ '0', \ '0', \ '0', \ '%{Called-Station-Id}', \ '%{Calling-Station-Id}', \ '%{Sip-Response-Code}', \ '%{Service-Type}', \ '%{ENUM-TLD}', \ '%{Framed-IP-Address}', \ UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ '%{Sip-Response-Code}', \ '%{Sip-Method}', \ '%{Sip-Translated-Request-URI}', \ '%{Sip-To-Tag}', \ '%{Sip-From-Tag}', \ '%{Sip-RPId}', \ '%{Source-IP}', \ '%{Source-Port}', \ '%{Canonical-URI}', \ '', \ '', \ '%{Sip-Application-Type}', \ '%{User-Agent}', \ '%{From-Header}' \ )" accounting_stop_query = "\ CALL update_radacct_record( \ 'radius', \ FROM_UNIXTIME('%S'), \ UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ '', \ '%{X-RTP-Stat}', \ '%{Acct-Session-Id}', \ '%{Sip-To-Tag}', \ '%{Sip-From-Tag}' \ )" -- Radius logs rad_recv: Accounting-Request packet from host 127.0.0.1 port 36379, id=11, length=152 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Invite Event-Timestamp = "Aug 12 2014 22:45:36 EDT" Sip-From-Tag = "59eb1875" Sip-To-Tag = "d443e36e" Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ." User-Name = "2002(a)192.168.1.5" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Tue Aug 12 22:45:36 2014 : Info: # Executing section preacct from file /etc/freeradius/sites-enabled/default Tue Aug 12 22:45:36 2014 : Info: +- entering group preacct {...} Tue Aug 12 22:45:36 2014 : Info: ++[preprocess] returns ok Tue Aug 12 22:45:36 2014 : Info: [acct_unique] Hashing 'NAS-Port = 5060,Client-IP-Address = 127.0.0.1,NAS-IP-Address = 127.0.0.1,Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ.",User-Name = "2002(a)192.168.1.5 "' Tue Aug 12 22:45:36 2014 : Info: [acct_unique] Acct-Unique-Session-ID = "117c7740de62ba6c". Tue Aug 12 22:45:36 2014 : Info: ++[acct_unique] returns ok Tue Aug 12 22:45:36 2014 : Info: [suffix] Looking up realm "192.168.1.5" for User-Name = "2002(a)192.168.1.5" Tue Aug 12 22:45:36 2014 : Info: [suffix] No such realm "192.168.1.5" Tue Aug 12 22:45:36 2014 : Info: ++[suffix] returns noop Tue Aug 12 22:45:36 2014 : Info: ++[files] returns noop Tue Aug 12 22:45:36 2014 : Info: # Executing section accounting from file /etc/freeradius/radiusd.conf Tue Aug 12 22:45:36 2014 : Info: +- entering group accounting {...} Tue Aug 12 22:45:36 2014 : Info: [detail] expand: /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d -> /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:45:36 2014 : Info: [detail] /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:45:36 2014 : Info: [detail] expand: %t -> Tue Aug 12 22:45:36 2014 Tue Aug 12 22:45:36 2014 : Info: ++[detail] returns ok Tue Aug 12 22:45:36 2014 : Info: [sql] expand: CALL insert_radacct_record( 'radius', '%{Acct-Session-Id}', '%{Acct-Unique-Session-Id}', '%{Billing-Party}', '%{Billing-Party}', '%{SIP-Proxy-IP}', '%{NAS-Port}', FROM_UNIXTIME('%S'), '0', '0', '0', '0', '%{Called-Station-Id}', '%{Calling-Station-Id}', '%{Sip-Response-Code}', '%{Service-Type}', '%{ENUM-TLD}', '%{Framed-IP-Address}', UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', '%{Sip-Response-Code}', '%{Sip-Method}', '%{Sip-Translated-Request-URI}', '%{Sip-To-Tag}', '%{Sip-From-Tag}', '%{Sip-RPId}', '%{Source-IP}' Tue Aug 12 22:45:36 2014 : Debug: rlm_sql (sql): Reserving sql socket id: 24 Tue Aug 12 22:45:36 2014 : Debug: rlm_sql (sql): Released sql socket id: 24 Tue Aug 12 22:45:36 2014 : Info: ++[sql] returns ok Sending Accounting-Response of id 11 to 127.0.0.1 port 36379 Tue Aug 12 22:45:36 2014 : Info: Finished request 0. Tue Aug 12 22:45:36 2014 : Info: Cleaning up request 0 ID 11 with timestamp +13 Tue Aug 12 22:45:36 2014 : Debug: Going to the next request Tue Aug 12 22:45:36 2014 : Info: Ready to process requests. rad_recv: Accounting-Request packet from host 127.0.0.1 port 35683, id=149, length=152 Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = "Aug 12 2014 22:46:20 EDT" Sip-From-Tag = "59eb1875" Sip-To-Tag = "d443e36e" Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ." User-Name = "2002(a)192.168.1.5" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Tue Aug 12 22:46:20 2014 : Info: # Executing section preacct from file /etc/freeradius/sites-enabled/default Tue Aug 12 22:46:20 2014 : Info: +- entering group preacct {...} Tue Aug 12 22:46:20 2014 : Info: ++[preprocess] returns ok Tue Aug 12 22:46:20 2014 : Info: [acct_unique] Hashing 'NAS-Port = 5060,Client-IP-Address = 127.0.0.1,NAS-IP-Address = 127.0.0.1,Acct-Session-Id = "M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ.",User-Name = "2002(a)192.168.1.5 "' Tue Aug 12 22:46:20 2014 : Info: [acct_unique] Acct-Unique-Session-ID = "117c7740de62ba6c". Tue Aug 12 22:46:20 2014 : Info: ++[acct_unique] returns ok Tue Aug 12 22:46:20 2014 : Info: [suffix] Looking up realm "192.168.1.5" for User-Name = "2002(a)192.168.1.5" Tue Aug 12 22:46:20 2014 : Info: [suffix] No such realm "192.168.1.5" Tue Aug 12 22:46:20 2014 : Info: ++[suffix] returns noop Tue Aug 12 22:46:20 2014 : Info: ++[files] returns noop Tue Aug 12 22:46:20 2014 : Info: # Executing section accounting from file /etc/freeradius/radiusd.conf Tue Aug 12 22:46:20 2014 : Info: +- entering group accounting {...} Tue Aug 12 22:46:20 2014 : Info: [detail] expand: /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d -> /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:46:20 2014 : Info: [detail] /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /var/log/freeradius/radacct/127.0.0.1/detail-20140812 Tue Aug 12 22:46:20 2014 : Info: [detail] expand: %t -> Tue Aug 12 22:46:20 2014 Tue Aug 12 22:46:20 2014 : Info: ++[detail] returns ok Tue Aug 12 22:46:20 2014 : Info: [sql] expand: CALL update_radacct_record( 'radius', FROM_UNIXTIME('%S'), UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', '', '%{X-RTP-Stat}', '%{Acct-Session-Id}', '%{Sip-To-Tag}', '%{Sip-From-Tag}' ) -> CALL update_radacct_record( 'radius', FROM_UNIXTIME('2014-08-12 22:46:20'), UNIX_TIMESTAMP('2014-08-12 22:46:20') - 'Aug 12 2014 22:46:20 EDT', '', '', 'M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ.', 'd443e36e', '59eb1875' ) Tue Aug 12 22:46:20 2014 : Debug: rlm_sql (sql): Reserving sql socket id: 23 Tue Aug 12 22:46:20 2014 : Info: [sql] expand: stop packet with zero session length. [user '%{User-Name}', nas '%{NAS-IP-Address}'] -> stop packet with zero session length. [user '2002(a)192.168.1.5', nas '127.0.0.1'] Tue Aug 12 22:46:20 2014 : Info: [sql] stop packet with zero session length. [user '2002(a)192.168.1.5', nas '127.0.0.1'] Tue Aug 12 22:46:20 2014 : Debug: rlm_sql (sql): Released sql socket id: 23 Tue Aug 12 22:46:20 2014 : Info: ++[sql] returns noop Sending Accounting-Response of id 149 to 127.0.0.1 port 35683 Tue Aug 12 22:46:20 2014 : Info: Finished request 1. Tue Aug 12 22:46:20 2014 : Info: Cleaning up request 1 ID 149 with timestamp +57 Tue Aug 12 22:46:20 2014 : Debug: Going to the next request Tue Aug 12 22:46:20 2014 : Info: Ready to process requests. -- SQL tables ( Time is wired 1969 ) mysql> select * from radacct196912; +-----------+----------------------------------------------+------------------+----------+-------+--------------+-----------+-------------+---------------------+---------------------+-----------------+---------------+-------------------+------------------+-----------------+------------------+-----------------+------------------+--------------------+-------------+---------+-----------------+----------------+---------------+-----------+-----------------+----------+------------+-------------------------+---------------+--------------------+-----------+---------+---------------+----------+------------+--------------+-----------+-----------+---------------+------+-------+------------+-----------+-----------+---------------+------------+-----------+---------+ | RadAcctId | AcctSessionId | AcctUniqueId | UserName | Realm | NASIPAddress | NASPortId | NASPortType | AcctStartTime | AcctStopTime | AcctSessionTime | AcctAuthentic | ConnectInfo_start | ConnectInfo_stop | AcctInputOctets | AcctOutputOctets | CalledStationId | CallingStationId | AcctTerminateCause | ServiceType | ENUMtld | FramedIPAddress | AcctStartDelay | AcctStopDelay | SipMethod | SipResponseCode | SipToTag | SipFromTag | SipTranslatedRequestURI | SipUserAgents | SipApplicationType | SipCodecs | SipRPID | SipRPIDHeader | SourceIP | SourcePort | CanonicalURI | DelayTime | Timestamp | DestinationId | Rate | Price | Normalized | BillingId | MediaInfo | RTPStatistics | FromHeader | UserAgent | Contact | +-----------+----------------------------------------------+------------------+----------+-------+--------------+-----------+-------------+---------------------+---------------------+-----------------+---------------+-------------------+------------------+-----------------+------------------+-----------------+------------------+--------------------+-------------+---------+-----------------+----------------+---------------+-----------+-----------------+----------+------------+-------------------------+---------------+--------------------+-----------+---------+---------------+----------+------------+--------------+-----------+-----------+---------------+------+-------+------------+-----------+-----------+---------------+------------+-----------+---------+ | 1 | Y2ExZjhkMTU4ODRmN2MwMGM1N2Q1Y2E5YTA5ZjAzNjg | e46f9d0439141495 | | | | 5060 | | 1969-12-31 19:00:00 | 1969-12-31 19:00:00 | 0 | NULL | NULL | | 0 | 0 | | | 200 | Sip-Session | | | 1407883391 | 0 | Invite | 200 | 70cc9216 | e2cb1f0c | | | | | | | | | | | 0 | | | NULL | 0 | | NULL | | | | | | 2 | MzYyY2JhM2Q3MzEyN2U1ODJkMDA1NGNiMTMzYmE0OTc | aa9b8bcb6effa2c7 | | | | 5060 | | 1969-12-31 19:00:00 | 1969-12-31 19:00:00 | 0 | NULL | NULL | | 0 | 0 | | | 200 | Sip-Session | | | 0 | 0 | Invite | 200 | c6ea6056 | 407f047e | | | | | | | | | | | 0 | | | NULL | 0 | | NULL | | | | | | 3 | M2IwZTVkY2Y5YTU2ZjdmOTQ5NWMzYzI1NTU3MWMwYjQ. | 117c7740de62ba6c | | | | 5060 | | 1969-12-31 19:33:34 | 1969-12-31 19:33:34 | 0 | NULL | NULL | | 0 | 0 | | | 200 | Sip-Session | | | 1407897936 | 1407897980 | Invite | 200 | d443e36e | 59eb1875 | | | | | | | | | | | 0 | | | NULL | 0 | | NULL | | | | | +-----------+----------------------------------------------+------------------+----------+-------+--------------+-----------+-------------+---------------------+---------------------+-----------------+---------------+-------------------+------------------+-----------------+------------------+-----------------+------------------+--------------------+-------------+---------+-----------------+----------------+---------------+-----------+-----------------+----------+------------+-------------------------+---------------+--------------------+-----------+---------+---------------+----------+------------+--------------+-----------+-----------+---------------+------+-------+------------+-----------+-----------+---------------+------------+-----------+---------+ 3 rows in set (0.00 sec)

2 3

Kamailio IMS - forking based on the SIP method
by Daniel Ciprus 13 Aug '14

13 Aug '14

Hello, is there a way to make SCSCF do forking based on SIP method ? The problem is that our implementation on app server is not fully done and we want INVITE to be forked the way it is but OPTIONS shouldn't be forked there since forking is happening twice in this case which results to storm of OPTIONS on the network. thanks -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

1 0

freeradius Duplicate attribute name Event-Timestamp
by Satish Patel 13 Aug '14

13 Aug '14

somehow i am getting following error, if i start radius Error: Errors reading dictionary: dict_init: /etc/freeradius/dictionary.opensips[93]: dict_addattr: Duplicate attribute name Event-Timestamp If i remove it from /etc/freeradius/dictionary.opensips file it works! but my it screwed my time in mysql table ATTRIBUTE Event-Timestamp 230 string Just run grep to find out but it is not anywhere define but still i am getting error root@sip:/etc/freeradius# grep "Event-Timestamp" * -r dictionary.opensips:ATTRIBUTE Event-Timestamp 230 string sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: FROM_UNIXTIME(%{Event-Timestamp}), \ sql.conf: FROM_UNIXTIME(%{Event-Timestamp}), \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \ sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \

2 3

Dialog module event_route problem
by betul.cerit＠netgsm.com.tr 13 Aug '14

13 Aug '14

Hello; I am trying to build a prepaid accounting system. I use dialog module's start/end event-route. When even_route[dialog:started] triggered, a http_query works for the started call. When event_route[dialog:end] triggered , a http_query works for call is ended. When http_query works in event_route[dialog:started], i get some CRITICAL ERROR like 'Aug 12 12:16:09 /usr/local/sbin/kamailio[10151]: CRITICAL: dialog [dlg_timer.c:205]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f7634d925d8 tl->next=(nil) tl->prev=(nil)' 'Aug 12 12:16:09 /usr/local/sbin/kamailio[10151]: ERROR: dialog [dlg_handlers.c:1263]: dlg_onroute(): failed to update dialog lifetime' Http_query works very well. I can get all call information on Web Servers but default_time out or other timeout isn't working. After i commented out http_query in event_route[dialog:started] , it stopped giving ERROR but i realized that dialog cannot uptade dialog hash map .I can see all finished call with "kamctl dialog show". After i closed http_query in even_route[dialog:end] , all problems are solved. I think a problem occured between http_query and dialog event_routes.It cannot control dialog lifetime.Any idea about this problem or suggest an alternative way will be appreciated.My configuration is as follows. Best Regards Betül. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu ; useropkodu ") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #!endif route[CHCK_DLG]{ xlog("L_INFO","DLG:AVP=> :>:$avp(i:77):$avp(op_kodu):$avp(i:69) "); $dlg_ctx(timeout_bye)=1; $avp(unique_id)=$sruid; /* Time out suresi i:10 konusabilecegi sure saniye */ $avp(i:10)=0; $avp(i:10)=(int)$avp(i:77); #$avp(i:10)=10; $dlg_ctx(timeout_route)=33; $dlg_var(uniqueid)=$avp(unique_id); $dlg_var(userid)=$avp(userid); $dlg_var(opkodu)=$avp(op_kodu); set_dlg_profile("userid","$avp(userid)"); set_dlg_profile("opkodu","$avp(op_kodu)"); set_dlg_profile("useropkodu","$avp(userid):$avp(op_kodu)"); if(get_profile_size("useropkodu","$avp(d_user_opkodu)")){ xlog("L_INFO","Userid_Size:$avp(d_user_opkodu) kk : $avp(userid):$avp(op_kodu) "); xlog("L_INFO","Userid_Limit:$sht(sayac=>$avp(userid):$avp(op_kodu))"); if(!($sht(sayac=>$avp(userid):$avp(op_kodu))>$avp(d_user_opkodu))){ sl_send_reply("403","Kapasite Asildi."); exit; } } dlg_manage(); return; } event_route[dialog:start]{ xlog("L_ALERT","START:CI:$dlg(callid):u_id:$dlg_var(uniqueid):U_id:$dlg_var(userid) "); xlog("L_ALERT","START:lifetime:$avp(i:10):opkodu:$dlg_var(opkodu):$avp(aranan)" ); #!ifdef WITH_UTILS $var(http_res)=http_query("http://bla.com/somestuff.php?userid=$avp(userid)&aranan=$avp(aranan)&callid…","$var(result)"); if($var(http_res)=="200"){ json_get_field("$var(result)","sonuc","$var(snc)"); json_get_field("$var(result)","yorum","$var(yorum)"); json_get_field("$var(result)","yorum","$var(debug)"); if($var(snc)!=200){ xlog("L_ERR","START:CI:$ci:Sonuc:$var(snc):Yorum:$var(yorum):Debug:$var(debug)"); }else{ xlog("L_INFO","START:CI:$ci:Sonuc:$var(snc)"); } }else{ xlog("L_ERR","START:CI:$ci:HTTP_RESULT:$var(http_res)"); } #!endif } event_route[dialog:end]{ xlog("L_ALERT","END ::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::: :"); xlog("L_ALERT","END:CI:$ci:START_TIME:$dlg(start_ts):NOW:$TS"); $var(billsec)=$TS-$dlg(start_ts); xlog("L_ALERT","END:SURE : $var(billsec): u_id: $dlg_var(uniqueid):code:$T_reply_code:$DLG_status "); xlog("L_ALERT","END:U_ID:$dlg_var(userid):O_Kodu:$dlg_var(opkodu):"); if($var(billsec)<0){ $var(billsec)=$DLG_lifetime; xlog("L_ERR","START:CI:$ci:Billsec Negatif geldi : lifetime alinacak:$DLG_lifetime"); } #!ifdef WITH_UTILS $var(http_res)=http_query("http://bla.com/somestuff.php?uniqueid=$dlg_var(uniqueid)&billsec=$var(bills…","$var(result)"); if($var(http_res)=="200"){ json_get_field("$var(result)","sonuc","$var(snc)"); json_get_field("$var(result)","yorum","$var(yorum)"); json_get_field("$var(result)","yorum","$var(debug)"); if($var(snc)!=200){ xlog("L_ERR","START:CI:$ci:Sonuc:$var(snc):Yorum:$var(yorum):Debug:$var(debug)"); } }else{ xlog("L_ERR","START:CI:$ci:HTTP_RESULT:$var(http_res)"); } #!endif xlog("L_ALERT","END ::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::: :"); }

2 2

Call disconnects within 30 seconds
by Rahul MathuR 13 Aug '14

13 Aug '14

Hello, I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and Sip server FreeSwitch. Whenever I call directly from UAC to Sip server, the call gets established for as long as I want, however when I use the proxy in between, it gets disconnected within 30 seconds. It seems that FS sends a BYE within 30 seconds. I would really appreciate if anybody please guide me where I am going wrong in this case ? -- Warm Regds. MathuRahul

3 5

SDP ending with CRLF and a=nortpproxy:yes
by Narsay, Deep 12 Aug '14

12 Aug '14

Hello, Please let me know if there is any work around for this. Details of the issue: kamailio.cfg has this sequence in kamailio.cfg, ...... modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)" ) ........ sdp_remove_line_by_prefix("a=candidate"); set_iattr("ice_priority", "1"); rtpproxy_manage("frwie"); I'm seeing an issue towards the bottom of the SDP message. .... a=ssrc:738345272 mslabel:VPELrlXvdfGmtevlYLEeVm6zq4JHvI4g a=ssrc:738345272 label:VPELrlXvdfGmtevlYLEeVm6zq4JHvI4ga0 a=ice-ufrag:YJuJWr9clIJHhENw a=ice-pwd:3v1D1ShCg8xIlDZ9 a=candidate:1297581420 1 udp 659136 10.1.2.3 17534 typ host generation 0 a=candidate:1297581420 2 udp 659136 10.1.2.3 17535 typ host generation 0 a=ptime:20 \r\n a=nortpproxy:yes\r\n The call gets rejected by UA due to an invalid (empty) line in SDP. Very old but somewhat similar issue: http://lists.sip-router.org/pipermail/users/2008-September/019605.html (Module authors, I'll be happy to code/test or assist in any way possible if needed.) Just to test it, I commented the following code snippet and the call worked. But that can't be the permanent fix. Thank you, - Deep N Code snippet:rtp_proxy.c::force_rtp_proxy starting from line 2850 * if (proxied == 0 && nortpproxy_str.len) { cp = pkg_malloc((nortpproxy_str.len + CRLF_LEN) * sizeof(char)); if (cp == NULL) { LM_ERR("out of pkg memory\n"); return -1; } anchor = anchor_lump(msg, body.s + body.len - msg->buf, 0, 0); if (anchor == NULL) { LM_ERR("anchor_lump failed\n"); pkg_free(cp); return -1; } memcpy(cp, CRLF, CRLF_LEN); memcpy(cp + CRLF_LEN, nortpproxy_str.s, nortpproxy_str.len); if (insert_new_lump_after(anchor, cp, nortpproxy_str.len + CRLF_LEN, 0) == NULL) { LM_ERR("insert_new_lump_after failed\n"); pkg_free(cp); return -1; } }*

2 3

kamctl not creates /tmp/kamailio_fifo file
by Yuriy Gorlichenko 12 Aug '14

12 Aug '14

Hello I try to use kamctl dispatcher reload cmd but in gives me an error kamctl dispatcher reload ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. when i try to user kamcmd dispather.reload it gives me: ERROR: connect_unix_sock: connect(/tmp/kamailio_ctl): No such file or directory At kamailio.cfg I defined modparams for ctl.so and enables ctl.so module too modparam("ctl", "mode", 0600) modparam("ctl", "user", "root") modparam("ctl", "fifo", "fifo:/tmp/kamailio_fifo") modparam("ctl", "fifo", "/tmp/kamailio_fifo2") modparam("ctl", "fifo", "udp:*:2050") # fifo protocol over udp modparam("ctl", "fifo", "tcp:*:2050") # fifo over tcp At the kamctlrc I defined FIFOPATH="/tmp/kamailio_fifo" So I can not understand what is wrong with my FIFO file?

3 5

New developer: Jorj Bauer
by Daniel-Constantin Mierla 11 Aug '14

11 Aug '14

Hello, I want to announce that a new developer got GIT write access to repository: Jorj Bauer from University of Pennsylvania, USA - he is mainly joining the efforts to maintain and develop further the SCA module: * http://kamailio.org/docs/modules/stable/modules/sca.html His git commit id is: jorj My warm welcome and thanks for joining to help 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 ::: Oct 15-17, San Francisco, USA

1 0

Routing for case of client's changing networks
by Dmytro Bogovych 11 Aug '14

11 Aug '14

Greetings all. I have following use case: 0) Peer A and B registers on kamailio via TLS. 1) Peer A establishes call to peer B. 2) Peer A changes network and call has to be refreshed. Peer A reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer. The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering. Is there way to solve or avoid this situation? Thank you :)

2 4

http_query results question
by Jonathan Hunter 11 Aug '14

11 Aug '14

Hi Guys, I have used the http_query function in the past, in conjunction with the json module, on version 3.3 and unbuntu. I am now using version 4.1 on Centos 6.5, and I am using the http_query, which I'm aware records in the results the first line of the response. Can you select the result to be recorded depending on the line in the response, for example if a number is returned only in line 4, can this be extracted from a 200 ok response using this function? Or is this then a need to use JSON module, and if so which repository for centos 6.5 contains it! :) Sorry for the question! Thanks Jon

3 8

Dispatcher caches IP
by Yuriy Gorlichenko 11 Aug '14

11 Aug '14

Hello. I try to integrate dispathcer module (with db table dispatcher) to my Kamailio server. I have problem with changing IP at db table, but when I calling - call goes to old IP that alread deleted from db. I tested my table with query that selects destination address from table, and it replyes to me correct address. I use round-robin ("4") algorythm modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.0.2") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") How to disable this cache?

4 5

Accessing Message Buffer in Kamailio
by AliReza Khoshgoftar Monfared 08 Aug '14

08 Aug '14

Hi list, Is there a way to survey the message buffer (iei the queue where arriving sip messages wait to be processed by worker threads) in Kamailio? Is there a way to check the number of messages in this buffer? Additionally, is it possible to survey the number of messages by method (INVITE, BYE, etc.)? I know that the ratelimit module ( http://kamailio.org/docs/modules/stable/modules/ratelimit.html) shall access such buffers, but I do not see any variable accessible in the config offered by that module. Am I missing sth or is there a way to do so? Thanks Alireza

2 2

[Kamailio-4.1.4]Make Install Error
by Djamel Bahamid 08 Aug '14

08 Aug '14

Hello everybody, I am new on this environment ( Kamailio). I try to install(settle) Kamailio 4.1.4, on an O.S Suse Sles11 SP3, all the compilation took place, with the exception of the installation (make install), where I obtain the following error: make[2]: Nothing to be done for `install-if-newer'. touch /usr/local/lib64/kamailio/modules/userblacklist.so install -m 755 userblacklist.so /usr/local/lib64/kamailio/modules make[2]: `libkmi.so.1.0' is up to date. make[2]: `libsrdb1.so.1.0' is up to date. make[2]: `libkcore.so.1.0' is up to date. make[2]: `libsrutils.so.1.0' is up to date. LD (gcc) [M usrloc.so] usrloc.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 18: syntax error near unexpected token `fi' */bin/sh: -c: line 18: ` fi ; \' make: *** [install-cfg] Error 2 * Could you help me please to solve this error, Best Regards, Djamel.

4 14

ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid"
by Manuel Camarg 08 Aug '14

08 Aug '14

I have this config: WebRTC sipml5 -> Kamailio -> Asterisk -> Kamailio -> WebRTC sipml5 When placing a communication between two sipml5 points I get this errors: ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid" ERROR: <core> [forward.c:532]: forward_request(): ERROR: forward_request: bad host name df7jal23ls0d.invalid, dropping packet ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) If I place a comm between WebRTC sipml5 -> Kamailio -> Asterisk -> Softphone (connected to Asterisk) Everything works fine I'm running RTPEngine as RTP Proxy for DTLS/SRTP issues SIP INVITE signalling Ringing 180 message and 200 OK gets done fine Any ideas where this issue might come from? Looks something related with media traffic (RTP). Kind regards!

2 2

Complex NAT Scenario,HELP!
by Agiftel 08 Aug '14

08 Aug '14

Hi all, I need help configuring a little bit complex NAT/PAT scenario. Idea is that clients on Internet registers themselvs against kamailio and then it routes requests to Alcatel PBX. kamailio acts also as NAT proxy ( so that clients on internet does not need vpn connection. Something like EDGE SERVER for MS Lync clients) Below the scenario Alcatel PBX: 10.9.6.3 Kamailio internal NIC: 172.16.52.240 Kamailio DMZ NIC: 10.9.23.180 PU.BL.I.C.IP: 1.2.3.4 ALCATEL PBX(LAN 10.9.6.3)————-------------- | |———-(LAN INT 172.16.52.240) | |KAMAILIO (2 NICs) ———————————— | | | | |————(LAN dmz 10.9.23.180) PAT/NAT | |————firewal | | PU.BL.I.C.IP ^ | | Client on internet As you can see there is a firewall that make NAT and PAT from PU.BL.I.C.IP to internal Kamailio DMZ nic. natted ports are 5060 and range from 30000 to 65000 ( for RTP ) I'am tryng to use kamailio.cfg that comes with installation.. Now I am making some changes but is not working. Internal client ( lan 172.16.52.x ), registered on kamailio, can make call to alcatel phones and viceversa. External client ( on internet ) can register to kamailio but cannot call alcatel phone nor client registered on kamailio. Here some traces CALL FROM EXTERNAL CLIENT TO INTERNAL CLIENT: *U 2014/07/25 11:26:55.709196 2.2.2.2:39464 -> 10.9.23.180:5060 INVITE sip:1001@lwsbc.longwave.eu SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:casa@2.2.2.2:39464>. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4.6.1 73073-dadc6769-W. Content-Length: 275. . v=0. o=- 13050753989744621 1 IN IP4 192.168.1.102. s=X-Lite release 4.6.1 stamp 73073. c=IN IP4 2.2.2.2. t=0 0. m=audio 65220 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:26:55.822346 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 100 LW-SBC is Trying. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Server: LW-SBC. Content-Length: 0. . U 2014/07/25 11:26:55.891270 172.16.52.240:5060 -> 172.16.52.53:1384 INVITE sip:bria@172.16.52.53:1384;rinstance=b31d64b3d4ead28c SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Max-Forwards: 69. Contact: <sip:casa@2.2.2.2:39464>. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4.6.1 73073-dadc6769-W. Content-Length: 510. . v=0. o=- 13050753989744621 1 IN IP4 192.168.1.102. s=X-Lite release 4.6.1 stamp 73073. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30108 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30109. a=ice-ufrag:wa6qnn48. a=ice-pwd:L2qWms8nR1SA9Z8rWtDiMpzjyEHV. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30108 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30109 typ host. U 2014/07/25 11:26:55.973284 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Content-Length: 0. . U 2014/07/25 11:26:55.986747 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. User-Agent: Bria 3 release 3.5.5 stamp 71243. Allow-Events: hold, talk. Content-Length: 0. . U 2014/07/25 11:26:56.040076 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. User-Agent: Bria 3 release 3.5.5 stamp 71243. Allow-Events: hold, talk. Content-Length: 0. . U 2014/07/25 11:26:58.473501 10.9.23.180:5060 -> 2.2.2.2:39464 OPTIONS sip:2.2.2.2:39464 SIP/2.0. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0. From: sip:pinger@lwsbc.eu;tag=uloc-53cfd16f-38c-1-b7eb1a5f-72d84677. To: sip:2.2.2.2:39464. Call-ID: b2033011-c4175f65-f074de1(a)10.9.23.180. CSeq: 1 OPTIONS. Content-Length: 0. . U 2014/07/25 11:26:58.551746 2.2.2.2:39464 -> 10.9.23.180:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0;received=1.1.1.1. Contact: <sip:2.2.2.2:39464>. To: <sip:2.2.2.2:39464>;tag=0867c967. From: <sip:pinger@lwsbc.eu>;tag=uloc-53cfd16f-38c-1-b7eb1a5f-72d84677. Call-ID: b2033011-c4175f65-f074de1(a)10.9.23.180. CSeq: 1 OPTIONS. Accept: application/sdp. Accept-Language: en. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: replaces, eventlist. User-Agent: X-Lite release 4.6.1 stamp 73073 52ef741c-W. Content-Length: 0. . U 2014/07/25 11:27:01.110665 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:01.152416 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:01.611529 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:01.649682 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:02.563132 172.16.52.53:1384 -> 172.16.52.240:5060 . . .............. U 2014/07/25 11:27:02.612548 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:02.629843 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:04.613706 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:04.617573 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:08.614089 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:08.615487 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:12.614697 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:12.616644 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:14.950640 2.2.2.2:39464 -> 10.9.23.180:5060 . . .............. U 2014/07/25 11:27:16.615213 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:16.617182 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:20.615928 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:20.617554 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:24.616354 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:24.617435 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:28.478391 10.9.23.180:5060 -> 2.2.2.2:39464 OPTIONS sip:2.2.2.2:39464 SIP/2.0. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0. From: sip:pinger@lwsbc.eu;tag=uloc-53cfd16f-38c-1-b7eb1a5f-82d84677. To: sip:2.2.2.2:39464. Call-ID: b2033011-d4175f65-d274de1(a)10.9.23.180. CSeq: 1 OPTIONS. Content-Length: 0. . U 2014/07/25 11:27:28.584706 2.2.2.2:39464 -> 10.9.23.180:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0;received=1.1.1.1. Contact: <sip:2.2.2.2:39464>. To: <sip:2.2.2.2:39464>;tag=86694f0d. From: <sip:pinger@lwsbc.eu>;tag=uloc-53cfd16f-38c-1-b7eb1a5f-82d84677. Call-ID: b2033011-d4175f65-d274de1(a)10.9.23.180. CSeq: 1 OPTIONS. Accept: application/sdp. Accept-Language: en. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: replaces, eventlist. User-Agent: X-Lite release 4.6.1 stamp 73073 52ef741c-W. Content-Length: 0. . U 2014/07/25 11:27:28.617317 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:28.618441 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:32.564887 172.16.52.53:1384 -> 172.16.52.240:5060 . . .............. U 2014/07/25 11:27:32.618675 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:32.620366 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. * KAMAILIO CODE: cfg: #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_DEBUG #!define WITH_ACCDB #!define WITH_XMLRPC #!define WITH_MULTIDOMAIN #!define WITH_ALIASDB #!define WITH_ALCATEL ##------!define WITH_PSTN #!define WITH_XHTTP #!define WITH_ANTIFLOOD #------#!define WITH_WEBSOCKETS #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif include_file "modules_lw.cfg" ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # sip_trace(); # setflag(22); #!ifdef WITH_WEBSOCKETS if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } #!endif # per request initial checks route(REQINIT); # NAT detection #!ifdef WITH_ALCATEL if(($rU=~"^25[0-9][0-9]$")) { route(ALCATEL); } #!endif route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # dispatch destinations to ALCATEL OXE route(ALCATEL); # user location service route(LOCATION); route(RELAY); } #!ifdef WITH_WEBSOCKETS route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); rtpproxy_manage("froc+SP"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); rtpproxy_manage("froc-sp"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); rtpproxy_manage("co"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($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 $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } #!endif route[RELAY] { #!ifdef WITH_WEBSOCKETS route(SETUP_BY_TRANSPORT); #!endif # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && is_uri_host_local()) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } xlog("L_NOTICE","#### message flags are: $mf [$mF]\n"); if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ALCATEL # do not auth traffic from ALCATEL - trusted! if(route(ALCATEL)) return; #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || is_from_local()) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!is_from_local() && !is_uri_host_local()) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT xlog("L_NOTICE","####$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); force_rport(); if (!nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); xlog("L_NOTICE","####$rm from $fu (IP:$si:$sp) in Route[NAT] fix_nat-register\n"); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT xlog("####NATMANAGE M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if((src_ip==ALCATEL_LAN) || (src_ip==LONGWAVE_LAN)) { rtpproxy_manage("ie"); xlog("L_NOTICE","####$rm $cs from $fu (IP:$si:$sp) in route[RTPPROXY] RTPproxy with IE Flags\n"); } else { rtpproxy_manage("ei"); xlog("L_NOTICE","####$rm $cs from $fu (IP:$si:$sp) in route[RTPPROXY] RTPproxy with EI Flags\n"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(!is_from_local()) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_NOTICE","####new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies #!ifdef WITH_WEBSOCKETS onreply_route[REPLY_TO_WS] { xlog("L_INFO", "####Reply from softphone: $rs"); if (t_check_status("183")) { change_reply_status("180", "Ringing"); remove_body(); exit; } if(!(status=~"[12][0-9][0-9]")) return; rtpproxy_manage("froc+SP"); route(NATMANAGE); } onreply_route[REPLY_FROM_WS] { xlog("L_INFO", "####Reply from webrtc client: $rs"); if(status=~"[12][0-9][0-9]") { rtpproxy_manage("froc-sp"); route(NATMANAGE); } } #!endif onreply_route[MANAGE_REPLY] { xlog("L_NOTICE","####incoming reply Code: $T_reply_code from IP:$si:$sp \n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } # ====================================================== # Populate CDRs Table of Siremis # ====================================================== route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); } # -- Charts route[STATS] { # clean very old records $var(tmc) = $var(tmc) + 1; $var(x) = $var(tmc) mod 144; if($var(x) == 0) sql_query("ca", "delete from statistics where time_stamp<$Ts - 864000", "ra"); # insert values for Kamailio internal statistics sql_query("ca", "insert into statistics (time_stamp,shm_used_size," "shm_real_used_size,shm_max_used_size,shm_free_used_size," "ul_users,ul_contacts) values ($Ts,$stat(used_size)," "$stat(real_used_size),$stat(max_used_size),$stat(free_size)," "$stat(location-users),$stat(location-contacts))", "ra"); # init the values for first execution, compute the diff for the rest if($var(tmc)==1) { $var(rcv_req_diff) = $stat(rcv_requests); $var(fwd_req_diff) = $stat(fwd_requests); $var(2xx_trans_diff) = $stat(2xx_transactions); } else { $var(rcv_req_diff) = $stat(rcv_requests) - $sht(stats=>last_rcv_req); $var(fwd_req_diff) = $stat(fwd_requests) - $sht(stats=>last_fwd_req); $var(2xx_trans_diff) = $stat(2xx_transactions) - $sht(stats=>last_2xx_trans); } # update the values for stats stored in cache (htable) $sht(stats=>last_rcv_req) = $stat(rcv_requests); $sht(stats=>last_fwd_req) = $stat(fwd_requests); $sht(stats=>last_2xx_trans) = $stat(2xx_transactions); # insert values for stats computed in config sql_query("ca", "update statistics set tm_active=$stat(inuse_transactions)," "rcv_req_diff=$var(rcv_req_diff),fwd_req_diff=$var(fwd_req_diff)," "2xx_trans_diff=$var(2xx_trans_diff) where time_stamp=$Ts", "ra"); } # ALCATEL GW routing route[ALCATEL] { #!ifdef WITH_ALCATEL # check if ALCATEL GW IP is defined if (strempty($sel(cfg_get.alcatel.gw_ip))) { xlog("SCRIPT: ALCATEL routing enabled but alcatel.gw_ip not defined\n"); return; } # route to ALCATEL dialed numbers starting with '25xx' # - update the condition to match your dialing rules for ALCATEL routing if(!($rU=~"^25[0-9][0-9]$")) return; # only local users allowed to call if(!is_from_local()) { sl_send_reply("403", "Not Allowed!!!!!"); exit; } if (strempty($sel(cfg_get.alcatel.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.alcatel.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.alcatel.gw_ip) + ":" + $sel(cfg_get.alcatel.gw_port); } set_advertised_address("10.9.23.180"); route(RELAY); exit; #!endif return; } modules: ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 mhomed=1 log_facility=LOG_LOCAL1 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT #tcp_connection_lifetime=3605 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 server_header="Server: LW-SBC" server_signature=yes user_agent_header="User-Agent: LW-SBC" #!ifdef WITH_XHTTP tcp_accept_no_cl=yes #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLT_OUT 8 #!define ALCATEL_LAN 10.9.6.0/24 #!define LONGWAVE_LAN 172.16.52.0/24 #!substdef "!MY_IP_ADDR_DMZ!10.9.23.180!g" #!substdef "!MY_IP_ADDR_INT!172.16.52.240!g" #!substdef "!MY_DOMAIN!lwsbc.longwave.eu!g" #!substdef "!MY_WS_PORT!10080!g" #!substdef "!MY_WSS_PORT!10443!g" #!substdef "!ALCATEL_IP!10.9.6.3!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR_DMZ:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR_DMZ:MY_WSS_PORT!g" listen=MY_IP_ADDR_DMZ:5060 advertised_address = "1.1.1.1" listen=MY_IP_ADDR_INT:5060 #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif alias = "MY_DOMAIN:5060" alias = "MY_IP_ADDR_DMZ:5060" alias = "MY_IP_ADDR_INT:5060" alias = "MY_DOMAIN:10080" alias = "MY_IP_ADDR_DMZ:10080" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=10080 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "10.9.6.3" desc "PSTN GW Address" pstn.gw_port = "5060" desc "PSTN GW Port" #!endif #!ifdef WITH_ALCATEL # ALCATEL routing # alcatel.gw_ip = "10.9.6.3" desc "Alcatel OXE GW, Longwave" alcatel.gw_port = "5060" desc "Alcatel OXE GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/i386-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "textopsx.so" #!ifdef WITH_WEBSOCKETS loadmodule "websocket.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "siptrace.so" loadmodule "sdpops.so" # -- for siremis CDRs -------------- loadmodule "rtimer.so" loadmodule "sqlops.so" # -- Charts Panel #loadmodule "htable.so" #------- GIA' CARICATO CON L'ANTIFLOOD---------- # -- MI Commands Panel loadmodule "mi_datagram.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy-ng.so" #----loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_XHTTP loadmodule "xhttp.so" #!endif #!ifdef WITH_XHTTP_RPC loadmodule "xhttp_rpc.so" #!endif #!ifdef WITH_XHTTP_PI loadmodule "xhttp_pi.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ------- Modifico il valore di default ( trying -- your call is important to us ) modparam("tm", "auto_inv_100_reason", "LW-SBC is Trying") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy-ng", "rtpproxy_sock", "udp:127.0.0.1:12222") #----modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@lwsbc.eu") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "a=>size=8;autoexpire=920;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #!ifdef WITH_XHTTP_RPC modparam("xhttp_rpc", "xhttp_rpc_root", "http_rpc") #!endif #!ifdef WITH_XHTTP_PI modparam("xhttp_pi", "xhttp_pi_root", "http_pi") modparam("xhttp_pi", "framework", "/etc/kamailio/pi_framework.xml") #!endif # -- for siremis CDRs -------------- modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@localhost/kamailio") # -- For CHARTS modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://kamailio:kamailiorw@localhost/kamailio") modparam("htable", "htable", "stats=>size=6;") # -- MI Commands Panel modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8033") # ----- siptrace params ----- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_on", 1) #modparam("siptrace", "trace_sl_acks", 1) #modparam("siptrace", "trace_local_ip", "10.9.23.180:5060") /etc/default/ngcp-rtpengine-daemon RUN_MEDIAPROXY=yes LISTEN_TCP=127.0.0.1:25060 LISTEN_UDP=127.0.0.1:22222 LISTEN_NG=127.0.0.1:12222 ADDRESS=10.9.23.180 ADV_ADDRESS=1.1.1.1 # ADDRESS_IPV6=... # ADV_ADDRESS_IPV6=... TIMEOUT=60 SILENT_TIMEOUT=3600 PIDFILE=/var/run/ngcp-rtpengine-daemon.pid FORK=yes # TOS=184 TABLE=0 NO_FALLBACK=yes PORT_MIN=30000 PORT_MAX=50000 # REDIS=127.0.0.1:6379 # REDIS_DB=1 # B2B_URL=http://127.0.0.1:8090/ LOG_LEVEL=7 Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Complex-NAT-Scenario-HELP-tp129226.… Sent from the Users mailing list archive at Nabble.com.

2 5

sdpops and rtpproxy/rtpengine
by Alex Balashov 08 Aug '14

08 Aug '14

Hi, Is it safe to use sdpops and the rtpproxy* modules together, given that the latter modify SDP? In other words, can I prune some codecs first, and then call rtpproxy_offer()? Is it safe? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 4

SDPOPS issue or append_hf
by Igor Potjevlesch 07 Aug '14

07 Aug '14

Hello, I have an issue with the module SDPOPS while using "sdp_keep_codecs_by_name". If the calling party sends only one codec description like: Content-Type: application/sdp Content-Length: 202 v=0 o=UserA 2966746938 1790378070 IN IP4 10.141.0.21 s=Session SDP c=IN IP4 10.141.0.21 t=0 0 m=audio 49152 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 The result of the function "sdp_keep_codecs_by_name("PCMA,PCMU,G729a");" is: Content-Type: application/sdp Content-Length: 170 P-Asserted-Identity: "+0123456789" <sip:+0123456789@sip.tld> v=0 o=UserA 2485672881 3000549892 IN IP4 a.b.c.d s=Session SDP c=IN IP4 a.b.c.d t=0 0 m=audio 40330 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=nortpproxy:yes If I open the capture in Wireshark, the PAI is not in the SDP part, and the end of the capture after "a=rtpmap:8 PCMA/8000" is seen as "Data (18 bytes)". I don't understand why the PAI is inserted within the SDP part. Adding the PAI is done after "sdp_keep_codecs_by_name": if (!is_present_hf("P-Asserted-Identity")) { $var(pai) = $(fU{re.subst,/^0/+33/g}); append_hf("P-Asserted-Identity: \"$var(pai)\" <sip:$var(pai)@$fd>\r\n"); } I guess that this cause my INVITE being dropped by 488 Media Not Acceptable Here. Regards, Igor.

2 6

SIP over WebSocket
by Azamat Ailbayev 07 Aug '14

07 Aug '14

Hello, I have a question regarding to SIP over WebSocket implementation. I have changed kamailio.cfg file accordingly with websocket configurations. However, when I am trying to run Kamailio, now I am getting 3 errors on the same line which is "syn_branch = 0". What could be the issue? And How it is possible to fix it? Thank you in advance. BR, Azamat Ailbayev, Kazakhstan

2 1

planning the release of v4.1.5
by Daniel-Constantin Mierla 07 Aug '14

07 Aug '14

Hello, I am considering to release v4.1.5 sometime next week, most likely on the 6th of August. Checking the 4.1 branch, there are not many fixes, few are on my list for backporting. That's good, indicating a high level of stability. If anyone is aware of issues not reported on tracker or patches that have not been backported, add to the tracker or write a message to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 5

Kamailio v4.1.5 Released
by Daniel-Constantin Mierla 06 Aug '14

06 Aug '14

Hello, Kamailio SIP Server v4.1.5 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.4. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.5. For more details about version 4.1.5 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/08/kamailio-v4-1-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio with open ims
by Senthil K 06 Aug '14

06 Aug '14

Hi All, i have a query here, the user is registered with the open IMS and also with Kamailio ( as a third party registration via SCSCF ) Here my issue is ; when the user makes a call, the INVITE reaches to kamailio from SCSCF( Scscf port is 6060) , when the Kamailio sends the INVITE back to IMS, it sends to the port 5060. problem is , the kamailio registers the SCSCF port as 6060, but when sending the INVITE back it uses the port 5060. why this is So ?? i checked in registered entry as well, it shows that the contact is SCSCF with 6060 port only. please help in this situation, Kind regards SenthilK

2 2

Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
by Olli Heiskanen 06 Aug '14

06 Aug '14

Hello, I've been experimenting with Kamailio with ws and sip clients and could need a hand in getting a call between those two to work. I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS 6.5 and a mediaproxy-ng running. I have clients wsclient(a)testers.com and gsclient(a)testers.com and I try to make call from wsclient to gsclient. The wsclient is a jssip client running on chrome and gsclient is a grandstream desk phone. My config file is the default one enhanced by online examples. I use a html5 <audio> element for the media streams, and configured my jssip phone to accept audio options like this: var options = { 'eventHandlers': eventHandlers, 'mediaConstraints': {'audio': true, 'video': false } }; sipUA.call(callto, options); I used the instructions from here: http://www.slideshare.net/crocodilertc/webrtc-websockets What I get is gsclient ringing, and as I answer there is no audio and call hangs up in a few seconds. I guess this is a SDP problem, something between Kamailio and Mediaproxy-ng but SDP is not my strong point so I'd appreciate advice. Question is where's my misconfiguration/problem? I would like to learn why this problem occurs and how to fix it rather than getting a solution right away, but please bear in mind I don't know much about SDP. In Kamailio log I see: kamailio[27059]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Error rewriting SDP kamailio[27058]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id kamailio[27057]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id Following are the INVITEs and 200 OKs from my SIP trace (1.1.1.1 is the ip of my Kamailio & mediaproxy-ng box and 2.2.2.2 is the public ip behind which both my clients are). The gsclient has port 5066. ****************************************************************************** U 2014/04/01 20:03:41.060009 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 2211. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/SAVPF 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=candidate:2999745851 1 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:2999745851 2 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:3350409123 1 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:3350409123 2 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0. a=candidate:4233069003 2 tcp 150995 T 2014/04/01 20:03:41.119806 2.2.2.2:38986 -> 1.1.1.1:5060 [A] ...... U 2014/04/01 20:03:41.159086 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Warning: 304 GS "Media type not available". Content-Length: 0. . U 2014/04/01 20:03:41.159392 1.1.1.1:5060 -> 2.2.2.2:5066 ACK sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Max-Forwards: 16. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 ACK. Content-Length: 0. . U 2014/04/01 20:03:41.161085 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 3136. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/AVP 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=fingerprint:sha-256 72:54:87:EC:D2:4C:D1:70:C2:FE:69:08:20:5C:92:1D:E0:EA:BD:45:09:E0:90:62:27:B6:34:60:54:E2:99:28. a=setup:actpass. a=mid:audio. a=sendrecv. a=rtpmap:111 opus/48000/2. a=fmtp:111 minptime=10. a=rtpmap:103 ISAC/16000. a=rtpmap:104 ISAC/32000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:106 CN/32000. a=rtpmap:105 CN/16000. a=rtpmap:13 CN/8000. a=rtpmap:126 telephone-event/8000. a=maxptime:60. a=ssrc:3298511848 cnam And here are the 200 OK messages when answering the call: U 2014/04/01 20:03:46.049711 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. T 2014/04/01 20:03:46.051127 1.1.1.1:5060 -> 2.2.2.2:38986 [AP] .~.dSIP/2.0 200 OK. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. ****************************************************************************** cheers, Olli

4 26

no fork mode and more than one listen address found (will use only the first one)
by Yuriy Gorlichenko 06 Aug '14

06 Aug '14

Hello. I Installed kamailio 4.1.4, when I starting my server I see following Warning no fork mode and more than one listen address found (will use only the first one) So I see only first listening port at my netstat At my kamailio.cfg file I write fork=yes? but it not helps me. I found the same issue at this mailing list http://comments.gmane.org/gmane.comp.voip.openser.user/10686 But it very old issue (2007 year) and I think patch at this issue does not help me. How I can start my Server for runnings with many interfaces?

3 5

Understanding DNS resolution done by Kamailio
by Igor Potjevlesch 05 Aug '14

05 Aug '14

Hello, I'm using Kamailio 4.1 and I'm wondering how to avoid external DNS resolution. I have the following config: dns_cache_init=no use_dns_cache=no dns=no rev_dns=no Even with this config, I have many and many DNS query on SRV _sip for the hostnames set in carrierroute module. These requests are not useful because the A resolution is done by /etc/hosts. Major problem with this, is that when I have a DNS issue or IP transit issue, Kamailio waits for resolution timeout and becomes overloaded. As a consequence, Kamailio can't treat others SIP requests like REGISTER because he stuck in DNS resolution. Regards, Igor.

2 4

[rtpengine] continuous 'Successful STUN binding request from' messages
by Paul Belanger 05 Aug '14

05 Aug '14

I was hoping somebody could confirm the following is a _normal_ log file for rtpengine. Specifically I am curious of the 'Successful STUN binding request from' messages are continuously logged. Are they part of the normal operation or does this indicate some issue with media? I only ask because I am getting one way audio at the moment. I get audio from rtpengine however rtpengine does not hear me. http://pastebin.com/hztxMFL3 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 3

rtp relaying and cseq - rtpengine/rtpproxy
by Daniel-Constantin Mierla 05 Aug '14

05 Aug '14

Hello, apparently none of the rtp relays I was looking at (rtpproxy and rtpenging) are taking in consideration the CSeq value. Quite recently I met a case with a b2bua that sends the invite without sdp, then after its ACK with SDP sends quickly a re-invite to get itself out of media stream. However, due to network and parallel processing races, the ACK gets processed after the re-INVITE, updating the rtp relay session with attributes that should be no longer used. The impact is practically no audio. I got it working using the 'brave' htable and tracking cseq numbers, skipping calling the relay for lower cseqs, but I wonder if wouldn't be better to push it to the relay, to avoid also races on the network between kamailio and the relay. I would be also curious to hear if others were facing such issues. 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

Modify Refer-To header
by Stefan Kohlhauser 05 Aug '14

05 Aug '14

2 1

Re: [SR-Users] missing BYE record on acc table when usingtimeout_avp
by betul.cerit＠netgsm.com.tr 05 Aug '14

05 Aug '14

Hi, In ‘event_route[dialog:end]’I can not get any variable they are all empty but using acc_db_requests() worked for me. Thanks for your help. BEST REGARDS From: Daniel-Constantin Mierla Sent: Monday, August 04, 2014 5:28 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] missing BYE record on acc table when usingtimeout_avp Hello, with latest version you can get the cdrs at once via dialog+acc module. If you want to use the classic acc, then add either event_route[dialog:end] or event_route[tm:local-request]. There you can get actions when the dialog is ended (including the time out), and for the second you get the BYE sent out. Perhaps SQLOPS with sql_query() or acc_db_requests() should be used to record the event in db table. Cheers, Daniel On 04/08/14 15:19, betul.cerit(a)netgsm.com.tr wrote: Hi to all, I want to limit call duration according to the credit of the user.To accomplish this I use timeout_avp .When the limit is reached call ends but no record associated with BYE method is inserted to the acc table and that makes impossible to bill that call.What can I do to solve this problem or is there an alternative way to limit call duration?My configuration and logs are as follows. BEST REGARDS. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif -------------------------------------------------------------------------------- route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } -------------------------------------------------------------------------------- Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 kamailio /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012! _______________________________________________ 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://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -------------------------------------------------------------------------------- _______________________________________________ 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

2 1

[rtpengine] DTLS error: 1 (message digest is null)
by Paul Belanger 04 Aug '14

04 Aug '14

Greetings, I'm having some trouble getting dtls-srtp working with kamailio 4.1 (mediaproxy-ng) and rtpengine (master). I believe I finally have my branch logic setup properly in kamailio, however when the calls get placed into rtpengine, it appears DTLS is not initialized properly. Attached is my debug log file. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 3

4.1 Compilation issue
by Gilbert T. Gutierrez, Jr. 04 Aug '14

04 Aug '14

CentOS6.5 Kamailio 4.1 I compiled Kamailio as follows so that the configuration file would be in "/etc/kamailio" rather than "/usr/local/etc/kamailio/"... make cfg_prefix=/ all make cfg_prefix=/ install When I try to start kamailio I get the following error... Not starting : invalid configuration file! ERROR: loading config file(/usr/local/etc/kamailio/kamailio.cfg): No such file or directory I can find no reference in the init file to that location (which does not exist). Did I use the wrong options in make? Gilbert

3 3

missing BYE record on acc table when using timeout_avp
by betul.cerit＠netgsm.com.tr 04 Aug '14

04 Aug '14

Hi to all, I want to limit call duration according to the credit of the user.To accomplish this I use timeout_avp .When the limit is reached call ends but no record associated with BYE method is inserted to the acc table and that makes impossible to bill that call.What can I do to solve this problem or is there an alternative way to limit call duration?My configuration and logs are as follows. BEST REGARDS. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif -------------------------------------------------------------------------------- route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } -------------------------------------------------------------------------------- Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 kamailio /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012!

2 2

LM_ERR and LM_DBG
by aawaise 04 Aug '14

04 Aug '14

I want to ask some queries regarding logging in kamailio. 1. First of all if I want to add LM_DBG or LM_ERR in lookup.c file of registrar module, where to add the command as on extraction of kamailio-3.1.1.zip, I found registrar module files in modules_k folder as well as modules_s folder. 2. I was advised to us macros from dprint.h to see LM_DBG and LM_ERR statements in log file, how to do so ?? 3. Third, will these logs also print in /var/log/messages in debian linux along with other kamailio.cfg logs ?? 4. What changes should be made in kamailio.cfg to see these logs, i.e log level etc ?? Any help will be highly appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/LM-ERR-and-LM-DBG-tp129392.html Sent from the Users mailing list archive at Nabble.com.

2 1

See XDBG and XERR Messages in Logs
by aawaise 04 Aug '14

04 Aug '14

I have added some XLOG and XDBG commands in registrar module of kamailio. I want to see messages in result of these commands being printed in logs to make sure that changes are being made surely. How can it be done ? I am using fedora and I see kamailio related logs in /var/log/messages.log -- View this message in context: http://sip-router.1086192.n5.nabble.com/See-XDBG-and-XERR-Messages-in-Logs-… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration - Problem
by Reza Kiani Moghaddam 03 Aug '14

03 Aug '14

Hi, I Installed Kamailio & Asterisk on Ubuntu 14.04 LTS, and both of them started successfully. I can register with Jitsi, But I can't call other users.. I don't know what to do ...

2 1

Kamailio behind NAT- alias issue
by Yuriy Gorlichenko 03 Aug '14

03 Aug '14

Hello. I have Kamailio running behind NAT. It lesten eth0 with ip 192.168.0.3 and I have external IP that have domain name (for example sip.myserver.com). Register packets from clients comes from external IP. If I write at kamailio.cfg: alias=sip.myserver.com I see error at log - bad_uri sip.myserver.com I think it happens because kamailio does not know atything about external ip pecause kamamiliol working with server interfaces (eth0, eth1, etc.) So my question - how to listen external IP with domain name on kamailio that running behind NAT? P.S. I do not have any access to router, that present NAT for me.

2 2

SIPp load testing
by bob sacamano 02 Aug '14

02 Aug '14

I am trying to follow the guide shown here:http://www.kamailio.org/docs/openser-performance-tests/ to load test my kamailio system.I am a little unclear as to how many instances of sipp are running. For the first part i see the command ./sipp -sf uac_msg.xml -rsa 192.168.2.102:5060 192.168.2.102:5070 -m 200000 -r 10000 -d 1 -l 70for generating the UAC part but is there another instance of sipp running on the kamailio computer acting as a UAS? I have seen come sites use for example: sipp 192.168.1.100:5060 -sn uas -p 5060 but is this not necessary? thanks for any help

2 3

Sample Kamailio Config for Siremis
by Noah Mehl 02 Aug '14

02 Aug '14

I have managed to install Kamailio and Siremis 4.x, but I have noticed that neither Kamaliio or Siremis come with a sample config that makes sense for Siremis (as far as I can tell). For instance, if I want to use LCR with dispatchers, the sample Kamilio configs don’t have the dispatcher module loaded or configured. Am I missing something? Can someone please share theirs? Thanks! ~Noah

3 2

ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene>
by Rene Montilva 01 Aug '14

01 Aug '14

Hi list i've been working with t_suspend, t_continue and async sleep, now the problem it's with register users, the softphones are frozen when try to establish a session with kamailio, now my users can't register and the only error i got is this: ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene> my kamailio config: # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 30) # max value for expires of registrations #modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "hash_size", 15) #!endif

1 0

kamailio routes packets with invalid From/To headers with uac.restore_mode=auto when incoming packet does not use exact same replaced From/To header
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am currently handling a system that runs kamailio and asterisk in the same machine. The kamailio instances are being used to emulate multiple SIP domains, by means of From/To mangling of incoming packets, which are then routed to Asterisk. The attached kamailio.cfg does this work. There is an problem when handling SUBSCRIBE requests (as required for BLF and voicemail indications). My configuration is written so that these SUBSCRIBE requests are not handled by kamailio, but instead routed to asterisk. There is a failure to check From/To headers to see whether NOTIFY packets generated as part of a subscription can be restored using the information in Record-Route. The end result is that kamailio ends up sending packets with garbled tags that are (rightly) rejected by the SIP endpoint. The following is an example that demonstrates the issue (using Jitsi as endpoint): After registration, Jitsi sends a SUBSCRIBE request: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com> Max-Forwards: 70 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Proxy-Authorization: Digest username="avillacisIM",realm="pbx.villacis.com",nonce="U9lZJlPZV/r06Xep/ukc1UzAIO0V3TbS",uri="sip:avillacisIM@pbx.villacis.com",response="0e18f4913c2693f6154c91f158fb17fe" Content-Length: 0 This packet is mangled by the configuration, and is sent to asterisk like this: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> Max-Forwards: 69 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Content-Length: 0 The asterisk response for the SUBSCRIBE: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080>;expires=3600 Content-Length: 0 This is in turn transformed back by kamailio, and sent to Jitsi like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080;alias=127.0.0.1~5080~1>;expires=3600 Content-Length: 0 Now asterisk wants to send a NOTIFY to the endpoint for the subscription. The NOTIFY looks like this: NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport Max-Forwards: 70 Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes>,<sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Here is where the bug appears. The autoprocessing does not recognize that the From header (From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e) from the above request has nothing to do with the saved information (vsf parameter). Instead, it blindly mangles the From header, and does not even run a sanity check on the result before routing it. The end result is shown below. NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as5562e95e> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as5562e95e> Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bK8333.8bfe7bc2bd554a8631f0d00d463b28ee.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport=5080 Max-Forwards: 69 From: "asterisk" <sip:asterisk@12(.0.0.1:5080.....@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) From examination of the source code, the vsf and vst strings are base64 encodings of the result of XORing the byte strings of the old and new tags. For this to work, the headers of future packets should match. However, here kamailio does not realize that the header does not match (by the ftag), and also does not check that the resulting "restored" header is a valid header.

2 2

SCA module - sca_call_info_update() function for 180 responses
by João Vitor Arruda 01 Aug '14

01 Aug '14

Hello all, I've had proposed a fix to SCA module in the tracker ( https://sip-router.org/tracker/index.php?do=details&task_id=446) but did´t hear anything about it until now. I would appreciate some comments on that to see if its feasible to be committed or even hear for anyone else using SCA module with Polycom phones to see if they have the same issue described. Thanks, João Vitor Arruda

4 5

Kamailio V4.1.3 - Dialog timeout -ACC ERROR
by Yasin Caner 01 Aug '14

01 Aug '14

Hello; i am trying to limit call duration with dialog module. So i use *timeout_avp *for limiting it. there is no problem to close calls but when it happens i got some errors and kamailio can not write Bye result in DB. There is code sample and result in below. 2. question : is there a big difference between dialog_ng and dialog? Which one is more stable? 3. question : After restarting kamailio , Dialog hash vanish so timeout isn't working. Can DB mode solve this problem? PS: event_route[dialog:end] isn't working in Kamailio V 4.1.4. I got some error like t_w_relay. Thanks for helps. ------------------------------------------------------------------------ #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif ------------------------------------------------------------------------ route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } ------------------------------------------------------------------------ Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 host-91-93-189-136 /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012!

1 0

sip
by Nar 01 Aug '14

01 Aug '14

Hi, I am developing a social application where I want to enable voice & video calling function and still I don't know what to use in server side, do you have any product that I can use to develop the server side(for 10.000+ users)? Thank you in advance.

2 1

[Kamailio]list members
by Djamel Bahamid 01 Aug '14

01 Aug '14

Hi, I would like to subscribe to the list. Best Regards, Djamel BAHAMID.

2 1

SIGUSR1 for memory status not working as documented - only one process reports back
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once memlog is set in the configuration file, a kamailio process will dump a report of the allocation map when shutting down, or when receiving a SIGUSR1. I have configured my kamailio.cfg with memlog=1 and no other change, and I see the memory report on shutdown for all processes. However, when I send a SIGUSR1 to a kamailio process, the process does absolutely nothing, with the exception of the first kamailio process (the one reported as Type=attendant by "kamctl ps"). All of the other processes just ignore SIGUSR1. What is going on? It is inconvenient to force a shutdown of all the kamailio processes just to get the memory report.

2 1

Unable to SUBSCRIBE for presence using sip.js through WSS
by Alex Villacís Lasso 31 Jul '14

31 Jul '14

My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using SIP.js via a websocket, Kamailio is unable to send the NOTIFY for the presence event. However, MESSAGE packets are routed correctly between peers, even through websockets. The location table after registration looks like this: +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | id | ruid | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods | instance | reg_id | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | 192 | uloc-53d96398-1afc-1 | admin | pbx.villacis.com | sip:nfinhooe@192.0.2.90;transport=ws | sip:192.168.3.2:41478;transport=WS | NULL | 2014-07-30 17:18:57 | -1.00 | 6a2dot9ihoobhng9ackqmq | 90 | 2014-07-30 17:08:57 | 0 | 64 | SIP.js/0.5.0 | tls:192.168.2.18:5061 | 782 | <urn:uuid:b8888e95-7833-46a4-a21e-8d2e809a66b8> | 1 | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ When running the following javascript code: var subs = ua.subscribe('admin(a)pbx.villacis.com', 'presence'); subs.on('notify', function (n) { alert(n.request.body); }); ... I see the following traffic in the browser console area: SUBSCRIBE sip:admin@pbx.villacis.com SIP/2.0 Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279 Max-Forwards: 70 To: <sip:admin@pbx.villacis.com> From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Proxy-Authorization: Digest algorithm=MD5, username="admin", realm="pbx.villacis.com", nonce="U9lswVPZa5WSu5jz1P/jIOJTvthKXw6v", uri="sip:admin@pbx.villacis.com", response="f7246e64573b5888d69896a70d4204aa" Event: presence Expires: 3600 Contact: <sip:nfinhooe@192.0.2.90;transport=ws> Allow: ACK,CANCEL,BYE,OPTIONS,MESSAGE Supported: outbound User-Agent: SIP.js/0.5.0 Content-Length: 0 SIP/2.0 202 OK Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279;rport=41478;received=192.168.3.2 To: <sip:admin@pbx.villacis.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-84c3 From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.2.18:5061;transport=ws> Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 However, I also get the following messages in the kamailio log: Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription I have found a discussion at http://marc.info/?l=sr-users&m=136624316816288&w=2 which recommends fix_nated_contact(), but I am not really sure about how to add it, or what conditions to check on whether it should be used. I added the call to fix_nated_contact() like this: # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); # # 1 - Contact header field is searched for occurrence of RFC1918 or rfc6598 addresses. # 2 - the "received" test is used: address in Via is compared against source IP address of signaling # 16 - test if the source port is different from the port in Via # 64 - test if the source connection of signaling is a WebSocket # ----- # 83 #!ifdef WITH_WEBSOCKET # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. if (nat_uac_test("83")) { #!else if (nat_uac_test("19")) { #!endif if (is_method("REGISTER")) { fix_nated_register(); #!ifdef WITH_WEBSOCKET if (nat_uac_test("64")) fix_nated_contact(); #!endif } else { if (is_first_hop()) if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); } #!endif return; } After registration, "kamctl ul show" reports this: [root@elx3 js]# kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: admin(a)pbx.villacis.com Contact:: sip:85017jar@192.168.3.2:41777;transport=ws Q= Expires:: 584 Callid:: argl25kg0r4qe8lbtbhn9f Cseq:: 82 User-agent:: SIP.js/0.5.0 Received:: sip:192.168.3.2:41777;transport=WS State:: CS_NEW Flags:: 0 Cflag:: 64 Socket:: tls:192.168.2.18:5061 Methods:: 782 Ruid:: uloc-53d971c7-297f-1 Instance:: <urn:uuid:75464840-96b8-4e22-8eaa-5012ba327086> Reg-Id:: 1 Last-Keepalive:: 1406759390 Last-Modified:: 1406759390 However, it still does not work: Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription

1 2

Obscuring SIP traffic and using with NoSIP
by Muhammad Shahzad 31 Jul '14

31 Jul '14

Hi, As the mobile voip is getting more and more popular these days, there has been a strong opposition from GSM operators against mobile voip apps. They often use tactics like blocking voip ports, or detect and block voip traffic and in some cases restricting udp traffic altogether to very low upload and download speeds. See below link for some details, http://www.linphone.org/eng/blog/linphone-over-3g.html While not all the problems can be solved right now (especially the limiting udp traffic, since RTP always uses udp transport) I was wondering if we can at least handle the sip related problems. The most important of them is SIP traffic detection. While some forks would suggest using TCP/TLS to encrypt SIP traffic, it has a few problems, e.g. 1. It requires somewhat high resources on mobile devices, so many low-end android phones simply can't use it. 2. There is possibility that encryption signature may identify it as SIP traffic. There exists firewalls (often deployed in middle eastern countries) which have huge database of encryption signatures and patterns which although may not decrypt the sip packet but at least identify it as sip packet and block it. Also with rough agencies of evil empires spying over millions of users worldwide makes the current encryption standards pretty much pointless, at least in terms of user privacy and network security. So there is a strong need to experiment with new ideas and concepts to regain internet freedom. Some of such ideas are, 1. Convert sip traffic which is plain text to binary format just before transmitting it and revert it to plain text upon reception. 2. XOR the sip traffic (pretty much same as binary sip). 3. Use some very lightweight but effective / non-standard encryption algorithm, e.g. https://github.com/mshary/itv All these ideas require that SIP server such as Kamailio is able to adopt to these, preferably with minimal or no change in native code. The NoSIP module seems an interesting module in this regard. It provides all traffic it thinks is not the SIP traffic to configuration script, where we can do our own parsing and do whatever we want with it. I have two questions about this, 1. If parsed message is SIP, we can we send it back to kamailio core to get it processed as if it is a normal SIP message received by kamailio? 2. Can this module or any other module available in kamailio, that can provide us full sip packet that is about to be transmitted over sip socket, so we can "encode" it just before it is sent to next hop? I know this would be like writing a SIP transport in kamailio script which would be very tough if not impossible to implement in native core. But it will really help in winning the modern mobile voip challenges. Thank you.

3 8

[Serusers] how to enable logs in SER
by pawan 31 Jul '14

31 Jul '14

Sir pl help for registration of soft phone to my local sip server my email vijay8538(a)gmail.com --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

1 0

multi-homed rtpengine
by Paul Belanger 30 Jul '14

30 Jul '14

Greetings, We have had good success in the past using rtpproxy with internal and external interfaces however, we are attempting to migrate to rtpengine now but are running into an issue (apparently this feature is not supported). So, my question is as follows, does this mean we need to multiple rtpengine and bind them to each interface? Additionally, in this setup can audio be bridged between the 2 rtpengine processes? If not, how do people over come this difference? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

1 0

Re: [SR-Users] LCR
by Keith 30 Jul '14

30 Jul '14

Hi, I want to do LCR based on from uri. I get I need to do regular expression matching but have no idea where to look to start. Any pointers would be good. Thanks Keith

3 2

Warning message on SIP websocket
by Alex Villacís Lasso 29 Jul '14

29 Jul '14

Just a quick question. What is supposed to be wrong with the below message? Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message (offset: 22) Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Via: SIP/2.0/TLS 192.168.3.2:58657#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: Er/CgVyBNzVGREBmt+w3FTff8s4=#015#012Server: kamailio (4.1.4 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012>

2 1

Plz help: MSILO on TLS + cert based client auth enabled server
by Marc M. 29 Jul '14

29 Jul '14

Hi, I have a SIP server with TLS + client certificate based authentication. The server is listening for SIP(UDP) on port 5060, and for SIPS(TLS) on port 5061) The client certificate based authentication is enforced on port 5061. I would like to fire up MSILO and face now a strange problem: At each REGISTER, the stored messages get dumped. Dumped messages get sent to the clients from senders logged in over port 5060(UDP) as SIP:xxxxxxxxx@xx.xx.xx.xx however Dumped messages are NOT sent to clients from senders logged in on port 5061(TLS) as SIPS:xxxxxxxx@xx.xx.xx.xx It looks like the messages get dumped so M-dump() is working, but they will never enter the route. I would assume TLS is stopping the dumped messages to entering the route logic. I would either 1. somehow force m_dump() to pipe the messages over SIP(and not SIPS), or 2. disable client auth for messages recieved from local ip Can you help me how to proceed? What would be the correct approach? Can you help me with either 1 or 2. You help is greatly appreciated!!! Marc

2 1

Only a redirect server
by Jack Smith 29 Jul '14

29 Jul '14

Dear friends, Just new to this mailing list, new to Kamailio, and new to running a sip server. I would like to try only a redirect server. I had a look at http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg which was friendly, but then the default file kamailio.cfg looked frightening ! Could someone help me set up a minimal Kamailio installation only to handle state-less redirect? Best, Jack

4 4

append_branch and t_replicate issue
by Asgaroth 28 Jul '14

28 Jul '14

Hi All, I have an issue that I cant seem to get to the bottom of, I would appreciate if someone could point me in the right direction. I have 3 registrar's and would like to replicate the registration regest from 1 to the other 2, I am trying this with t_replicate, the docs say that to send to multiple destinations an append_branch should be performed prior to the t_replicate, which is what I am trying to do. What appears to happen is that the replication only sends the REGISTER to the server defined in the t_replicate command, it doesnt appear to be sending it to the destination defined in append_branch. Kamailio Version: version: kamailio 4.1.4 (x86_64/linux) 84c1ff flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 84c1ff compiled on 14:36:04 Jul 20 2014 with gcc 4.4.7 Here is what I have tried so far (on registrar 1): #!define BACKUP_REGISTRAR_1 '1.1.1.1' #!define BACKUP_REGISTRAR_2 '1.1.1.2' #!define BACKUP_REGISTRAR_3 '1.1.1.3' save("location"); if ( src_ip != BACKUP_REGISTRAR_2 && src_ip != BACKUP_REGISTRAR_3 ) { append_branch("sip:" + BACKUP_REGISTRAR_2 + ":5060"); t_replicate("sip:" + BACKUP_REGISTRAR_3 + ":5060"); }; With the above in place, I only see the system attempting to replicate to BACKUP_REGISTRAR_3, I dont see any request to BACKUP_REGISTRAR_2. Am I doing something wrong here? Thanks Bruce

4 29

possible regression on 4.1.3 vs 4.1.4
by Kelvin Chua 26 Jul '14

26 Jul '14

using the default cfg file, whenever a WITHINDLG BYE is received, 4.1.4 needs a retransmission before passing it through, the first BYE would generate this in syslog tm [tm.c:1518]: _w_t_relay_to(): ERROR: w_t_relay_to: unsupported route type: 64 and the 2nd retransmitted BYE would go on fine. i believe route type 64 is branch failure, but using the same cfg and reverting back to 4.1.3 solves it. so whatever changes was done to 4.1.4 broke it. Kelvin Chua

2 4

Rtpengine vs. TURN?
by Peter Villeneuve 25 Jul '14

25 Jul '14

Hi, On my server, I have the option of using either Rtpengine for NAT traversal or pure TURN without rtpengine. Rtpengine has the obvious plus that it only needs 1 public IP, while TURN (with STUN) will need 2 public IPs, although that's not a problem in my case. Having said that, I'd like to take advantage of the huge experience that users of this list have in real world deployments. in your experience, which option is more reliable in a real world deployment? Cheers, Peter

2 3

MSILO not checking Expires header if Allow: is absent and Contact: is present but without expires
by Anthony Messina 25 Jul '14

25 Jul '14

In setting up Kamailio with MSILO, I've noticed that some sip clients (I'm using CSipSimple), upon un-REGISTER-ing do not send an Allow header, but do send an "Expires: 0" header as well as a Contact header. However as the Contact header doesn't contain "expires=0", it appears as though the MSILO skips over checking the Expires header. As such, I've had to add the following workaround to my route[REGISTRAR] block or else these clients trigger the m_dump. # Ensure we aren't unregistering # Right now, CSipSimple doesn't send an Allow header with an un-REGISTER, # so Kamailio skips to checking the Contact header for expires, and appears # to ignore the fact that an Expires: 0 header is present. if($hdr(Expires) != 0) m_dump(); Looking through the msilo check_message_support function, it appears that it never checks for the Expires header if there is no Allow header and the Contact header does not specify an expire setting. I do believe that we're missing the check for the Expires header, regardless of whether or not there is an Allow header, or a Contact header with expires=0 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 4

407 Proxy Authentication Required with Cisco
by proLogika 25 Jul '14

25 Jul '14

Hello, I am using Kamailio as SIP register with asterisk integration describe from hire: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . asterisk is listen on public ip: 2.3.4.5:5060 kamailio is listen on public ip: 1.2.3.4:5060 everything is working well except some software VoIP clients (like Yate) and CISCO phone like Cisco-CP7940G/8.0 and the new one from Cisco series. I’m testing now with Yate client and Cisco. They are register OK but when a call is made Kamailio is answer back with 407 Proxy Authentication Required. When I register Yate or Cisco to asterisk directly the call is passing normaly. I was trying to manipulate kamailio.cfg and more specifically the part: #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; If i commented out this part the call is passing, but I do not have auth anymore (everyone can register) Here is ngrep: U 2014/07/23 19:17:08.108458 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.108805 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. Proxy-Authenticate: Digest realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I". Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.128626 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. User-Agent: YATE/5.3.0. Content-Length: 0. . U 2014/07/23 19:17:08.129076 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.129622 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.130107 1.2.3.4:5060 -> 2.3.4.5:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Record-Route: <sip:1.2.3.4;lr=on;ftag=838449717;nat=yes>. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Content-Type: application/sdp. Content-Length: 499. . v=0. o=yate 1406132227 1406132227 IN IP4 1.2.3.4. s=SIP Call. c=IN IP4 1.2.3.4. t=0 0. m=audio 21888 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. a=nortpproxy:yes. U 2014/07/23 19:17:08.130593 2.3.4.5:5060 -> 1.2.3.4:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0;received=1.2.3.4;rport=5060. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.130770 1.2.3.4:5060 -> 2.3.4.5:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Content-Length: 0. . U 2014/07/23 19:17:08.131361 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.149847 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. User-Agent: YATE/5.3.0. Content-Length: 0. — And kamailio.cfg attached: proLogika Sent with Airmail

2 1

MESSAGE forking bug(?)
by Allen Zhang 24 Jul '14

24 Jul '14

Hi, I believe there is a bug on MESSAGE forking. Test scenario: User A has two records in the location table, with different instance ids: 111 and 222. One of user A's instance is killed, hence 222 is not reachable. User B send a MESSAGE to user A. Kamailio forked the MESSAGE. Kamailio got a '200 Message delivered' response from 111 and forwarded it to user B. The MESSAGE forked to 222 timed out. Kamailio send a '408 Time out' to user B. This violates RFC 3428 section 4. It states that the forking UAS should only send ONE final response to the sender UAC. Forking is done by t_relay(), and the registrar parameter 'append_branches' is set to default value 1. Did I do something wrong, or there is a bug? Regards, Allen

2 5

Late parallel forking
by Guillaume Bour 24 Jul '14

24 Jul '14

Hi, I would like to do some kind of "late parallel forking" (see the example below). Is there a way to do such thing with Kamailio ? Here is an example: . A call B . B device 1 is connected so INVITE is forwarded to it . B device 2 registers succesfully . as device 1 as not yet answered and INVITE transaction is not expired, I want INVITE message to be forwarded to device 2 (so both are ringing) Regards, Guillaume Bour

4 6

Outbound registration?
by James Cloos 23 Jul '14

23 Jul '14

I don't see anything relevant in the docs, nor greping the src, but in case I missed something: Does the core, or do any of the modules support originating registration requests to other proxies a/o endpoints? I'd like to move registration responsibility to kama, and have it add a header on incoming INVITEs and the like from anything with which it has REGISTERed indicating which outbound registration is relevant to said request. If I'm right that there isn't support for that, do any of the app modules expose enough sip capability easily to write such? -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

3 4

Configuration kamailio + siremis
by Tomasz Sidło 23 Jul '14

23 Jul '14

Hello all I'm looking for the person who configures my kamailio + siremis I'll pay for the full configuration. Kamailio is to work together with a asterisk if anyone willing please contact me mail tomasz(a)sidlo.pl

1 0

Kamailio RtpProxy MHomed
by Pascal Fautré 23 Jul '14

23 Jul '14

Hi, I tried to use Kamailio / RTPProxy in mhomed setup without any luck. I had no problem to configure it with only 1 interface, without mhomed, everything worked perfectly. The RTP streams where not established correctly even if I managed to have to proper IP in the SIP INVITE (C & O). Versions: version: kamailio 4.1.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 04:23:19 Jun 13 2014 with gcc 4.7.2 RTPProxy -v: Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Here is my RTPProxy config (/etc/default/rtpproxy) : CONTROL_SOCK=udp:127.0.0.1:7722 EXTRA_OPTS=“-l PU.BL.IC.IP/PRI.VA.TE.IP -m 11000 -M 12000 -d DBUG:LOG_LOCAL3 Here are snippets of my kamailio.cfg: port=5060 mhomed=1 # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; xlog("NATMANAGE M=$rm OU=$ou RURI=$ru RD=$rd F=$fu T=$tu NH=$nh(d) IP=$si ID=$ci\n"); if(dst_ip == PUBLIC_IP) { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coei\n"); rtpproxy_manage("coei", PRIVATE_IP); } else { xlog("NATMANAGE coee\n"); rtpproxy_manage("coee", PUBLIC_IP); } } else { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coii\n"); rtpproxy_manage("coii", PRIVATE_IP); } else { xlog("NATMANAGE coie\n"); rtpproxy_manage("coie", PUBLIC_IP); } } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } Calls were correctly going to the desired rtpproxy_manage options. Now I’m not quite sure I’m using the correct ones. I had to specify the PUBLIC_IP or PRIVATE_IP in the rtpproxy_manage calls in order to have the correct IP address in the C and O headers of the SIP INVITE. Without that, the public IP would be sent as C and O params to phones on the private subnet. In fact not a single call direction would give correct RTP streams. Any idea where I missed the turn? Cheers

4 10

core:drop_requests meaning
by Alex Balashov 23 Jul '14

23 Jul '14

Hi, I was wondering what precisely these stats mean: core:drop_replies = 0 core:drop_requests = 3 Assuming one does not explicitly drop() requests anywhere in script, would a continuous monotonic increase in drop_requests be considered a problem? Or are there some normal circumstances and ordinary behaviours (i.e. auth challenges) that would register a request drop? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 3

server for git repository is down
by Daniel-Constantin Mierla 22 Jul '14

22 Jul '14

Hello, a short note to say that the server for git repository on sip-router.org is down. The hosting provider hasn't reported any maintenance, I am digging further into it to see what happened. Meanwhile, the mirror on github can be used if anyone wants to fetch the source tree: - https://github.com/kamailio/kamailio Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

How install Kamailio with Asterisk
by ANTHONY HESNAUX 22 Jul '14

22 Jul '14

hello all, I would install kamailio with ASTERISK in realtime on CENTOS 6.5 x64. I followed the tutorial aspito: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but its not work. I have a problem with kamailio.cfg Its not possible to start kamailio When I loadmodule "uac.so" I want to make this configuration: SIP phone1 -> Register Kamailio -> ASTERISK Register SIP phone 2 -> Register Kamailio -> ASTERISK Register kamailio: 192.168.0.100 Port: 5060 ASTERISK: 192.168.0.100 Port 5080

3 6

Unknown caller gets online user's identity
by g.aloitus＠gmail.com 21 Jul '14

21 Jul '14

Hello, I'm using Kamailio version 4.1.4+precise (amd64). I have followed "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). One main difference in my setup compared to that one is that I continued use of Kamailio's database. The problem is as follows: I decided to put Kamailio and through it Asterisk reachable from internet. I have tried to configure Asterisk so that only calls of registered users would be possible, and they could only call to other registered users or conference rooms and echo test number. Then I took the following steps: I ensured that there was no online users with kamctl online. Then I launched MicroSIP (www.microsip.org), but I did not defined account, I simply set the protocol to tls and media encryption to mandatory, because I'm using these. I called to extension with xxx(a)my.public.ip.address (where xxx is extension) getting "unauthorized". And that was what I wanted. But if there is online users, calls go through, and incoming call is coming from Asterisk (in syslog I can find out that src_user=asterisk). Kamailio and Asterisk are listening the same IP address, but different port. I have refused connections to the Asterisk's port with iptables. I have defined my public IP address as domain in sip.conf. There is also other domain defined which corresponds to users' domain I am using in Kamailio's database. In kamailio.cfg there is if statement which prevents Kamailio not to be open relay: if (from_uri!=myself && uri!=myself) ... If I change this for example: if (from_uri!=myself || uri!=myself) I get what I want this time: no calls from outside, but I somewhat think that this is not a final solution. I have not found from log files such information which would have helped me. I have not yet investigated this problem so much that I could tell the logic behind the selection of online user's identity which is used. However, if I make a call to conference room I notice that Asterisk is thinking that one of online users has joined the conference. If I can recall correctly, I started with Kamailio version 3.2, and integrated it with Asterisk 11 (currently 11.10.2). Is there something which has changed in Kamailio, but what I have not changed in my setup which could explain this. Best, Teijo

6 18

kamcmd htable access problem, probably with int->string conversion
by Moritz Graf 21 Jul '14

21 Jul '14

Hi There, i implemented a mapping via htable. For debugging purpose I use kamcmd to see if the values are correctly filled. Doing so I discovered that there is a problem getting the values of the keys, where the name is a integer value. So it's the following behaviour: ##################################################################### root@kamailio-stage:~# kamcmd htable.dump maptodest { entry: 55 size: 1 slot: { item: { name: 1 value: 0000000000000000001 } } } { entry: 504 size: 1 slot: { item: { name: blub value: 0000000000000000001 } } } { entry: 686 size: 1 slot: { item: { name: 001 value: 0000000000000000001 } } } root@kamailio-stage:~# kamcmd htable.get maptodest blub { item: { name: blub value: 0000000000000000001 } } root@kamailio-stage:~# kamcmd htable.get maptodest 001 error: 500 - Not enough parameters (htable name and key name) ##################################################################### Is there a reasonable explanation why it works with blub, but not with 001? Why is he complaining about the number of parameters? Does anyone know a workarround? Thanks for any advise!! greeetz Moritz -- Moritz Graf, M.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg

2 3

pkg.stats
by Vitaliy Aleksandrov 21 Jul '14

21 Jul '14

Hi list, I have a problem with pkg.stats for tcp.main process on kamailio "4.1.2 73ea61-dirty". kamailio has 64M of private memory for each process (-M 64). pkg.stats for tcp.main process shows that is uses 75M and still has 62M free: { entry: 36 pid: 10032 rank: -4 used: 73109720 free: 65891192 real_used: 79146048 } root@webRTC:~# ps -p 10032 -o pid,tid,state,pcpu,pmem,rss,size,sz,vsz,args PID TID S %CPU RSS SIZE SZ VSZ COMMAND 10032 10032 S 0 26420 67508 76821 307284 /.../kamailio -f /.../kamailio.cfg -P /.../kamailio.pid -m 128 -M 64 -u kamailio -g kamailio This server handles 1000+ tcp/tls connections and works just fine for the last two months. That's why i think this is not a memory leak and just a pkg.stats problem. Does anybody else saw this or maybe somebody is already working to fix it ?

2 3

Msilo not delivering message (status 408)
by Peter Villeneuve 21 Jul '14

21 Jul '14

Hi, This is most likely a stupid mistake on my route config, but msilo correctly stores messages for offline users, but when they come back online, the message never arrives. Looking through the debug logs indeed there is an error, with status 408, which suggests kamaiio can't find a route to the user that just logged. Before I spend hours looking through all the routing blocks, perhaps someone more exprienced in kamailio can see the problem right away and offer a solution? Thanks DEBUG: msilo [msilo.c:1148]: m_dump(): msg [1-12] for: sip:1010@my.domain.com DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e3040, 9, -1, 0x10) fd_no=2 called DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f61ced1b1b0, state 1, fd=9, id=1 DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f61ced1c4f8 DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f61ced1b1b0, 1 from 0 DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89ec40, 26, 2, 0x7f61ced1b1b0), fd_no=21 DEBUG: <core> [tcp_main.c:3515]: handle_tcp_child(): handle_tcp_child: CONN_RELEASE 0x7f61ced1b1b0 refcnt= 1 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced1d868, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): *completed with status 408* [mid: 140058058335472/13] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <13> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:13 fl:8 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced5f098, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): completed with status 408 [mid: 140058058329184/12] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <12> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:12 fl:8

4 19

please help with Msilo config
by Miguel Rios 19 Jul '14

19 Jul '14

Hi list, I'm a newbie when it comes to kamailio, although I have a fair amount of SIP experience. I'm trying to setup a very basic kamailio install (4.1 on Wheezy) with Msilo support. I just used the default kamailio.cfg file (changing obviously the relevant parameters for my setup) and have not touched the routing blocks. I'm very confused about how routing works, and I don't have a background in C. I've read the wiki up and down and have looked at inumerous tutorials spread out on the internet (most of which just add to my confusion because they seem out of date), but I still don't get it. I managed to setup a working kamailio server where local users can call each other fine. Now when I try to add the msilo module, and specially the routing example from MSILO Module is when things get tricky. I'm sure my error is a fairly basic one and has to do with the routing logic and syntax. Could some kind soul please share a copy of the whole ####### Routing Logic ######## on downwards for a basic no frills kamailio with msilo setup? Thanks, Miguel MSILO Module 3. Parameters 3.1. db_url (string) Database URL. Default value is “mysql://kamailio:kamailiorw@localhost/kamailio”. Example 1.1. Set the “db_url” parameter View on kamailio.org Preview by Yahoo

3 3

Fwd: Msilo not delivering message (status 408)
by Peter Villeneuve 18 Jul '14

18 Jul '14

Thanks Daniel. UDP is the same story as TCP and TLS, unfortunately. Looking through the logs, I see strange errors in parsing the contact header when msilo prepares to send the notification back to 1000. This is for the notification message that should have been sent back to 1000 after he sent a message to 1010 (offline). I also see that kamailio reports that 1000's location was not found, even though he's obviously registered. The more I play around with this, the more confused I become. Do you think this contact parsing error could be the root of the problems? Cheers, Peter Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:1000@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK90dc.68116a97000000000000000000000000.0>; state=16 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:193]: receive_msg(): preparing to run routing scripts... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=10 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [28]; uri=[sip:1000@my.domain.com] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [< sip:1000@my.domain.com>#015#012] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=25 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-bb2e Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: *ERROR: <core> [parser/contact/contact.c:175]: skip_name(): skip_name(): Error in contact, scheme separator not found* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/contact.c:202]: parse_contacts(): parse_contacts(): Error while skipping name part Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: nathelper [nhelpr_funcs.c:300]: get_contact_uri(): failed to parse Contact body Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: !------------------------------------>WE HAVE A NEW MESSAGE Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: registrar [lookup.c:181]: *lookup(): contact for [1000] found by address* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==9 && [79.my.server.IP1] == [127.0.0.1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==14 && [79.my.server.IP1] == [79.my.server.IP1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: ------------------MESSAGE PROCESSING: *location not found*, storing offline message ID: (sip:1000@85.xx.xx.xx:59287;transport=TLS) Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:605]: m_store(): ------------ start ------------ Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:648]: m_store(): using user id [sip:1000@85.xx.xx.xx:59287;transport=TLS] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:732]: m_store(): 'From' header: <sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:849]: m_store(): message stored. T:<sip:1000@my.domain.com> F:< sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:858]: m_store(): sending info message. Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:899]: m_store(): contact header found On Thu, Jul 17, 2014 at 11:01 PM, Daniel-Constantin Mierla < miconda(a)gmail.com> wrote: > You should not call m_store() if src_ip==myself -- in this way you avoid > to store notification messages that cannot be delivered. > > Why notifications are not delivered has to be troubleshooted there. Have > you tried also with UDP? > > Cheers, > Daniel > >

1 0

[OT] LVUG, the London VoIP User Group
by Richard Brady 18 Jul '14

18 Jul '14

Hi folks LVUG is a monthly get-together in London to discuss open-source VoIP technology, including Kamailio. If you're based in or near London please join us this coming Tuesday for the third LVUG which will be held in Old Street. The format is a couple short talks followed by beers at the venue or in a nearby pub. Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/ Hope to see you there! Richard -- Richard Brady E: rnbrady(a)gmail.com

1 0

Kamailio presence (NAT, XCAP)
by Radu Vatav 18 Jul '14

18 Jul '14

Hello, I'm just trying to use SIP as a replacement for Skype, nothing more than that. All is working except of presence. Using Jitsi 2.4.4997 (Windows x64) from home behind NAT and kamailio 4.1.4+wheezy on a rented server somewhere. My biggest problem is that I don't know exactly how it is supposed to work. Ok, this is what I understood so far: the clients connect via SIP to the server and can call each other, but for some reason the server can't notify them via SIP of presence changes when they are behind NAT. So the clients also connect to the built in xcap-server, register themselves as presence "watchers" and the xcap-server notifies them via udp/tcp of presence changes of the other users which are watched. All is very nice but it's not working and I don't know where to look for the problem. Which are exactly the steps in communication and what data exactly is there sent which way (when working correctly, which it doesn't) - no clue. I somehow put the configuration together from the 2-3 examples I found on the internet, I don't know if it is ok to attach something to this email, so here it is: https://vatav.cc/kam/kam_cfg.7z Here is the network activity on the server: https://vatav.cc/kam/kam_ngrep.7z So me ("radu") and "alice" are home behind the NAT, one is online and wants to see how the other one goes online and offline. I don't know what exactly to look for, but it doesn't look to me like there is anything sent to one user when the other one goes on-/off-line (there is a lot of "proxy authentication required" which drives me nuts, again I don't know if this is correct or not; Jitsi seems to be connected to the XCAP server, if I type in a wrong address for "XCAP server" Jitsi complains it can't connect). Some other things: - "kamctl online" shows almost real-time who is logged-in, so this works - siremis shows a bunch of users in "watchers list", but none in "active watchers list" - actually when a user goes online it contains the contacts of that user, but only briefly, when I refresh the view, the list is empty again. I don't know if this is how it is supposed to work, anyway, Jitsi doesn't show any change of the status of the user who changed status at all. Any help would be appreciated. It's sad something this simple as I need (basic Skype replacement) is so difficult to get working. Maybe this is why there is relatively low interest in SIP and everybody stays with MS/Google/... because "there isn't something else". Best regards, Radu Vatav

2 1

db_postgres Error
by Rene Montilva 18 Jul '14

18 Jul '14

Hi list i'm config SQLOps Module with postgresql db module or is postgresql driver? this is my kamailio config: #!ifndef DBURL #!define DBURL "postgres://user:ouser@x.x.x.x/kamailio" #!endif sql_query("ca", "select * from xxxxx","ra"); if($dbr(ra=>rows)> 0){ xlog("L_INFO", "saldo:$dbr(ra=>[0,0]) y seg:$dbr(ra=>[0,1]) "); $var(credit) = $dbr(ra=>[0,0]); # 30$ of credit $var(cost_per_sec) = $dbr(ra=>[0,1]); # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second sql_result_free ("ra"); if($var(cost_per_sec) > $var(credit)) { sl_send_reply("402","payment required"); exit; } } i got this warning and would like why this happen: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [SSL SYSCALL error: Connection timed out#012] WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [SSL SYSCALL error: EOF detected#012]

2 1

how to m_store to database not just a "body" with MSILO?
by denis kremov 18 Jul '14

18 Jul '14

Hello All, Subj, need you help, How to store this type of message to MSILO database? Content-Type: message/external-body;access-type=URL;URL="https://xxx.com//download/539c28884_3e071c9db68044f687bf.jpg" I modified and recompiled msilo.c to resolve this error: ERROR: msilo [msilo.c:626]: m_store(): body of the message is empty! but after that - m_store just store this information to database: mysql> select * from silo; +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | id | src_addr | dst_addr | username | domain | inc_time | exp_time | snd_time | ctype | body | extra_hdrs | callid | status | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | 30 | sip:777351300@xxx.in | sip:777581288@xxx.in | 777581288 | xxx.in | 1402479947 | 1403479946 | 0 | message/external-body | | | | 0 | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ 1 row in set (0.00 sec) How can I store to database "access-type=URL" and and URL="xxxx" ? Thanks All! -- denis kremov

3 2

Groups in Kamailio vs Freeswitch
by Peter Villeneuve 18 Jul '14

18 Jul '14

Hi guys, I'm designing a new service for a client and I was wondering what your opinion is of the 2 options I'm considering to separate users on the server. Basically I want to create closed groups where each user can only call and receive calls from members of the same group (single domain install). Initially I was thinking of just forwarding all the invites to FS and using the dialplan to enforce this segregation, but now I realize there is a groups module in kamailio too. Which 2 courses of action do people recommend? I'm more familiar with FS than Kamailio, so it would be easier for me to just forward everything to FS, although that will be more expensive and probably unnecessary if I could do it all in Kamailio with Siremis. Would kamailio's group module do the trick? Are there any tutorials out there I can follow? Cheers, Peter

2 1

trying to conifgure telepresence client software
by Fred Hunter 18 Jul '14

18 Jul '14

Hi, I recently registered with IP tel for the purposes of carrying out SiP based teleconferencing. I am attempting to register my software client, Polycom m100 desktop but am having some difficulty. Can you tell me what I should be entering into the 'registrar server' field, 'domain' and 'registrar server'? [cid:image001.png@01CFA03E.F01888F0] ---------------------------------------------------------------------------- www.puntersouthallgroup.com Punter Southall Group is a trading name of Punter Southall Group Limited. Registered office: 11 Strand, London, WC2N 5HR. Registered in England and Wales No. 04096788. This message and any attachments are confidential. If you have received this message in error please notify the sender and delete this email immediately. Unauthorised dissemination, disclosure, copying or use is prohibited. This email has been scanned for viruses and other malicious items. We reserve the right to monitor the content of all emails sent and received by our systems. Please consider the environment before printing this email. ----------------------------------------------------------------------------

2 1

Re: [SR-Users] how to remove non-digits using dp_translate
by Daniel-Constantin Mierla 18 Jul '14

18 Jul '14

Hmm, I see there is none inside dialplan for the moment. The option would be to do it in config file with a function from textops. Also, check if regex module is a good alternative for you: - http://kamailio.org/docs/modules/stable/modules/regex.html Cheers, Daniel On 16/07/14 16:50, Henry Fernandes wrote: > Can anyone show me how to use the regexp global flag using dp_translate? > -H > > From: Henry Fernandes <henry(a)usinternet.com <mailto:henry@usinternet.com>> > Reply-To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Date: Friday, 11 July, 2014 4:18 PM > To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > I don't know how to apply the global flag in Kamailio. Can you give > me an example? > -H > > From: Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > Reply-To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio > (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Date: Thursday, 3 July, 2014 1:47 AM > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > Hello, > > you have to replace [^0-9] with empty, using the global flag. > > Cheers, > Daniel > > On 02/07/14 23:11, Henry Fernandes wrote: >> I'd like to use the dialplan module to remove all non-digits from a >> string. For example, "1-306-555-1234" becomes "13065551234" and >> "(306) 555-1234" becomes "3065551234". >> >> Unfortunately, I can't figure out the regular expressions for >> subst_exp and repl_exp. I can easily replace one instance of a >> non-digit, however I can't figure out how to replace all non-digits. >> >> -H >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.comhttp://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda > _______________________________________________ SIP Express Router > (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ SIP Express Router > (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Some of the calls drops after 15 minutes + some seconds
by Andras FOGARASI 18 Jul '14

18 Jul '14

Hi, I have a simple kamailio install (2 servers, using location service and a failover node with dispatcher, STUN, clients behind different NATs), without rtpproxy, only peer-to-peer RTP and TURN server if the connection is really messy (it's not relevant here). Signaling is over TLS. Both of the clients are behind NAT. Basically everything works, but some of the calls are dropped after 15 minutes and some seconds, for me it seems the RTP connection is dropped but not sure. I cannot find find out wether it's a kamailio problem or client/NAT problem. Has anyone idea what is going on? Thanks, Andras

3 8

Really need help on kamailio-snmpstats module
by Allen Zhang 17 Jul '14

17 Jul '14

Hi This is a re-post. I'm desperate. What I'm trying to do: Use rpmbuild to build kamailio rpms from source code. We made a lot of changes in the kamailio code. I can't use the rpms from the build service. The plan is to push these changes to kamailio but that's not done, yet. The problem is when I try to install kamailio-snmpstats module from the rpm built, it reports error: Error: Package: kamailio-snmpstats-4.0.0-0.el6.x86_64 (company_name) Requires: net-snmp-agent-libs I've installed net-snmp on the build machine and the machine that needs to install kamailio-snmpstats. I have tried to use the kamailio.spec file from kamailio build server but it didn't solve the problem. Can anyone help, please? Regards, Allen

2 6

Condition Statements - How many levels
by Joel White 16 Jul '14

16 Jul '14

I was wondering how many levels deep in conditional _if_ _else_ statements can go. I need to create a statement that has 4 possible outcomes and need to know if this is possible. I have two variables I want one thing to happen if only the first variable is present Another if only the second variable is present Yet another if they are both present And a fail over if None of the variables are present I do see that is can be done with 3 conditions using _if_ _if else_ _else_ how can I accomplish 4 conditional statements? Thank you in advance Joel

2 2

Re: [SR-Users] how to remove non-digits using dp_translate
by Henry Fernandes 16 Jul '14

16 Jul '14

Can anyone show me how to use the regexp global flag using dp_translate? -H From: Henry Fernandes <henry(a)usinternet.com> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Friday, 11 July, 2014 4:18 PM To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate I don't know how to apply the global flag in Kamailio. Can you give me an example? -H From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Thursday, 3 July, 2014 1:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate Hello, you have to replace [^0-9] with empty, using the global flag. Cheers, Daniel On 02/07/14 23:11, Henry Fernandes wrote: > > I'd like to use the dialplan module to remove all non-digits from a string. > For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" > becomes "3065551234". > > > > > Unfortunately, I can't figure out the regular expressions for subst_exp and > repl_exp. I can easily replace one instance of a non-digit, however I can't > figure out how to replace all non-digits. > > > > > -H > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listi > nfo/sr-users > -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/lis tinfo/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

rtpengine upgrade fails
by Juha Heinanen 16 Jul '14

16 Jul '14

i build debian packages for rtpengine branch 3.3.0.0+0~mr3.4.1.0. after that i did 'apt-get dist-upgrade' and it failed like this: Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Installing new version of config file /etc/init.d/ngcp-rtpengine-daemon ... Restarting RTP/media proxy: ngcp-rtpengine-daemonFATAL: Module xt_MEDIAPROXY not found. iptables: No chain/target/match by that name. ip6tables: No chain/target/match by that name. FAILED TO CREATE KERNEL TABLE 0, KERNEL FORWARDING DISABLED invoke-rc.d: initscript ngcp-rtpengine-daemon, action "restart" failed. dpkg: error processing ngcp-rtpengine-daemon (--configure): subprocess installed post-installation script returned error exit status 255 however, if i now run 'apt-get install ngcp-rtpengine-daemon', setting up ngcp-rtpengine-daemon succeeds and also packages that depended on it get installed fine: Reading package lists... Done Building dependency tree Reading state information... Done ngcp-rtpengine-daemon is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. 2 not fully installed or removed. After this operation, 0 B of additional disk space will be used. Do you want to continue [Y/n]? Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Restarting RTP/media proxy: ngcp-rtpengine-daemon. Starting RTP/media proxy: ngcp-rtpengine-daemon already running. Setting up ngcp-rtpengine (3.3.0.0+0~mr3.4.1.0) ... any ideas why setting up ngcp-rtpengine-daemon fails with dist-upgrade? -- juha

2 1

KAMILIO-4.1.4 REGISTRATION ISSUE
by Cibin Paul 16 Jul '14

16 Jul '14

I have kamailio 4.1.4 installed with DB Engine as mysql. I added users using the command kamctl add and I could see the entry in subscriber table. When I am trying to register, kamailio is accepting any combination of username and password not listed in the table. What could be wrong? I followed the INSTALLATION document provided in the source file. Please help me troubleshoot this. Kamailio is up and running and not throwing any errors. kamctl moni shows 1 user registered. But the registered user doesn’t even exist Server Details OS : Centos 5.5 Kamailio version - 4.1.4 MySQL version - 5.0.95 Cibin

2 4

Rtpproxy redundancy
by Dren Daka 16 Jul '14

16 Jul '14

Hi, I'm using kamailio with two network interfaces also running rtpproxy on the same interfaces so rtpproxy is load balancing, when one of the connections is down rtprpoxy still sends 50% of the calls in that interface. How can i make rtprpoxy to do failover from the non working connection to the working one instead of load balancing. Thanks.

1 0

BUG: ip_addr2sbuf: unknown address family 0
by AliReza Khoshgoftar Monfared 16 Jul '14

16 Jul '14

Hi, I am running a simple command on timer in a kamailio proxy. Here is the related parts of the config scripts: loadmodule "timer" > modparam("timer", "declare_timer", "tmr1=LOG_STATS,1,fast,enable"); > > route{ > ... > } > > route[LOG_STATS] { > exec_avp("netstat -ul | grep ':sip' | awk '{print > $$2}'","$avp(s:test)"); > xlog("L_ALERT","logged RecvQ as $avp(s:test)"); > return; > } > Each time I run the "netstat" using the exec() inside the timer route block, I get the following error from Kamailio in my log: kamailio[28546]: : exec [../../parser/../ip_addr.h:669]: ip_addr2sbuf(): > BUG: ip_addr2sbuf: unknown address family 0 > What is the meaning of this error, and what does kamailio mean by reporting it as a "BUG"? Is there any obvious reason, I am getting this? Is it related to calling netstat to survey the RecvQ? Thanks, Alireza

2 1

have problem with iptel account
by Samhu 16 Jul '14

16 Jul '14

hi, This is Sam from Draytek Crop. We have a IPPBX device with iptel for SIP account as below. User B dial to IPPBX could heart the answer from Auto Attendant of IPPBX. Then user B cannot heart anything (after input the extension of user A) when user A pickup his telephone . At this momemt, user A could hear everything from user B. This Network Architecture worked fine before. And it still works fine now if we select the account of "draytel.org". Please show us how to solve the problem. Best Regards, Sam Hu

2 1

SCA and PRESENCE
by Joel White 15 Jul '14

15 Jul '14

I am having an issue where I enable presence modules and the SCA module stops working. I disable presence and SCA works fine on Polycom phones. Does anyone know what I am missing or doing wrong? Thank you in advance Joel

3 3

function to reset all flags or bflags?
by Juha Heinanen 15 Jul '14

15 Jul '14

there is a function to reset a flag or bflag, but i haven't found one that would reset all flags or bflags. is there a way to do it? if not, is it ok to add such function? -- juha

2 2

Many instances of "new URI shorter than old URI" in log
by Alex Villacís Lasso 15 Jul '14

15 Jul '14

Jun 27 15:28:42 elx /usr/sbin/kamailio[25189]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:44 elx /usr/sbin/kamailio[25187]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:48 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:49 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI I am not really sure of what this message is supposed to indicate. It looks something related to uac_restore_[from,to] functions, but some debugging statements just before the use show that the message is not originating from there. This message is showing up hundreds of times per day.

2 4

set_rtp_proxy_set behaviour
by Igor Potjevlesch 15 Jul '14

15 Jul '14

Hello, I set two entries of rtpproxy like : modparam("rtpproxy", "rtpproxy_sock", "1 == udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "2 == udp:127.0.0.1:7723") For a particular case, I want to use the second RTPProxy and the first one as the default one. If in a conditional block I set set_rtp_proxy("2"), all other call to rtpproxy_manage will use the first RTPProxy? Or there is a load-balancing introduce by default. Regards, Igor.

2 3

Routing calls to Asterisk using dispatch
by Olli Heiskanen 15 Jul '14

15 Jul '14

Hello, I've started playing with an idea to add multiple asterisk servers and using dispatcher to balance the sip load between them. I added the code according to dispatcher module documentation ( http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I think there's something off in my setup: kamctl ul output shows 2 AORs for one client: AOR:: 770(a)testers.com Contact:: sip:770@2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP Q= Expires:: 3221 Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35ae-2a2 Reg-Id:: 0 Last-Keepalive:: 1405174150 Last-Modified:: 1405174150 AOR:: 770(a)1.1.1.1 Contact:: sip:770@1.1.1.1:5070 Q= Expires:: 68 Callid:: 327fcf07641f80006e962821112a61b5(a)testers.com Cseq:: 754 User-agent:: Asterisk PBX 11.10.2 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35af-a82 Reg-Id:: 0 Last-Keepalive:: 1405174477 Last-Modified:: 1405174477 I don't think I should be seeing an AOR for 770 where Contact is the public address of my server (here 1.1.1.1) and User-Agent which is Asterisk. I'm using Asterisk Realtime integration, and by what I can tell the sip messages are going nicely, client authenticates with Kamailio and sends this message to Asterisk (which is on the same machine; Kamailio at 5060 and Asterisk at 5070): 1.1.1.1.sip > 1.1.1.1.vtsas: SIP, length: 374 REGISTER sip:1.1.1.1:5070 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKbc8a.f4473947000000000000000000000000.0 To: <sip:770@1.1.1.1> From: <sip:770@1.1.1.1>;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1 CSeq: 10 REGISTER Call-ID: 7ffa0191-13742(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 Contact: <sip:770@1.1.1.1:5060> Expires: 3600 Currently I can make calls from 770 to 123 which is an Asterisk extension that answers, plays hello world and hangs up. However I can't call another sip clients when I route calls through Asterisk, they do work fine if I don't use Asterisk for handling calls, but I'd like Kamailio to be in the role of proxy/loadbalancer and Asterisk to handle calls. My config is the simple default config, added with realtime stuff and then dispatcher according to the documentation. I wonder if there's something wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something else going wrong? Has anyone seen results like this and do you spot something here that needs fixing? Thanks, Olli

1 1

siremis/ser/xrcmds error help
by MARY 15 Jul '14

15 Jul '14

sr-users，hello：　 http://127.0.0.1:8890/siremis/ser/xrcmds error info: [2014-07-14 08:49:30 (GMT)] An exception occurred while executing this script: Error message: #2, require_once(XML/RPC.php): failed to open stream: No such file or directory Script name and line number of error: /xxxx/siremis-4.1.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php:2 why? can not find the XML/RPC.php file。 2014-07-1417:56:16 MARY

2 1

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