- sr-users - kamailio.org

SIP contact matching - location db ops
by Daniel-Constantin Mierla 30 Mar '13

30 Mar '13

Hello, apparently all the db operations to update/delete contacts are done by matching on username, contact, call-id and domain (when use_domain=1) -- it is like this for way many years, therefore needs some debate if/how to change. While the module has a matching_mode parameter for various contact matching algorithms, it is used for in-memory matching only. - http://kamailio.org/docs/modules/stable/modules/usrloc.html#matching-mode IIRC, for initial SIP specs (with no GRUU extensions, etc), aor contact address should be used for matching. Call-Id has a relevance only when CSeq is lower than previous received REGISTER with same call-id. Do I miss any specs requirements? Following the report on tracker FS#278, matching on path value for db ops will be probably very inefficient. Matching on call id for this case should be sufficient if the REGISTER comes from different UA. Can it happen that same UA sends same REGISTER via two paths? With this question I mean if anyone is aware of an UA able to do this. Relying on previous hop address to use in the db key, would not be enough, as the different path can be in the hops before. I plan to keep for now call-id for db operations when matching mode involves path, one way to fix it properly could be usage of ruid as db key, but needs further analysis. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 3

Help with SIP over Websocket audio call
by Brad Johns 30 Mar '13

30 Mar '13

Hi, New to Kamailio. I have my Kamailio 4.0 server with websocket support, and the users can register using the JsSIP Tryit sample WebRTC application. However, after registration, the users can't place an audio call. I see no ringing on the remote browser. I don't know how to debug this further to find out what the problem is. Can anyone help with clues or debug? In Debug log I can see the websocket ws_frame.c decode the websocket message into SIP, and I see normal SIP call flow for an INVITE. However, nothing indicating a call. With this JsSIP, I can do chat through Kamailio SIP over WebSockets. With this Kamailio server, SIP User Agent Clients work just fine to register and place SIP call with audio. It's just that WebRTC audio calls don't work with JsSIP sample application with Kamailio 4.0 websocket module. Kamailio websocket configuration borrowed from: https://gist.github.com/jesusprubio/4066845 Any help debugging this appreciated. Brad

2 7

Help with SIP over Websocket audio call: 405 Method Not Allowed
by Brad Johns 30 Mar '13

30 Mar '13

Hi, New to Kamailio. I have my Kamailio 4.0 server with websocket support, and the users can register using the JsSIP Tryit sample WebRTC application. They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over websockets module. However, after registration, the users can't place an audio call. I see no ringing on the remote browser. Can anyone help with clues or debug? In Debug log I can see the websocket ws_frame.c decode the websocket message into SIP, and I see normal SIP call flow for an INVITE. However, nothing indicating a call. I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm getting an error response to browser UA of "405: Method Not Allowed". I've isolated it down to the this snippet in the kamailio.cfg for route[LOCATION]: $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", "TEST: Method Not Allowed"); exit; } } The switch case is returning -2, for some reason. Any help in debugging this appreciated.

3 5

sems.conf
by Maaz Bin Mahmood 30 Mar '13

30 Mar '13

Hi there, I am trying to try SEMS but after make install, file sems.conf is not available in /usr/local/etc/sems/ Can you please suggest a solution? as tutorial here ( http://ftp.iptel.org/pub/sems/doc/current/howtostart_noproxy.html ) asks to change sems.conf as a first step. Regards, Max

2 1

carrierroute module
by Angel Angelov 29 Mar '13

29 Mar '13

Hello, When I tried to start kamailio-4.0 with carrirroute module include, the follow error appeared 0(8732) ERROR: [sr_module.c:572]: ERROR: load_module: could not open module : /usr/local/kamailio-4.0/lib/kamailio/modules/carrierroute.so: undefined symbol: cfg_getnint . Do you need more information? Thanks in advanced. Regards. Angelov?

2 1

IPv4 / v6 request to global parameter "TOS"
by Klaus Feichtinger 29 Mar '13

29 Mar '13

Hello list, in a bridging scenario with kamailio 3.3.4 and rtpproxy 1.2.1 for bridging signalling and media from an IPv4 to an IPv6 network and vice versa I found that the TOS value, which is set in kamailio.cfg, is used for IPv4 packets only. IPv6 packets have the traffic class value set to the default value 0x0. In other words: kamailio doesn´t use this variable for IPv6 packets. In the cookbook (http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tos) I haven´t found any hint that it _is_ limited to IPv4 only ("...for the sent IP packages"). I know that the name "tos" may be misleading, as the original definition was outdated by dscp+ecn, but it was/is working fine now. However, as IPv6 is using dscp+ecn, too, I wonder if the tos variable should support IPv6 packets, too. Could anybody give me a hint? Is there maybe an alternative way to prioritise SIP in IPv6 with kamailio? Thanks in advance, Klaus Feichtinger

2 3

wiki accounts cleanup
by Daniel-Constantin Mierla 29 Mar '13

29 Mar '13

Hello, as there were many spamming accounts created in the wiki, used to add inappropriate content (http://www.kamailio.org/wiki/start?do=recent), I went through them and deleted those having more like random name and random id for email addresses. Hopefully I haven't deleted any valid account, but if you try yours and does not work, try to create it again. If you still have troubles, write me mentioning the account id. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] Iimitations of xcap server in kamailio
by David Thomson 29 Mar '13

29 Mar '13

Thanks Peter! I'll try that out!! Is there a doc that goes through how to tweak Kamailio into production mode? What about having two Kamailio servers sharing one db and allowing calls between the two servers to be completed? That's another hurdle I've yet to get over. Ttyl, Dave ------Original Message------ From: Peter Dunkley To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Iimitations of xcap server in kamailio Sent: Mar 28, 2013 10:20 PM There are a number of core parameter and modparam values you need to increase from their defaults in order to be able to handle large HTTP requests (and therefore large XCAP documents). For example, * tcp_rd_buf_size (core parameter) * tcp_wq_max (core parameter) * sql_buffer_size (core parameter) * buf_size (xcap_server modparam) Regards, Peter > Has anyone else noticed issues when adding a large number of contacts to > your buddy list? I was using Jitsi and at about 40 contacts I was > encountering issues in that the subsequent contacts would not get added to > my buddy list. > > Ttyl, > Dave > > > > _______________________________________________ > 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 > -- Peter Dunkley Technical Director Crocodile RCS Ltd _______________________________________________ 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

500 ERROR Reloading data when reloading hash table using FIFO
by kawarod 29 Mar '13

29 Mar '13

Dear list users, I encounter an issue when reloading a hash table using FIFO command. My hash is getting a list of "Username / Serial Number" from SQL for each phone connecting to Kamailio. Then I use this hash to authenticate the phone based on a X-SerialNumber attribute. This hash has to be updated once or twice a day, to accomodate new users or a change of the Phone. I'm using the following command for reloading: /opt/kamailio/sbin/kamctl fifo sht_reload hash_cpe_serial Everything is working fine, but after 25-26 reload I have an error like this: 500 ERROR Reloading data The only way to recover is do a restart with ./kamctl restart The hash size has approximatively 22 000 users. The hash has been defined like this in the configuration file: modparam("htable", "htable", "hash_cpe_serial=>size=14;dbtable=subscriber;") When using kamcmd, i have this: kamcmd> htable.listTables { name: ipban dbtable: dbmode: 0 expire: 300 updateexpire: 1 size: 256 } { name: hash_cpe_serial dbtable: subscriber dbmode: 0 expire: 0 updateexpire: 1 size: 16384 } When trying to dump with kamcmd, I have an error: kamcmd> htable.dump hash_cpe_serial error: 500 - Internal server error processing '{': buffer too small (overflow) (-2) But it works fine with: /opt/kamailio/sbin/kamctl fifo sht_dump hash_cpe_serial I have upgraded from 3.2.2 to 4.0.0 to be sure that it was not relative to the old version, but without success. Any idea on how to correct this. Kind regards, rod

1 1

kamailio 4.0.0; km_append_branch problem
by Pavel Miskov 29 Mar '13

29 Mar '13

Hi all, I installed Kamailio 4.0.0 from source an tried configuration from Kamailio 3.3.2. problem is with km_append_branch function. Not starting Kamailio400: invalid configuration file! -e 0(20346) ERROR: <core> [cfg.y:3431]: cfg. parser: failed to find command km_append_branch 0(20346) : <core> [cfg.y:3570]: parse error in config file /usr/local/kamailio400/etc/kamailio/kamailio.cfg, line 2448, column 22: unknown command, missing loadmodule? ERROR: bad config file (1 errors) configuration: ------------------ mpath="/usr/local/kamailio400/lib64/kamailio/modules_k/:/usr/local/kamailio400/lib64/kamailio/modules/" loadmodule "kex.so" ... km_append_branch(); route(RELAY); exit; } ... Please advise Best regards Pavel

2 2

Iimitations of xcap server in kamailio
by David Thomson 28 Mar '13

28 Mar '13

Has anyone else noticed issues when adding a large number of contacts to your buddy list? I was using Jitsi and at about 40 contacts I was encountering issues in that the subsequent contacts would not get added to my buddy list. Ttyl, Dave

2 1

Homer / Kamailio / Freeswitch
by Antonio Teixeira 28 Mar '13

28 Mar '13

Hello. First of all i just did a fresh install of Homer and this thing is perfect only really missing the RTP analysis. Anyway I have a Freeswitch PBX with several internal profiles and also external profiles binded into external Ips ( Public not Private). Capture on the private network profiles works flawlessly when i try to track public interface , calls going into the VOIP Operators , there are no entries .... So : Public call data inside the DB , No Freeswitch sending data into Kamailio , Yes TCPDUMP on Kamailio Shows SIP Data from the PBX Interface , Yes , Kamailio complains with Mar 28 08:30:27 debian /usr/local/sbin/kamailio[23709]: ERROR: <core> [parser/msg_parser.c:705]: ERROR: parse_msg: message=<#015#012> Mar 28 08:30:27 debian /usr/local/sbin/kamailio[23709]: ERROR: <core> [receive.c:146]: ERROR: receive_msg: parse_msg failed Problem Scenario : A call from a phone inside the network 1.1.1.0 Into the PBX 1.1.1.1 is tracked by Homer a call from the external IP Binded into the PBX lets say something like 80.0.80.8(External Public IP) , Call not tracked and Kamailio complains with parse_msg_error. Helper Data : Uname Linux debian 2.6.32-5-amd64 #1 SMP Mon Feb 25 00:26:11 UTC 2013 x86_64 GNU/Linux LSB Distributor ID: Debian Description: Debian GNU/Linux 6.0.7 (squeeze) Release: 6.0.7 Codename: squeeze Kamailio as of TODAY from GIT Homer as of TODAY from GIT WebHomer 3.3.1 From GIT Also Using the SOFIA SIPTRACE , problem shows both using the global sip trace or individual profile siptrace. Thanks for your time Antonio

2 1

base64 transformation
by Andreas Granig 28 Mar '13

28 Mar '13

Hi guys Maybe I missed something, but looks like there is no base64 encode/decode transformation in PV, right? I think we're going to add one, as it might be handy to do such transformations directly from within the config file. One case where this could be used is that some UAs are picky in regards to URIs as URI params, e.g. P-Foo: <sip:alice@example.org;x='sip:1.2.3.4:1234'> which could be changed into P-Foo: <sip:alice@example.org;x=c2lwOjEuMi4zLjQ6MTIzNAo=> which might be understood more easily (although it's still just a work-around for broken UAs, but that's how life is). Any objections against such a transformation? Andreas

3 2

Tag maximum size/length
by Grant Bagdasarian 28 Mar '13

28 Mar '13

Hello, Is there a maximum length/size defined for the From and To Tag? The RFC states: "it MUST be globally unique and cryptographically random with at least 32 bits of randomness". Though I can't find anything in the RFC about the maximum. The sip_capture table stores the to_tag and from_tag in a varchar(64). Does this mean the max is 64? Source: https://code.google.com/p/homer/source/browse/sql/create_sipcapture.sql Thanks, G

2 3

Need help with Kamailio+RLS configuration.
by Aleksandrs Semenenko 27 Mar '13

27 Mar '13

Hello everyone, I am running Kamailio 3.3.2 server and I need to configure it to support RLS to avoid big amounts of SUBSCRIBE requests from clients. By this moment I have managed to configure XCAP server and learned the XML structure for press-rules, rls-services and resource-list documents. Also I have successfully put rls_update_subs and rls_handle_subscribe calls to kamailio.cfg, so now when I PUT new rls-services or resource-list document to XCAP I can see these calls triggered. The problem is that on client side I always get empty notification XML: <?xml version="1.0"?> <list uri="sip:user1-list@sip.server" xmlns="urn:ietf:params:xml:ns:rlmi" version="1" fullState="true"> <resource uri="sip:user2@sip.server"/> <resource uri="sip:user3@sip.server"/> </list> As I understand, user2 and user3 must have added user1 to pres-rules document? Correct? So it is… I assume there should be a call to rls_handle_notify function somewhere in kamailio.cfg, but where exactly? I have tried to add it after SUBSCRIBE condition, but it is never called in fact: if( is_method("SUBSCRIBE")) { xlog("BEFORE rls_handle_subscribe"); $var(ret_code)= rls_handle_subscribe(); xlog("AFTER rls_handle_subscribe"); if($var(ret_code)== 10) handle_subscribe(); t_release(); } else if (is_method("NOTIFY")) { xlog("BEFORE rls_handle_notify"); rls_handle_notify(); xlog("AFTER rls_handle_notify"); } Any comment would be appreciated! Regards, Alex.

1 1

SIP hard phones with TLS client certificate
by Daniel-Constantin Mierla 27 Mar '13

27 Mar '13

Hello, I am trying to build a list of SIP hardphones that have support for TLS with client side certificate (i.e., import custom ssl certificate in the phone specific for each SIP account). Anyone that can report back the vendor and model (eventually the minimum/current firmware version)? Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Implementing encryption
by Techie Sup 27 Mar '13

27 Mar '13

Hello, Any successful implementation of adding some customized security (encryption) over sip and rtp in kamailio? (non TLS) Also looking for assistance on the concept of rtp over tcp using kamailio and rtpproxy. Warm Regards Selvam -- Sent from my mobile device

4 10

Re: [SR-Users] How to enable TLS in Kamailio ?
by David Thomson 27 Mar '13

27 Mar '13

Didn't read the whole thread, but, in some clients you need to specify the transport is TLS and the port is 5061 (assuming you use the default). Hth. Ttyl, Dave -----Original Message----- From: Khoa Pham <onmyway133(a)gmail.com> Date: Thu, 21 Mar 2013 03:52:12 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] How to enable TLS in Kamailio ? @Moacir, thanks 1. You said that " then use "listen="my ip address" and all enabled services will be bound to the especified IP address." So which port does Kamailio listen for TCP and TLS? How does client know which port to connect to Kamailio ? On Thu, Mar 21, 2013 at 12:41 AM, Moacir Ferreira <moacirferreira(a)hotmail.com <mailto:moacirferreira@hotmail.com> > wrote: Hummm. When I start playing with Kamailio I had some problems that were related to the compilation process not really to the final product. Now, if you comment out the listen statement then all server interfaces will listen for all enabled services (SIP-UDP, SIP-TCP and SIP-TLS if you enabled it). AS a troubleshooting suggestion, just comment out the "listen" statement and all enabled services will bind to all available IP interfaces. If it works, then use "listen="my ip address" and all enabled services will be bound to the especified IP address. Best regards, Moacir ---------------- From: oej(a)edvina.net <mailto:oej@edvina.net> Date: Wed, 20 Mar 2013 12:43:06 +0100 To: sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> Subject: Re: [SR-Users] How to enable TLS in Kamailio ? 20 mar 2013 kl. 10:33 skrev Khoa Pham <onmyway133(a)gmail.com <mailto:onmyway133@gmail.com> >: Hi Olle, I follow these 2 tutorials 1. http://nil.uniza.sk/network-security/tls/configuring-tls-support-kamailio-3… which only describes to listen on tls listen=tls:158.193.139.51:5061 <http://158.193.139.51:5061/> 2. http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates which only describes to listen on udp and tcp listen=udp:<ip-address-for-receiving-sip-requests>:5060 listen=tcp:<ip-address-for-receiving-sip-requests>:5060 It is not until I listen on both TCP and LTS does it work. You document http://kamailio.org/docs/modules/4.0.x/modules/tls.html seems to lack of these "listen" Right. But it's in the core cookbook. We should propably add the listen to the TLS docs too. Thanks! /O On Wed, Mar 20, 2013 at 4:10 PM, Olle E. Johansson <oej(a)edvina.net <mailto:oej@edvina.net> > wrote: 20 mar 2013 kl. 07:55 skrev Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com> >: Hello, when tls module is installed, a self signed pair of certificate-private key is generated in /usr/local/etc/kamailio If you need one that is signed by a trusted CA (e.g., Verisign), you will have to buy it. Cheers, Daniel On 3/19/13 8:08 AM, Khoa Pham wrote: Hi, I want to enable TLS in Kamailio, as in here http://kamailio.org/docs/modules/stable/modules/tls.html But how can I get the certificate and private key ? The documentation for the TLS module actually includes a quick howto. http://kamailio.org/docs/modules/4.0.x/modules/tls.html What part of this needs clarification? Please help us make the documentation better if there are parts you do not undertand or isn't explained. Thanks, /O _______________________________________________ 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 -- Khoa Pham HCMC University of Science Faculty of Information Technology _______________________________________________ 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 _______________________________________________ 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 -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 3

kamailio > user agent client...
by Douglas Adami 27 Mar '13

27 Mar '13

hello guys, I'm new here, already working with Asterisk for some time and decided to learn more about kamailio, I'm studying your settings and features and want to start a project to integrate with Asterisk, but leaving the asterisk for voicemail and IVR. http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb I have a question, can I use kamailio as client for the various gateways and voip providers, saw that possibility in module UAC, but how do I authenticate to multiple servers? anyone have any sample settings? Hugs to all! Doug

2 1

SIP over WebSockets error: bad request
by Brad Johns 26 Mar '13

26 Mar '13

Hi, Wondering if anyone could please help me debug what is wrong with my kamailio.cfg for Websockets support. I'm using JsSIP sample application, and I've compiled Kamailio 4.0 for WS support. My config is borrowed from link this link [1]. Wondering if anyone can help me debug this. Error is bad request, from syslog: Mar 26 19:26:42 ace /usr/local/sbin/kamailio[25774]: ERROR: <core> [tcp_read.c:1296]: ERROR: tcp_read_req: bad request, state=7, error=4 buf:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:8888#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012#012parsed:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:8888#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012 Any ideas? Can send my kamailio.cfg if necessary. [1] Kamailio sample config for WebSockets https://gist.github.com/jesusprubio/4066845

2 3

Dialog module - shared state
by Charles Chance 26 Mar '13

26 Mar '13

Hi all, I was wondering, how are others sharing dialog module (particularly profile) state between multiple servers? 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: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

2 5

Kamailio World Updates - March 26
by Daniel-Constantin Mierla 26 Mar '13

26 Mar '13

Hello, the agenda is pretty much set for Kamailio World Conference, see it at: - http://conference.kamailio.com/k01/schedule/ The event starts in three weeks from now, we are glad to be able to host a great group of speakers. There are well known names in the industry able to share insights of the telecom and RTC/RCS market as well as talks about new projects that have potential to grow and help building new services. Check the speakers list at: - http://conference.kamailio.com/k01/speakers/ Regarding Kamailio project, two weeks after release of v4.0.0, the development branch is very active, with two new modules already imported there (app_java - embedded execution of java code; cnxcc - call control/prepaid system module), other new modules in developers personal branches and new features to old modules (e.g., snmpstats, usrloc). If you haven't registered yet for the Kamailio World conference, be aware of registration timeline updates - standard registration period has been shortened, last date for it being April 5, 2013. As the number of participants exceeded initial estimation, the company taking care of offering drinks and meals needs to know attendees' details more in advance. Registration form and more details at: - http://conference.kamailio.com/k01/registration/ Take in cosideration also the companies coming with demos, presenting products or services: Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. You may want to reserve some time for private discussions with them, contact us for further details. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio on AWS
by Felipe Ramos 26 Mar '13

26 Mar '13

Can anyone please point me on a configuration example for kamailio when it is behind NAT, i need to change the contact ip from internal to external on sdp and proxy all media from outside to internal asterisk farm. Please any help will be appreciated. most of documentation available is when client is behind Nat but not when Server behind Nat like is the case with Amazon aws vpc instances. Again thank you before hand.

2 3

Debugging and syslog
by Daniel W. Graham 26 Mar '13

26 Mar '13

Hello, Reaching out for some assistance with debugging, I am not seeing any xlog messages in my log file. If there is any other info I didn't include that would be helpful please let me know. Any ideas are appreciated. Using Kamailio 3.3.2 & Cent OS 6 Rsyslog.conf -------------------- *.info;mail.none;authpriv.none;cron.none;local0.none /var/log/messages # Kamailio local0.* -/var/log/kamailio Kamailio CFG ------------------------------ #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif log_facility=LOG_LOCAL0 loadmodule "xlog.so" Example xlog- xlog("L_INFO", "No more gateways \n"); Thanks, Dan-

5 10

KAMAILIO Config Hot/Live reload
by Omar 26 Mar '13

26 Mar '13

Kamailio Team. Was this feature added already or we still need to restart kamailio after we modify the Kamailio.cfg. O

3 4

str2ip6 bug is also in 3.3.3
by Juha Heinanen 26 Mar '13

26 Mar '13

i tried to compile latest 3.3 and found that it failed to compile due to same str2ip6 bug that was fixed in 4.0: dns_cache.c: In function ��dns_cache_do_request��: dns_cache.c:1894: warning: implicit declaration of function ��str2ip6�� -- juha

2 1

Help improving the modules readmes & docs
by Daniel-Constantin Mierla 25 Mar '13

25 Mar '13

Hello, I thought some people may have spare cycles and are willing to help with the documentation. Here is some updates I would like to get for modules: 1) adding ids for parameters, functions, and other sections of the docbook xml files. Now we generate the alphabetic indexes pointing to the online html readmes, would be better to point directly to the exact section. Practically, for example, sections such as next from async module: <section> <title><varname>workers</varname> (int)</title> should become: <section id="async.p.workers"> <title><varname>workers</varname> (int)</title> To get unique ids over all files, I suggest using following pattern: [module name] [dot] [type] [dot] [title] The [type] should be: - p - parameters - f - functions - m - mi commands - r - rpc commands - s - statistics 2) Add RPC commands to readmes -- some of modules already do, but many don't. There is a perl script that should get the list of RPC commands from source code, but it requires some patch to a perl lib, not working on all systems. Moreover, the output is quite minimalistic, not easy to improve the content or add examples. Last generated index is: - http://www.kamailio.org/docs/docbooks/3.4.x/rpc_list/rpc_list.html 3) Document selects -- although their name are quite suggestive, a bit of text around won't harm. - http://www.kamailio.org/wiki/cookbooks/devel/selects Because 1) and 2) requires git access, the easiest way to contribute is to provide patches to the xml files. If anyone commits to do an extensive work on these tasks in short term, we may eventually arrange for git access. The 3) is in the wiki, everyone can register an account and then update the pages. I would recommend these guidelines to be followed from now on by developers when adding new elements to the module documentation. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Acknowlidging BYE in advance using t_reply
by Efelin Novak 25 Mar '13

25 Mar '13

Hi, I'd like to ask you whether it is a good idea to acknowledge a BYE message in advance using t_reply("200", "OK") on my border proxy. Not acknowledged BYE messages by clients cause me problems in the middle of my network. Is there any problem with this behavior from Kamailio side, from script side or from SIP side? Do you have any experience with such logic manipulation? I am aware that this is not a proxy function, but it would solve many problems. My script is as follows: if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... t_newtran(); t_reply("200", "OK"); } route(RELAY); } My test shows that 200 OK from client are absorbed. Also if there is no reply from a client, Kamailio does not generate 408 Request Timeout. I'm using older version kamailio 3.1.0 (i386/linux) 21a375. Thanks for an advice Efelin

2 1

Does Kamailio work with ICE ?
by Khoa Pham 25 Mar '13

25 Mar '13

Hi, 1. Does Kamailio work with ICE? What does Kamailio do with ICE candidates in SDP ? 2. Does rtpproxy acts as TURN server ? Many thanks. Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 1

Store GeoIP in Accounting
by Ali Jawad 25 Mar '13

25 Mar '13

Hi I am storing accounting entries in the db, and I am also inserting some custom AVP values, however I also need to store the GeoIP value that I am extracting using gip. I did try the below, but it is not working, any other ideas how I could achieve this ? modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;CountryCode=$gip(src=>cc)") Regards

2 1

How to enable TLS in Kamailio ?
by Khoa Pham 25 Mar '13

25 Mar '13

Hi, I want to enable TLS in Kamailio, as in here http://kamailio.org/docs/modules/stable/modules/tls.html But how can I get the certificate and private key ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

7 14

Problems with Mar portion of registration
by mrichardson 22 Mar '13

22 Mar '13

Date: Fri, 15 Mar 2013 08:44:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Problems with Mar portion of registration [IMS] To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5142D14D.7080808(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, I haven't played with IMS modules, but you said is causing scscf to die -- is that a kamailio instance? What means 'to die' -- does it crash? If yes, can you get the syslog messages and the gdb backtrace? Cheers, Daniel ============================================ *Response:* ============================================ It appears any way we force the scscf to process a mar causes a crash. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 64 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 async_cdp_callback (is_timeout=<value optimized out>, param=0x7f641b158e70, maa=0x7f641b159040, elapsed_msecs=<value optimized out>) at cxdx_mar.c:414 414 start_reg_await_timer(avlist[0]); //start the timer to remove stale or unused Auth Vectors Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 libxml2-2.7.6-8.el6_3.4.x86_64 nss-pam-ldapd-0.7.5-18.el6.x86_64 pcre-7.8-6.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) syslog: 7(2211) DEBUG: tm [t_lookup.c:1657]: DEBUG: t_lookup_ident: transaction found 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_experimental_result_code: Failed finding avp 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_sip_number_auth_items: Failed finding avp 7(2211) DEBUG: <core> [msg_translator.c:206]: check_via_address(10.12.92.191, 10.12.92.191, 0) 7(2211) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 7(2211) DEBUG: tm [t_reply.c:1547]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 7(2211) DEBUG: tm [t_reply.c:706]: DEBUG: reply sent out. buf=0x7f64e586b950: SIP/2.0 403 Forbidde..., shmem=0x7f64de66c1b0: SIP/2.0 403 Forbidde 7(2211) DEBUG: tm [t_reply.c:716]: DEBUG: _reply_light: finished 0(2204) DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 0(2204) DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> 0(2204) DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:scscf.somecompany.com:6060> 0(2204) DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 0(2204) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 0(2204) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [34]; uri=[sip:bob@somecompany.com] 0(2204) DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["bob" <sip:bob@somecompany.com> ] 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.929b9c33.1>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 0(2204) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 0(2204) DEBUG: <core> [receive.c:149]: After parse_msg... 0(2204) DEBUG: <core> [receive.c:190]: preparing to run routing scripts... 0(2204) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.a11580a2575452ad784ca00f869952fb.0>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-333530-035cb8a7773aa017e8e89aa8db3c4f01>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 0(2204) DEBUG: <core> [parser/msg_parser.c:106]: found end of header 0(2204) DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1f9be4d3 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 0(2204) DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 0(2204) DEBUG: siputils [checks.c:103]: no totag 0(2204) DEBUG: <core> [select.c:425]: Calling SELECT 0x7f64e5839ea0 0(2204) DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffffffffffff 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil 0(2204) ERROR: <script>: Enter register block 0(2204) DEBUG: tm [t_lookup.c:1395]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_hooks.c:374]: DBG: trans=0x7f64de66c4d0, callback type 1, id 0 entered 0(2204) DEBUG: ims_auth [authorize.c:567]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:597]: Nonce or response missing: nonce len [1910316154], response16 len[-440163170] 0(2204) INFO: ims_auth [cxdx_mar.c:81]: created AVP successfully : [maa_return_code] - [-2] 0(2204) DEBUG: ims_auth [authorize.c:298]: Need to challenge for realm [somecompany.com] 0(2204) DEBUG: ims_auth [authorize.c:305]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:435]: Suspending SIP TM transaction 0(2204) DEBUG: ims_auth [authorize.c:1271]: Sending MAR 0(2204) DEBUG: cdp [session.c:276]: adding a session with id scscf.somecompany.com;1782936534;5 0(2204) DEBUG: cdp [routing.c:80]: get_first_connected_route in list 0x7f64de63dbc0 for app_id 16777216 and vendor_id 10415 0(2204) DEBUG: cdp [routing.c:88]: The peer hss.somecompany.com state is opened 0(2204) DEBUG: cdp [routing.c:63]: Checking if peer hss.somecompany.com handles application 16777216 for vendord 10415 0(2204) DEBUG: cdp [routing.c:90]: The peer hss.somecompany.com matches - will forward there 0(2204) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event Send_Message 0(2204) DEBUG: cdp [peerstatemachine.c:1096]: Snd_Message called to peer [hss.somecompany.com] for request with code 303 0(2204) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 0(2204) DEBUG: cdp [session.c:306]: no session found 0(2204) DEBUG: cdp [diameter_msg.c:83]: AAABuildMsgBuffer(): len=348 0(2204) DEBUG: cdp [receiver.c:967]: peer_send_msg(): Pipe push [0x7f64de66edd0] 0(2204) DEBUG: ims_auth [cxdx_mar.c:516]: Successfully sent async diameter 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) 0(2204) DEBUG: <core> [receive.c:293]: receive_msg: cleaning up 10(2214) DEBUG: cdp [receiver.c:743]: select_recv(): There is something on the send pipe 10(2214) DEBUG: cdp [receiver.c:756]: select_recv(): Send pipe says [0x7f64de66edd0] 8 10(2214) DEBUG: cdp [diameter_msg.c:412]: AAAFreeMessage: Freeing message (0x7f64de66edd0) 303 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [1] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:567]: receive_loop(): [hss.somecompany.com] Recv Version 1 Length 172 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [2] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:1061]: receive_message(): [hss.somecompany.com] Recv msg 303 10(2214) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event I_Rcv_Message 10(2214) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 10(2214) DEBUG: cdp [session.c:306]: no session found 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 0(2204) ALERT: <core> [main.c:788]: child process 2212 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on 0(2204) ALERT: <core> [main.c:788]: child process 2211 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on This is the modified register block to get a 401 unauthorized to even be generated (probably broken).... route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); if (!ims_www_authenticate(NETWORKNAME)) { ims_www_challenge("$td"); exit; } if ($avp(maa_return_code) != 1 ) { if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } }

3 6

Re: [SR-Users] 2 kamailio servers sharing one database and one dns name
by David Thomson 22 Mar '13

22 Mar '13

Hi Barry, Thanks for trying to help me out. Unfortunately your suggestion didn't work out. Not sure why. I tried a few other functions like t_relay_to. Guess I still have a lot to learn about this kamailio stuff in more complex setups. Any other ideas, anyone? Ttyl, Dave -----Original Message----- From: Barry Flanagan <barry(a)flanagan.ie> Date: Thu, 21 Mar 2013 12:01:18 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] 2 kamailio servers sharing one database and one dns name On 20 March 2013 03:05, David Thomson <jdavidthomson(a)hotmail.com <mailto:jdavidthomson@hotmail.com> > wrote: hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com <http://siptest.testdomain.com> with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows: User 1 registers to Server1 User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring. Hi I think you can use the t_replicate function within your register route so that each server will replicate the registration request to the other. I did this many years ago on OpenSER using the following just before the consume_credentials() line. On each of the servers you want to check that the src_ip is not that of the OTHER sever so they don't keep sending back and forth forever :-) So in the example below 10.0.1.1 is the IP of the other server. --------8<----------------------------------------- if (!src_ip==10.0.1.1) { t_replicate("udp:10.0.1.1:5060 <http://10.0.1.1:5060> "); xlog("L_INFO", "Replicating $ct, URI = $ru"); }; consume_credentials(); --------8<----------------------------------------- Have a look at http://www.kamailio.org/docs/modules/3.3.x/modules/tm.html#t_replicate Hope this helps. -Barry

1 0

GeoIP Issue
by Ali Jawad 21 Mar '13

21 Mar '13

Hi I did setup geoIP module and got the latest geoipcity free dat file and csv file and did setup all parameters in kamailio, when doing a test the instead of showing the country name the code is showing the region name, while region name is empty "I only need the country name", see : Config : if(geoip_match("$si", "src")) xlog("L_ERR","SIP message from: $gip(src=>cc) $gip(src=>regn) $gip(src=>regc)\n"); Log : SIP message from : A2 <null> <null> >From the CSV file of geoipcity just for cross reference : locId,country,region,city,postalCode,latitude,longitude,metroCode,areaCode 190132,"FR","A2","Lézardrieux","",48.7851,-3.1059,, As you can see above A2 is the region it is the 3rd column, src=>CC should show FR instead of A2. Working on Kamailio 3.3.4. Is there anything I missed here ? Regards

1 2

2 kamailio servers sharing one database and one dns name
by David Thomson 21 Mar '13

21 Mar '13

hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows:User 1 registers to Server1User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring. Any ideas? ttyl,Dave

3 2

New developer: Hugh Waite
by Daniel-Constantin Mierla 21 Mar '13

21 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Carlos Ruiz Díaz - he is going to add and maintain the credit control module, that was announced earlier this year on mailing list - the module is part of a framework that has three parts: - The module itself: https://github.com/caruizdiaz/cnxcc (to be added in Kamailio git repository) - The web interface: https://github.com/caruizdiaz/cnxcc-web - A helper script: https://github.com/caruizdiaz/cnxcc-db-sync His git commit id is: caruizdiaz My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 1

New developer: Carlos Ruiz Díaz
by Daniel-Constantin Mierla 21 Mar '13

21 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Carlos Ruiz Díaz - he is going to add and maintain the credit control module, that was announced earlier this year on mailing list - the module is part of a framework that has three parts: - The module itself: https://github.com/caruizdiaz/cnxcc (to be added in Kamailio git repository) - The web interface: https://github.com/caruizdiaz/cnxcc-web - A helper script: https://github.com/caruizdiaz/cnxcc-db-sync His git commit id is: caruizdiaz My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Need help with port fowards.
by Mini UN 20 Mar '13

20 Mar '13

Hi, was on the chat channel, so emailed as no activity. I have a new Cisco/Linksys EA-2700 and port forwarding is the same as my old Linksys WRT-610N. However my VoIP and website don't work. I tried DMZ aswell. Confused? Unfortunately, this router isn't compatibel for DD-WRT firmware. Thanks if anyone answers.

3 2

kamailio 4.0.0 and websockets
by Kelvin Chua 20 Mar '13

20 Mar '13

Hi guys, i have yet to finish my readings on the websocket standards but just wanted to fire away with this question. is the behavior of protocol conversions between UDP and TCP the same as if you include websockets? i have this twinkle issue (an old SIP stack) Received from: udp:192.168.122.100:5060 INVITE sip:kelvin@192.168.122.1 SIP/2.0 Record-Route: <sip:192.168.122.100;r2=on;lr=on> Record-Route: <sip:192.168.122.100:8080;transport=ws;r2=on;lr=on> Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0 Via: SIP/2.0/WS z173czhz21tk.invalid;rport=54765;received=192.168.122.1;branch=z9hG4bK3818745 Max-Forwards: 16 To: sip:kelvin@192.168.122.100 From: sip:kelvin2@192.168.122.100;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Contact: <sip:kelvin2@z173czhz21tk.invalid ;alias=192.168.122.1~54765~5;transport=ws;ob> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, SUBSCRIBE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.2.1 Content-Length: 2103 v=0 o=- 3148117784 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio m=audio 55736 RTP/SAVPF 103 104 111 0 8 106 105 13 126 c=IN IP4 192.168.122.1 a=rtcp:55736 IN IP4 192.168.122.1 a=candidate:2625852906 1 udp 2113937151 <cut off> --- +++ 16-3-2013 10:41:25.584732 INFO SIP ::send_sip_udp Send to: udp:192.168.122.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745 To: <sip:kelvin@192.168.122.100> From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Server: Twinkle/1.4.2 Content-Length: 0 --- +++ 16-3-2013 10:41:25.589231 INFO SIP ::send_sip_udp Send to: udp:192.168.122.100:5060 SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745 To: <sip:kelvin@192.168.122.100>;tag=pxtmo From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Server: Twinkle/1.4.2 Warning: 302 X340precise "Incompatible transport protocol" Content-Length: 0 Kelvin Chua

2 1

SIP debug.
by Victor V. Kustov 20 Mar '13

20 Mar '13

Good day. How could I debug SIP dialogs? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.2-calculate GNU/Linux

4 5

NAT & Firewall Issues
by Amr Elsharqawi 20 Mar '13

20 Mar '13

Hello, I am facing NAT & Firewall problems how I can solve this issue I've tried RTPProxy but voice quality is bad. any tutorial, please Amr

4 8

Re: [SR-Users] dialog ka_timer and ka_interval problem
by dolphin 20 Mar '13

20 Mar '13

Hello, Thank you for your reply I follow your suggestions to modify SHM_MEMORY ,PKG_MEMORY value and the system only two uac register. See the program after I start kamailio. shm memory 1G pkg memory 10M sh-3.00# ps -ef | grep kamailio root 23627 1 0 12:57 ? 00:00:00 /home/pkg/kamailio-3.3.4/sbin/kamailio -f /home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio root 23635 23627 0 12:57 ? 00:00:03 /home/pkg/kamailio-3.3.4/sbin/kamailio -f /home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio This problem is still the same last crazy the sending options message for uac But the /var/log/syslog not the same as follows Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime 2013/3/20 Rinor Hoxha <rinorhoxha(a)gmail.com> > Hi, > I had this "issue" some times ago. It turned out that I needed to increase > the value of SHM_MEMORY, since by mistake I forgot a 0 on the value (1024 > vs. 10240). However this happened only when high loads on system. > I needed also to tune PKG_MEMORY to my needs. ( both on > /etc/default/kamailio ) > > Check your settings and see if this helps. > > Br, Rinor > > > > On Tue, Mar 19, 2013 at 4:07 PM, dolphin <dolphinctk(a)gmail.com> wrote: > >> hi , >> >> i use kamailio 3.3.4 version >> >> I set and load dialog module >> >> modparam("dialog", "ka_timer", 10) >> modparam("dialog", "ka_interval", 20) >> >> use dlg_set_property("ka-src"); >> >> When uac A call uac B connection success , After 20 seconds, sip proxy >> send the OPTIONS message uac A >> >> Crazy sent to fill my bandwidth , I use ngrep tools see sip server >> >> U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296 >> OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0. >> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0. >> To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e. >> From: sip:xxxxxxx@xxx.xxx.xxx.xxx >> :5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O. >> CSeq: 1 OPTIONS. >> Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY.. >> Content-Length: 0. >> User-Agent: SIP Server. >> >> and /var/log/syslog >> >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> dialog [dlg_req_within.c:381]: failed to send the BYE request >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> dialog [dlg_req_within.c:381]: failed to send the BYE request >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> >> Please help me >> >> >> >> >> _______________________________________________ >> 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

Re: [SR-Users] 2 kamailio servers sharing one database and one dns name
by David Thomson 19 Mar '13

19 Mar '13

Thanks for the idea. After sending the original email I got user2 to call through to user1. Zrtp even was negotiated. So I think usrloc is sort of sharing the registration information but sometimes the call doesn't get sent through under certain circumstances. Maybe it's because user1 is behind a nat and rtpproxy is being used to keep its connection alive while user2 is not nat'd. I think it is almost working. -----Original Message----- From: David | StyleFlare <david(a)styleflare.com> Date: Wed, 20 Mar 2013 03:27:06 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] 2 kamailio servers sharing one database and one dns name I dont think it works like that, obviously the server 1 does not have a connection if the user is registered on server 2. On 3/19/13 11:05 PM, David Thomson wrote: hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows: User 1 registers to Server1 User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060> <udp:xxx.xxx.xxx.xxx:5060> ...ignoring. Any ideas? ttyl, Dave _______________________________________________ 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

1 0

sip-router mailing list
by Nikita Kozlov 19 Mar '13

19 Mar '13

Hello, I don't know if it's something wanted, but the email header "List-Id" has changed for both sr-dev and sr-users. The modifications : List-Id: "SIP Router - Kamailio" -> List-Id: "Kamailio $SER$ - Users Mailing List" List-Id: "Development mailing list of the sip-router project" -> List-Id: "Kamailio $SER$ - Development Mailing List" Regards, -- Nikita

2 1

dialog ka_timer and ka_interval problem
by dolphin 19 Mar '13

19 Mar '13

hi , i use kamailio 3.3.4 version I set and load dialog module modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 20) use dlg_set_property("ka-src"); When uac A call uac B connection success , After 20 seconds, sip proxy send the OPTIONS message uac A Crazy sent to fill my bandwidth , I use ngrep tools see sip server U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296 OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0. To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e. From: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O. CSeq: 1 OPTIONS. Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY.. Content-Length: 0. User-Agent: SIP Server. and /var/log/syslog Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: dialog [dlg_req_within.c:381]: failed to send the BYE request Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: dialog [dlg_req_within.c:381]: failed to send the BYE request Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Please help me

2 1

Kamailio not increasing cseq
by Camila Troncoso 19 Mar '13

19 Mar '13

Hi all, I’m working for a while with Kamailio+Freeswitch as SBC. I have this structure: * * * * * * * * * * * * * * * * * * When I make a call from one client connected to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing though a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , because the second gateway is also in the cloud, but when this happens, FreeSWITCH answer with : “482 Request merged” because it detects that is the same call. This is because the second INVITE has the same Call-ID and same Cseq. Kamailio *is not* increasing CSeq. Is there a way to resolve this? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 5

CentOS 6 crash
by Krishna Kurapati 19 Mar '13

19 Mar '13

Daniel, Thanks for the help. YES after disabling TLS for Kamailio, it is working fine. Is there a solution to have both TLS for Kamailio and SSL for LUA? Thanks Krish Kura

2 1

Multiple calls uac_replace_from()
by Alex Balashov 19 Mar '13

19 Mar '13

Forgive me if I failed to follow a recent discussion to its conclusion, but what is the bottom line on calling uac_replace_from() consecutively, in the course of processing one request? Is it possible, while using restore_mode 'auto'? Is there a correct way to do it, such as calling msg_apply_changes() after the first call but not the second one? If not, what is the alternative? -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

4 7

Kamailio 4.0 crashing in LUA on CentOS 6 - What could be wrong?
by Krishna Kurapati 19 Mar '13

19 Mar '13

36(8606) DEBUG: <core> [db.c:327]: connection 0x7fd474542320 found in pool 36(8606) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (32): sqlops 9(8567) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 36(8606) DEBUG: <core> [sr_module.c:674]: find_export_record: found <db_bind_api> in module db_mysql [/usr/local/lib64/kamailio/modules/db_mysql.so] 36(8606) DEBUG: <core> [db.c:213]: using db bind api for db_mysql 36(8606) DEBUG: <core> [db.c:327]: connection 0x7fd4745424f0 found in pool 36(8606) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (32): app_lua 36(8606) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 39(8615) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 34(8604) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 40(8616) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 41(8617) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 41(8617) DEBUG: <core> [tcp_main.c:3592]: DBG: handle_ser_child: dead child 6, pid 8558 (shutting down?) 0(8543) ALERT: <core> [main.c:788]: child process 8558 exited by a signal 11 0(8543) ALERT: <core> [main.c:791]: core was generated And here is backtrace done. Loaded symbols for /usr/lib64/lua/5.1/ssl.so Core was generated by `/usr/local/sbin/kamailio -L /usr/local/lib64/kamailio/modules/:/usr/local/lib64'. Program terminated with signal 11, Segmentation fault. #0 qm_detach_free (qm=0x7fd370304000, size=32) at mem/q_malloc.c:269 269 FRAG_END(next)->prev_free=prev; Missing separate debuginfos, use: debuginfo-install lua-5.1.4-4.1.el6.x86_64 (gdb) backtrace #0 qm_detach_free (qm=0x7fd370304000, size=32) at mem/q_malloc.c:269 #1 qm_malloc (qm=0x7fd370304000, size=32) at mem/q_malloc.c:386 #2 0x00007fd47052ae71 in shm_malloc (size=24) at ../../mem/shm_mem.h:262 #3 ser_malloc (size=24) at tls_init.c:285 #4 0x0000003928c5d95e in CRYPTO_malloc (num=24, file=0x3928d24a84 "o_names.c", line=195) at mem.c:306 #5 0x0000003928c6085f in OBJ_NAME_add (name=0x3928d27c52 "RSA-SHA512", type=1, data=0x3928d27c56 "SHA512") at o_names.c:195 #6 0x0000003928cc076a in EVP_add_digest (md=0x3928f84440) at names.c:99 #7 0x000000392b442f18 in SSL_library_init () at ssl_algs.c:165 #8 0x00007fd36ea1d3c9 in luaopen_ssl_core () from /usr/lib64/lua/5.1/ssl.so #9 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #10 0x00007fd36f678a54 in ?? () from /usr/lib64/liblua-5.1.so #11 0x00007fd36f673d56 in lua_call () from /usr/lib64/liblua-5.1.so #12 0x00007fd36f68cf2c in ?? () from /usr/lib64/liblua-5.1.so #13 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #14 0x00007fd36f683249 in ?? () from /usr/lib64/liblua-5.1.so #15 0x00007fd36f678a9d in ?? () from /usr/lib64/liblua-5.1.so #16 0x00007fd36f673d56 in lua_call () from /usr/lib64/liblua-5.1.so #17 0x00007fd36f68cf2c in ?? () from /usr/lib64/liblua-5.1.so #18 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #19 0x00007fd36f683249 in ?? () from /usr/lib64/liblua-5.1.so #20 0x00007fd36f678a9d in ?? () from /usr/lib64/liblua-5.1.so ---Type <return> to continue, or q <return> to quit--- #21 0x00007fd36f678137 in ?? () from /usr/lib64/liblua-5.1.so #22 0x00007fd36f6781b2 in ?? () from /usr/lib64/liblua-5.1.so #23 0x00007fd36f673b61 in lua_pcall () from /usr/lib64/liblua-5.1.so #24 0x00007fd36f89e69d in lua_sr_init_child () at app_lua_api.c:227 #25 0x00000000004e4aa3 in init_mod_child (m=0x7fd474485060, rank=-1) at sr_module.c:893 #26 0x00000000004e4a24 in init_mod_child (m=0x7fd474485980, rank=-1) at sr_module.c:890 #27 0x00000000004e4a24 in init_mod_child (m=0x7fd474485bf0, rank=-1) at sr_module.c:890 #28 0x0000000000489d15 in fork_process (child_id=-1, desc=<value optimized out>, make_sock=<value optimized out>) at pt.c:347 #29 0x0000000000466dec in main_loop () at main.c:1700 #30 0x0000000000468afe in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2566

2 1

Survey on SIP overload control
by Yang Hong 18 Mar '13

18 Mar '13

Dear all, SIP overload may block too many calls unnecessarily. "Requirements for Management of Overload in the Session Initiation Protocol. IETF RFC 5390." http://www.ietf.org/rfc/rfc5390.txt Build-in overload mechanism cannot prevent overload effectively. Therefore, IETF SIP Overload Control (soc) Working Group works on IETF RFC "SIP Overload Control" currently. http://tools.ietf.org/html/draft-ietf-soc-overload-control-12 IETF SIP Overload Control discussion archives http://www.ietf.org/mail-archive/web/sip-overload/current/threads.html#00874 We have published a survey on SIP overload control algorithms (including IETF RFC "SIP Overload Control"). http://www.researchgate.net/publication/231609451_A_Comparative_Study_of_SI… http://arxiv.org/abs/1210.1505 http://www.igi-global.com/chapter/comparative-study-sip-overload-control/67… We have also used control theorectic approaches to model the interactions between an overloaded SIP server and its upstream servers as a feedback control system in two different scenarios - redundant retransmission ratio control and round trip delay control (IEEE Globecom 2010 and ICC 2011). "Mitigating SIP Overload Using a Control-Theoretic Approach" IEEE Globecom 2010 http://www.researchgate.net/publication/221284946_Mitigating_SIP_Overload_U… http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=5683124&url=http%3A%2… Implementation of implicit SIP overload control in the real system "An Efficient Earthquake Early Warning Message Delivery Algorithm Using an in Time Control-Theoretic Approach" http://link.springer.com/chapter/10.1007%2F978-3-642-23641-9_15# "Design Of A PI Rate Controller for Mitigating SIP Overload" IEEE ICC 2011 http://www.researchgate.net/publication/224249824_Design_of_a_PI_Rate_Contr… http://features.rr.com/article/01vBgtc8TR17z?q=Skype http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=5963029&url=http%3A%2… Best regards, Winston Hong Software Engineer InBay Technologies Inc. Ottawa, Canada K2K 1Y3 http://www.inbaytech.com/

1 0

Kamailio, NAT and RTPProxy
by Amr Elsharqawi 18 Mar '13

18 Mar '13

Dears, Finally I've got Kamailio 4 running on FreeBSD 9 with MySQL. in the beginning it was working fine with NAT enabled, but after I've configured NAT & RTPProxy users can register, ring but voice is very bad. here is my kamailio.cfg #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: #!define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql #!define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql #!define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: #!define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _._._._ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql #!define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://sip:s510Sg4lrw@localhost/sip" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 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 /* add local domain aliases */ alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:XXXXXXXXX:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!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 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.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" #!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.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 # ----------------- 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) # ----- 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", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:22222") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle 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); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # 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()) { 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; } } # 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); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server 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_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # 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; rtpproxy_manage(); 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 (!uri==myself) { 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}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($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) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif }

1 0

Runaway TCP Log
by Krishna Kurapati 18 Mar '13

18 Mar '13

Hi, My Log filled to 20G overnight with these Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files^C I looked at the code segment it is here where it is logging. new_sock=accept(si->socket, &(su.s), &su_len); if (unlikely(new_sock==-1)){ if ((errno==EAGAIN)||(errno==EWOULDBLOCK)) return 0; LOG(L_ERR, "WARNING: handle_new_connect: error while accepting" " connection(%d): %s\n", errno, strerror(errno)); return -1; } I am using 3.3.x Is this an issue with Kamailio or something else? Appreciate any help regarding this matter Krish Kura

3 3

Kamailio strange behavior : bad Via, bad port
by Khoa Pham 18 Mar '13

18 Mar '13

Hi, When A calls B. The INVITE goes through Kamailio. In the Kamalio 's log file, I see it say "bad Via", bad port. And it doesnot reply A with 100 Trying not forwarding INVITE to B. The INVITE message (espcially the Via header) is OK. This problem sometimes happens. Do you know why ? Please help. Thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 4

Kamailio not send 100/180 for INVITE !
by Khoa Pham 18 Mar '13

18 Mar '13

Hi, When A calls B. Kamailio will handle A's INVITE message. But I see that B doesnot receive the INVITE. And A doesnot receive any 100/180 privisional message. Why does Kamailio not send 100/180 ? Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

How does Kamailio handle INVITE message by default ?
by Khoa Pham 16 Mar '13

16 Mar '13

Hi, After asking many questions, I haven't got any clues about how Kamailio handles INVITE message by default, in terms of modifying c= line in SDP According to rtpproxy flow http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf When client register, SIP proxy will call nat_uac_test() to detected if client is NATed or not, then save this info. When client A calls client B, the INVITE message will go through SIP proxy. Here the SIP proxy can do 3 things (as in section "INVITEs behind NAT" in the pdf). 1. Add an SDP command direction:active to the SDP content 2. Change the c= line to a.b.c.d 3. Force RTP to go through a proxy by changing the c-line to c=IN IP4 address-of-proxy and the m-line to m=audio port-on-proxy RTP/AVP 0 101. When will SER do 2, 3 ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 3

Kamailio only allowing traffic with machines in the dispatcher.cfg?
by Benjamin Henrion 16 Mar '13

16 Mar '13

Hi, Does someone know/uses a simple rule so that Kamailio only exchanges traffic with machines in the dispatcher? Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

6 5

Re: [SR-Users] sr-users Digest, Vol 94, Issue 51 - Problem with Mar portion of registration with Kamailio ims modules
by mrichardson 15 Mar '13

15 Mar '13

Date: Fri, 15 Mar 2013 08:44:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Problems with Mar portion of registration [IMS] To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5142D14D.7080808(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, I haven't played with IMS modules, but you said is causing scscf to die -- is that a kamailio instance? What means 'to die' -- does it crash? If yes, can you get the syslog messages and the gdb backtrace? Cheers, Daniel ============================================ *Response:* ============================================ It appears any way we force the scscf to process a mar causes a crash. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 64 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 async_cdp_callback (is_timeout=<value optimized out>, param=0x7f641b158e70, maa=0x7f641b159040, elapsed_msecs=<value optimized out>) at cxdx_mar.c:414 414 start_reg_await_timer(avlist[0]); //start the timer to remove stale or unused Auth Vectors Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 libxml2-2.7.6-8.el6_3.4.x86_64 nss-pam-ldapd-0.7.5-18.el6.x86_64 pcre-7.8-6.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) syslog: 7(2211) DEBUG: tm [t_lookup.c:1657]: DEBUG: t_lookup_ident: transaction found 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_experimental_result_code: Failed finding avp 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_sip_number_auth_items: Failed finding avp 7(2211) DEBUG: <core> [msg_translator.c:206]: check_via_address(10.12.92.191, 10.12.92.191, 0) 7(2211) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 7(2211) DEBUG: tm [t_reply.c:1547]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 7(2211) DEBUG: tm [t_reply.c:706]: DEBUG: reply sent out. buf=0x7f64e586b950: SIP/2.0 403 Forbidde..., shmem=0x7f64de66c1b0: SIP/2.0 403 Forbidde 7(2211) DEBUG: tm [t_reply.c:716]: DEBUG: _reply_light: finished 0(2204) DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 0(2204) DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> 0(2204) DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:scscf.somecompany.com:6060> 0(2204) DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 0(2204) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 0(2204) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [34]; uri=[sip:bob@somecompany.com] 0(2204) DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["bob" <sip:bob@somecompany.com> ] 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.929b9c33.1>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 0(2204) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 0(2204) DEBUG: <core> [receive.c:149]: After parse_msg... 0(2204) DEBUG: <core> [receive.c:190]: preparing to run routing scripts... 0(2204) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.a11580a2575452ad784ca00f869952fb.0>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-333530-035cb8a7773aa017e8e89aa8db3c4f01>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 0(2204) DEBUG: <core> [parser/msg_parser.c:106]: found end of header 0(2204) DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1f9be4d3 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 0(2204) DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 0(2204) DEBUG: siputils [checks.c:103]: no totag 0(2204) DEBUG: <core> [select.c:425]: Calling SELECT 0x7f64e5839ea0 0(2204) DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffffffffffff 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil 0(2204) ERROR: <script>: Enter register block 0(2204) DEBUG: tm [t_lookup.c:1395]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_hooks.c:374]: DBG: trans=0x7f64de66c4d0, callback type 1, id 0 entered 0(2204) DEBUG: ims_auth [authorize.c:567]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:597]: Nonce or response missing: nonce len [1910316154], response16 len[-440163170] 0(2204) INFO: ims_auth [cxdx_mar.c:81]: created AVP successfully : [maa_return_code] - [-2] 0(2204) DEBUG: ims_auth [authorize.c:298]: Need to challenge for realm [somecompany.com] 0(2204) DEBUG: ims_auth [authorize.c:305]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:435]: Suspending SIP TM transaction 0(2204) DEBUG: ims_auth [authorize.c:1271]: Sending MAR 0(2204) DEBUG: cdp [session.c:276]: adding a session with id scscf.somecompany.com;1782936534;5 0(2204) DEBUG: cdp [routing.c:80]: get_first_connected_route in list 0x7f64de63dbc0 for app_id 16777216 and vendor_id 10415 0(2204) DEBUG: cdp [routing.c:88]: The peer hss.somecompany.com state is opened 0(2204) DEBUG: cdp [routing.c:63]: Checking if peer hss.somecompany.com handles application 16777216 for vendord 10415 0(2204) DEBUG: cdp [routing.c:90]: The peer hss.somecompany.com matches - will forward there 0(2204) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event Send_Message 0(2204) DEBUG: cdp [peerstatemachine.c:1096]: Snd_Message called to peer [hss.somecompany.com] for request with code 303 0(2204) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 0(2204) DEBUG: cdp [session.c:306]: no session found 0(2204) DEBUG: cdp [diameter_msg.c:83]: AAABuildMsgBuffer(): len=348 0(2204) DEBUG: cdp [receiver.c:967]: peer_send_msg(): Pipe push [0x7f64de66edd0] 0(2204) DEBUG: ims_auth [cxdx_mar.c:516]: Successfully sent async diameter 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) 0(2204) DEBUG: <core> [receive.c:293]: receive_msg: cleaning up 10(2214) DEBUG: cdp [receiver.c:743]: select_recv(): There is something on the send pipe 10(2214) DEBUG: cdp [receiver.c:756]: select_recv(): Send pipe says [0x7f64de66edd0] 8 10(2214) DEBUG: cdp [diameter_msg.c:412]: AAAFreeMessage: Freeing message (0x7f64de66edd0) 303 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [1] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:567]: receive_loop(): [hss.somecompany.com] Recv Version 1 Length 172 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [2] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:1061]: receive_message(): [hss.somecompany.com] Recv msg 303 10(2214) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event I_Rcv_Message 10(2214) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 10(2214) DEBUG: cdp [session.c:306]: no session found 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 0(2204) ALERT: <core> [main.c:788]: child process 2212 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on 0(2204) ALERT: <core> [main.c:788]: child process 2211 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on This is the modified register block to get a 401 unauthorized to even be generated (probably broken).... route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); if (!ims_www_authenticate(NETWORKNAME)) { ims_www_challenge("$td"); exit; } if ($avp(maa_return_code) != 1 ) { if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } } ============================================ On 03/15/2013 03:06 AM, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. FreeBSD build from git: i use that patch (Victor V. Kustov) > 2. Re: Problems with Mar portion of registration [IMS] > (Daniel-Constantin Mierla) > 3. Re: UAC module and SRV records (Daniel-Constantin Mierla) > 4. Re: registrar and check registered (Daniel-Constantin Mierla) > 5. Re: FreeBSD build from git: i use that patch > (Daniel-Constantin Mierla) > 6. Re: FreeBSD build from git: i use that patch (Victor V. Kustov) > 7. Re: FreeBSD build from git: i use that patch > (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 15 Mar 2013 11:43:55 +0400 > From: "Victor V. Kustov" <coyote(a)bks.tv> > Subject: [SR-Users] FreeBSD build from git: i use that patch > To: "Kamailio $SER$ - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <20130315114355.2409d851@bkstv> > Content-Type: text/plain; charset=US-ASCII > > diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c > index faabd85..24686e7 100644 > --- a/modules/ims_icscf/cxdx_lir.c > +++ b/modules/ims_icscf/cxdx_lir.c > @@ -51,6 +51,11 @@ > #include "mod.h" > #include "location.h" > > +#if defined (__OS_freebsd) > +#include "sys/limits.h" > +#define MAXINT INT_MAX > +#endif > + > //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable > int create_lia_return_code(int result) { > int rc; > diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c > index bd5216c..8b6b79c 100644 > --- a/modules/ims_icscf/scscf_list.c > +++ b/modules/ims_icscf/scscf_list.c > @@ -47,6 +47,11 @@ > #include "db.h" > #include "../../lib/ims/useful_defs.h" > > +#if defined (__OS_freebsd) > +#include "sys/limits.h" > +#define MAXINT INT_MAX > +#endif > + > extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list > > extern struct tm_binds tmb; //Structure with pointers to tm funcs > > > -- > SY, > Victor > JID: coyote(a)bks.tv > JID: coyote(a)bryansktel.ru > I use FREE operation system: 3.8.3-calculate GNU/Linux > > > > ------------------------------ > > Message: 2 > Date: Fri, 15 Mar 2013 08:44:13 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Problems with Mar portion of registration > [IMS] > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D14D.7080808(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > I haven't played with IMS modules, but you said is causing scscf to die > -- is that a kamailio instance? What means 'to die' -- does it crash? If > yes, can you get the syslog messages and the gdb backtrace? > > Cheers, > Daniel > > On 3/13/13 3:01 PM, mrichardson wrote: >> The MAR portion of registration is either not happening or failing >> causing scscf to die. Using Kamailio 4.0 with the ims_auth module. >> We're trying to determine what >> the REGISTER block should look like and what ims_auth parameters are >> necessary for a successful MAR challenge with auth vector delivery. We >> currently only get the UAR by ICSCF then the SAR by SCSCF. >> >> Module parameters: >> # -- ims_auth params -- >> modparam("ims_auth", "name", URI) >> modparam("ims_auth", "registration_default_algorithm", >> REG_AUTH_DEFAULT_ALG) >> modparam("ims_auth","ignore_failed_auth",0) >> #!ifdef CXDX_FORCED_PEER >> modparam("ims_auth", "cxdx_forced_peer", CXDX_FORCED_PEER) >> #!endif >> modparam("ims_auth", "cxdx_dest_realm", NETWORKNAME) >> modparam("ims_auth", "av_request_at_once", 1) >> modparam("ims_auth", "registration_qop", "auth") >> >> route[REGISTER] { >> xlog("L_ERR", "Enter register block"); >> t_newtran(); >> >> ims_www_authenticate(NETWORKNAME); >> #check to see if user is authenticated - ie sip header has auth >> information - (already challenged) >> if ($avp(maa_return_code) == 1) { >> # user has not been authenticated. Lets send a challenge >> via 401 Unauthorized >> ims_www_challenge("$td"); >> exit; >> } else { >> # We need to check if this user is registered or not >> if (!impu_registered("location")) { >> save("location"); >> if ($avp(saa_return_code) == 1) { >> isc_match_filter_reg("0","location"); >> exit; >> } >> } else { >> save("location"); >> if($avp(saa_return_code) == 1) { >> isc_match_filter_reg("1","location"); >> exit; >> } >> } >> } >> } >> >> >> 12(22978) INFO: cdp [receiver.c:910]: peer_connect(): Peer >> hss.ims.somecompany.com:3868 connected >> 0(22966) ERROR: <script>: Enter register block 0(22966) INFO: >> ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : >> [saa_return_code] - [-2] >> 5(22971) INFO: ims_registrar_scscf [cxdx_avp.c:138]: >> cxdx_get_experimental_result_code: Failed finding avp >> 5(22971) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP >> successfully : [saa_return_code] - [1] >> 5(22971) WARNING: tm [t_lookup.c:1559]: WARNING: script writer didn't >> release transaction >> >> _______________________________________________ >> 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 > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 3 > Date: Fri, 15 Mar 2013 08:48:53 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] UAC module and SRV records > To: "Olle E. Johansson" <oej(a)edvina.net> > Cc: "Kamailio $SER$ - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D265.3060104(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 3/13/13 10:00 AM, Olle E. Johansson wrote: >> 13 mar 2013 kl. 09:35 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> On 3/13/13 9:25 AM, Olle E. Johansson wrote: >>>> 13 mar 2013 kl. 09:08 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>> >>>>> Hello, >>>>> >>>>> do you set the port in r-uri/dst-uri? >>>> No. That would invalidate SRV record lookup, right? >>> yes. >>> >>> Have you watched the dns traffic? Is directly A/AAAA query? Afaik, this function should use just the usual logic to determine the next hop based on dst-uri/r-uri. >> I will debug in more detail. Had to set an outbound URI with a hostname and a port number to get it to work. > There could be the possibility that some of the functions in the chain > of uac_req_send() set port to default 5060 if not set explicitly, but > from sending point of view, it should the same as for the rest of > messages. Thus a look over the code should be done as well. I'll do it > soon if nobody gets the time meanwhile. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 4 > Date: Fri, 15 Mar 2013 08:53:29 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] registrar and check registered > To: "Victor V. Kustov" <coyote(a)bks.tv>, sr-users(a)lists.sip-router.org > Message-ID: <5142D379.20402(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > On 3/13/13 10:00 AM, Victor V. Kustov wrote: >> Hello, Daniel. Thanks for reply. >> >>> do you want to check if caller (origin) is registered or if callee >>> (destination) is registered? Like one can call to pstn only if has a >>> phone registered for itself? >> In our scheme all calls forwards to PSTN GW, if that call from our to >> our users, GW turn call back and kamailio receive it as incoming. >> So, no "local calls" in this scheme. >> >> For outgoing calls (to PSTN) i want check caller (origin) is registered. >> For incoming calls (from PSTN) i want callee is registered. >> >> Need cache registrations (DB, memcached or something). I try that way: >> 1. REGISTER - if ok, than we save("location"). >> 2. Other methods - check !registered("location") or !lookup("location"). >> But in INVITE seems both !registered("location") >> and !lookup("location") return TRUE. I dont understand why. >> >> In topicstart letter i citate part of my cfg: in INVITE i check >> registration again and it work - registered users may call to PSTN, >> unregged may not. But no cache for registration and look as overhead. >> Must be more "beauty" solution. > when you need to check if caller is registered use: > > - registered("location", "$fu") > See more details at: > - http://kamailio.org/docs/modules/stable/modules/registrar.html#id2512620 > > If used without the second parameter is checking if R-URI ($ru) is > registered. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 5 > Date: Fri, 15 Mar 2013 08:55:28 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] FreeBSD build from git: i use that patch > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D3F0.4090408(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > thanks for checking on bsd, can you please sent the patch as attachment? > Because it can saved locally and applied easier. > > Cheers, > Daniel > > On 3/15/13 8:43 AM, Victor V. Kustov wrote: >> diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c >> index faabd85..24686e7 100644 >> --- a/modules/ims_icscf/cxdx_lir.c >> +++ b/modules/ims_icscf/cxdx_lir.c >> @@ -51,6 +51,11 @@ >> #include "mod.h" >> #include "location.h" >> >> +#if defined (__OS_freebsd) >> +#include "sys/limits.h" >> +#define MAXINT INT_MAX >> +#endif >> + >> //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable >> int create_lia_return_code(int result) { >> int rc; >> diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c >> index bd5216c..8b6b79c 100644 >> --- a/modules/ims_icscf/scscf_list.c >> +++ b/modules/ims_icscf/scscf_list.c >> @@ -47,6 +47,11 @@ >> #include "db.h" >> #include "../../lib/ims/useful_defs.h" >> >> +#if defined (__OS_freebsd) >> +#include "sys/limits.h" >> +#define MAXINT INT_MAX >> +#endif >> + >> extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list >> >> extern struct tm_binds tmb; //Structure with pointers to tm funcs >> >> > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 6 > Date: Fri, 15 Mar 2013 12:01:37 +0400 > From: "Victor V. Kustov" <coyote(a)bks.tv> > Subject: Re: [SR-Users] FreeBSD build from git: i use that patch > To: sr-users(a)lists.sip-router.org > Message-ID: <20130315120137.72b404c4@bkstv> > Content-Type: text/plain; charset="utf-8" > > ? Fri, 15 Mar 2013 08:55:28 +0100 > Daniel-Constantin Mierla <miconda(a)gmail.com> ?????: > >> Hello, >> >> thanks for checking on bsd, can you please sent the patch as >> attachment? Because it can saved locally and applied easier. >> > > > > -- > SY, > Victor > JID: coyote(a)bks.tv > JID: coyote(a)bryansktel.ru > I use FREE operation system: 3.8.2-calculate GNU/Linux >

1 0

FreeBSD build from git: i use that patch
by Victor V. Kustov 15 Mar '13

15 Mar '13

diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c index faabd85..24686e7 100644 --- a/modules/ims_icscf/cxdx_lir.c +++ b/modules/ims_icscf/cxdx_lir.c @@ -51,6 +51,11 @@ #include "mod.h" #include "location.h" +#if defined (__OS_freebsd) +#include "sys/limits.h" +#define MAXINT INT_MAX +#endif + //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable int create_lia_return_code(int result) { int rc; diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c index bd5216c..8b6b79c 100644 --- a/modules/ims_icscf/scscf_list.c +++ b/modules/ims_icscf/scscf_list.c @@ -47,6 +47,11 @@ #include "db.h" #include "../../lib/ims/useful_defs.h" +#if defined (__OS_freebsd) +#include "sys/limits.h" +#define MAXINT INT_MAX +#endif + extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list extern struct tm_binds tmb; //Structure with pointers to tm funcs -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.3-calculate GNU/Linux

3 4

registrar and check registered
by Victor V. Kustov 15 Mar '13

15 Mar '13

Hi. I try to use Kamailio 4.1 git version as registrar and PSTN gw. Need accept register from users. Its OK, I made it and use save() function - from keep registrations in case of kamailio restart. Need forward calls to gw, but only for registered users. If I understand correctly in INVITE enough check registered("location"), isnt it? but it's not work, i need verify auth again. Routing part of my config: route { # initial sanity checks - skip in letter # We are registrar! if (is_method("REGISTER")) { sl_send_reply("100", "Trying"); # send "Trying, dont threat on me if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } # we must check auth right now # WORK: if (!www_authorize("$fd", "subscriber")) if (!auth_check("$fd", "subscriber", "1")) { # WORK: www_challenge("$fd", "0"); auth_challenge("$fd", "0"); # send me requisites plz exit; } else { # ok. auth accepted, save in location if (save("location")) #if (save("location","0x04","$fd")) { xlog("L_WARN", "Save OK in location sip:$au@$fd"); } else { xlog("L_WARN", "Save FAIL in location sip:$au@$fd"); } } xlog("L_WARN", "REGISTERED method done"); exit; } if (is_method("INVITE")) { if ($si == $sel(cfg_get.pstn.gw_ip)) { # incoming calls xlog("L_WARN", "Incoming call"); } else { # outgoing calls xlog("L_WARN", "Outgoing call"); # check auth # we must check auth right now if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } consume_credentials(); ### THAT FUNCTIONS ALWAYS WRITE IN LOGS # "!registered" "!lookup" for both registered and unregged users # if (!registered("location")) xlog("L_WARN","!registered"); # if (!lookup("location")) xlog("L_WARN", "!lookup"); # number change - FIX late # check may be used dialplan automatically? # route $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_WARN", "Rewrite to $ru, send to PSTN."); if (!t_relay()) { sl_reply_error(); } } } else { # reply 2 keepalive and drop all # if (!t_relay()) { sl_reply_error(); } } } -- SY, Victor V. Kustov +7 (952) 966 76 24 JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.7.7-calculate GNU/Linux

2 3

UAC module and SRV records
by Olle E. Johansson 15 Mar '13

15 Mar '13

Hi! When using the UAC module to send a request to a domain that only exists in SRV records, I get an error message that no A or AAAA records can be found for the domain. Is this by design or possibly a bug? Anyone else that have seen this? /O

2 5

Problems with Mar portion of registration
by mrichardson 15 Mar '13

15 Mar '13

The MAR portion of registration is either not happening or failing causing scscf to die. Using Kamailio 4.0 with the ims_auth module. We're trying to determine what the REGISTER block should look like and what ims_auth parameters are necessary for a successful MAR challenge with auth vector delivery. We currently only get the UAR by ICSCF then the SAR by SCSCF. Module parameters: # -- ims_auth params -- modparam("ims_auth", "name", URI) modparam("ims_auth", "registration_default_algorithm", REG_AUTH_DEFAULT_ALG) modparam("ims_auth","ignore_failed_auth",0) #!ifdef CXDX_FORCED_PEER modparam("ims_auth", "cxdx_forced_peer", CXDX_FORCED_PEER) #!endif modparam("ims_auth", "cxdx_dest_realm", NETWORKNAME) modparam("ims_auth", "av_request_at_once", 1) modparam("ims_auth", "registration_qop", "auth") route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); ims_www_authenticate(NETWORKNAME); #check to see if user is authenticated - ie sip header has auth information - (already challenged) if ($avp(maa_return_code) == 1) { # user has not been authenticated. Lets send a challenge via 401 Unauthorized ims_www_challenge("$td"); exit; } else { # We need to check if this user is registered or not if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } } 12(22978) INFO: cdp [receiver.c:910]: peer_connect(): Peer hss.ims.somecompany.com:3868 connected 0(22966) ERROR: <script>: Enter register block 0(22966) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : [saa_return_code] - [-2] 5(22971) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_experimental_result_code: Failed finding avp 5(22971) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : [saa_return_code] - [1] 5(22971) WARNING: tm [t_lookup.c:1559]: WARNING: script writer didn't release transaction

2 1

A weird message looping scenario
by Ajay Sabat 15 Mar '13

15 Mar '13

Hi, We have an ugly message looping scenario where a SIP request message received at Kamailio is forwarded to the switch/router and back to Kamailio in a loop until the Max-forwards header value becomes 0 or the transaction times out. After receiving an incoming INVITE request from a local user, Kamailio forwards the INVITE request to another proxy located in a different network. The call is established successfully. During the call setup the external proxy sends an OPTIONS request to Kamailio (actually to the external public IP address from where it received INVITE request from Kamailio). The OPTIONS request arrives at Kamailio. But the request uri contains the mapped public address instead of the local ip address where Kamailio is listening. So Kamailio forwards the request to the public IP address (i.e. the switch) which again sends it back to Kamailio again and it keeps looping. Here is the message flow. 192.168.1.3 is the local user registered to Kamailio which is running at 192.168.1.5. IP address of the external proxy server is 10.139.90.161 and 10.139.90.137 is the public IP address of the switch/router. NAT isn't enabled at Kamailio. Internet Protocol Version 4, Src: 192.168.1.3 (192.168.1.3), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:1002@video-conf SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1001@192.168.1.3> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 510 Message Body Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 192.168.1.3 (192.168.1.3) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Server: kamailio (3.3.3 (i386/linux)) Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.161 (10.139.90.161) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:1002@vc-vcs-control.air2gnd.net SIP/2.0 Message Header Record-Route: <sip:192.168.1.5;lr=on> Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1001@192.168.1.3> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 69 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 510 Message Body Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 101 Dialog Establishement Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 Record-Route: <sip:192.168.1.5;lr=on> From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@ video-conf>;tag=689763673 Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1002@10.139.90.161:5060> Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 192.168.1.3 (192.168.1.3) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 101 Dialog Establishement Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 Record-Route: <sip:192.168.1.5;lr=on> From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf>;tag=689763673 Call-ID: 1110783207 CSeq: 20 INVITE Contact: sip:1002@10.139.90.161:5060 Content-Length: 0 P-Received: 10.139.90.137 Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.139.90.161:5060;rport;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.161>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.137 (10.139.90.137) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 69 Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.137 (10.139.90.137), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "uuuu" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 69 Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.137 (10.139.90.137) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.0b94bab7.0 Via: SIP/2.0/UDP 192.168.1.5;received=10.139.90.137;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 68 Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.137 (10.139.90.137), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.0b94bab7.0 Via: SIP/2.0/UDP 192.168.1.5;received=10.139.90.137;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 68 Content-Length: 0 This message forwarding goes on for a long time in a loop, each time with an extra Via header and Max-Forwards value decremented by one. After sometime it eventually times out and Kamailio starts sending 408 response for the OPTIONS request which again keeps looping. Probably adding an alias for the public IP address i.e., 10.139.90.137 in the Kamailio config file would solve the problem. But this IP address is not fixed and may change and I guess there is no way Kamailio can learn the public IP address on its own. Any help or reference to previous posts if it is already discussed and solved will be much helpful. Regards, Ajay

2 1

What factor determine NATed or not ?
by Khoa Pham 15 Mar '13

15 Mar '13

Hi, When we send REGISTER or INVITE to Kamailio, which part of the request message does Kamailio use to determine if client is NATed or not ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 2

search_hf returns always true
by Ajay Sabat 15 Mar '13

15 Mar '13

Hi, I want to check whether a parameter exists in a header and I am using the search_hf() function defined in textops module. I am surprised that is always returning true whether the searched pattern exists or not. To be specific, I need to check if the "received" parameter exists in the top most via header and this is how I do it. The statements inside the if block get executed all the time irrespective of whether the received parameter exists in the top most via or not. Any help what am I doing wrong here? if (search_hf("Via", "received=", "f")) { xdbg("received param $sel(via.received) in top most via header \n"); # perform necessary action } Case 1: no received parameter in via Via header in response message looks like: Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK39bc.c1076d93.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK56870034 And the corresponding debug message from Kamailio is: ./sbin/kamailio[22600]: DEBUG: <script>: received param <null> in top most via header Case 2: received parameter exists Via header in response message looks like: Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK006e.18417697.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK960971659 And the corresponding debug message: ./sbin/kamailio[22600]: DEBUG: <script>: received param 10.139.90.137 in top most via header Thanks, Ajay

2 1

testing Supported and Require keywords
by Juha Heinanen 14 Mar '13

14 Mar '13

is there currently a proper way to test if a particular keyword appears in Supported or Require header or is a regex test on $hdr(X)[*] the only possibility? -- juha

3 3

Kamailio World Updates - March 14
by Daniel-Constantin Mierla 14 Mar '13

14 Mar '13

Hello, the groups of speakers at Kamailio World Conference got new members, among them Prof. Dr. Thomas Magedantz from Technische Universitat Berlin/FhG Fokus and Alan Duric, a pioneer in VoIP, among other positions being co-founder of Telio, more details at: - http://conference.kamailio.com/k01/speakers/ It is about one month till the start of the event, agenda is pretty much done, with possibility of few updates in the near future. Regarding Kamailio project, Monday we released v4.0.0, with really good feedback so far. Thank you again to everyone making it possible. New contributions started to show up on master branch, once you are done upgrading to v4.0.0, don't forget to give a try to development version and check the list of new additions. If you haven't registered yet for the Kamailio World conference, you should hurry up, the speakers and the other participants form an excellent group of professionals working with rich communication services and products, it is your chance to meet with them and get updated on the field: - http://conference.kamailio.com/k01/registration/ Also, if you consider sponsoring or participate to surrounding exhibition space, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ We are glad to announce the first group of participating companies with all day availability for demos and presentations of their products and services: Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. The list will be updated soon with more participants. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

I added the append_hf() inside the if condition in oureply_route[] block as given below, and it is working. I still want to learn how to store data in a variable to use it later in another block. Thanks, Ajay Hi Daniel, Thanks for your response. But I don't want the IP in the received parameter of the incoming request. Instead, I want to read it (i.e. the public IP address of my own network where the local user and Kamailio are located) from the incoming response received from another proxy located outside the network. And I need that because Kamailio strips the top most via (where the received parameter is present ) in the incoming response before sending it back to the local user, which I guess is correct. But the user needs to know that IP address. So we are planning to send that IP address in a custom header in the response back to the user. After going through some Kamailio documents I found that $sel(via.received) would return the IP address in top most via of a message. Can I store it in a variable inside onreply_route[MANAGE_REPLY] block and use that variable later in some appropriate route block to insert a new header in the response to the local user? Can I do something as given below to fetch and store the IP address from received parameter in the via header of incoming response onreply_route[MANAGE_REPLY] { ... if(search_hf("Via", "received", "f")) { xdbg("received param exists on top most via header \n"); $var(public_ip) = $sel(via.received); } ... } and insert a new header as given below in an appropriate route block which is executed while sending response back? append_hf("My-custom-hdr: $var(public_ip) \r\n"); I am not sure if this logic would work. Is the $var() variable retained through two different sip transactions? And which route block should be used to insert the header as mentioned above? Please help. Thanks, Ajay Hello, the received parameter is constructed from source IP of the incoming request. You get it via $si script variable. Cheers, Daniel On 3/12/13 7:22 PM, Ajay Sabat wrote: > > Hi, > > I want to extract the IP address in the received parameter of the top > most via header of a response and use that in a custom header while > sending response back to the caller. Can somebody please tell how to > implement that in the Kamailio configuration file? > > I need to handle a NAT traversal scenario without enabling NAT in > Kamailio which is used as the proxy server. The caller which is in the > same network as Kamailio first sends a OPTIONS request to the callee > before sending the INVITE. It uses the public IP address returned in > the custom header of the OPTIONS response to build the contact header > and SDP of the INVITE request. In fact, it first looks for the IP > address in the received parameter of the via header to do that. But > Kamailio removes the top most via with received parameter and then > sends the OPTIONS response back to the caller. So we are using a > custom header for that purpose. > > I would appreciate other suggestions to achieve this in Kamailio, may > be by enabling NAT or some other alternative. > > Thanks, > > Ajay > > > > _______________________________________________ > 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 Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio with PostgreSQL as persistence
by Hoai-Anh Ngo-Vi 13 Mar '13

13 Mar '13

Dears, I have tried out Kamailio 3.3.4 with PostgreSQL as persistence but it does not work. make FLAVOUR=kamailio include_modules="db_postgres" cfg was OK. I found /usr/local/lib/kamailio/modules/db_postgres.so after installation. I have successfully set up database openser using /usr/local/sbin/kamdbctl create with DBENGINE=PGSQL in /usr/local/etc/kamailio/kamctlrc. I have edited /usr/local/etc/kamailio/kamailio.cfg with: #!define WITH_PGSQL #!define WITH_AUTH #!define WITH_USRLOCDB ... #!ifdef WITH_PGSQL #!define DBURL "postgres://openser/openserrw@localhost:5432/openser" #!endif ... #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif Run /etc/init.d/kamailio start|debug I got Listening on .... kamailio error, failed to start. Does someone know some hints or have a working example? When I comment out the first 3 define kamailio starts fine. Thank you for your help. -- Ngô Vi Hoài Anh Biedenkopfer Weg 13 60489 Frankfurt am Main 069 74 22 36 63 0172 53 97 0 82

2 2

Re: [SR-Users] how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

Hi Daniel, Thanks for your response. But I don't want the IP in the received parameter of the incoming request. Instead, I want to read it (i.e. the public IP address of my own network where the local user and Kamailio are located) from the incoming response received from another proxy located outside the network. And I need that because Kamailio strips the top most via (where the received parameter is present ) in the incoming response before sending it back to the local user, which I guess is correct. But the user needs to know that IP address. So we are planning to send that IP address in a custom header in the response back to the user. After going through some Kamailio documents I found that $sel(via.received) would return the IP address in top most via of a message. Can I store it in a variable inside onreply_route[MANAGE_REPLY] block and use that variable later in some appropriate route block to insert a new header in the response to the local user? Can I do something as given below to fetch and store the IP address from received parameter in the via header of incoming response onreply_route[MANAGE_REPLY] { ... if(search_hf("Via", "received", "f")) { xdbg("received param exists on top most via header \n"); $var(public_ip) = $sel(via.received); } ... } and insert a new header as given below in an appropriate route block which is executed while sending response back? append_hf("My-custom-hdr: $var(public_ip) \r\n"); I am not sure if this logic would work. Is the $var() variable retained through two different sip transactions? And which route block should be used to insert the header as mentioned above? Please help. Thanks, Ajay Hello, the received parameter is constructed from source IP of the incoming request. You get it via $si script variable. Cheers, Daniel On 3/12/13 7:22 PM, Ajay Sabat wrote: > > Hi, > > I want to extract the IP address in the received parameter of the top > most via header of a response and use that in a custom header while > sending response back to the caller. Can somebody please tell how to > implement that in the Kamailio configuration file? > > I need to handle a NAT traversal scenario without enabling NAT in > Kamailio which is used as the proxy server. The caller which is in the > same network as Kamailio first sends a OPTIONS request to the callee > before sending the INVITE. It uses the public IP address returned in > the custom header of the OPTIONS response to build the contact header > and SDP of the INVITE request. In fact, it first looks for the IP > address in the received parameter of the via header to do that. But > Kamailio removes the top most via with received parameter and then > sends the OPTIONS response back to the caller. So we are using a > custom header for that purpose. > > I would appreciate other suggestions to achieve this in Kamailio, may > be by enabling NAT or some other alternative. > > Thanks, > > Ajay > > > > _______________________________________________ > 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 Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Need help understanding/separating signaling from media
by Grant Bagdasarian 13 Mar '13

13 Mar '13

Hello, I want to separate signaling and media in our architecture but I'm not entirely sure if the following will work (note that it is still in a design phase and these kinds of constructs are still new to me): User Agent 1-------------- Kamailio -------------- User Agent 2 | | | | | | -------------- Media Server Cluster -------------- The signaling will go through Kamailio but the media stream should pass through one of the media servers in the cluster for each call. Is it possible to achieve this by having Kamailio instruct the media server which ports to open and alter sdp information? I think that the RTP Proxy module does precisely this, but what I still don't understand is how the rtp stream is passed between these two agents for each call. Also, is RTP proxy the best way to do this or are there any other third party media servers which I could use? I don't need any fancy functionality, just basic media handling/pass-through. Regards, Grant

2 8

Send Bye to all calls in one profile
by Camila Troncoso 13 Mar '13

13 Mar '13

Hi, I have a kamailio proxy server that´s controlling the limit of calls that clients can make with profile grouping. In other side I also limit clients call duration base on users credit ( DATABASE) and *timeout *feature . Before the calls get answer I calculate how much time the client has left for this call and set the timeout variable. When the time is reached the proxy sends BYEs two both sides of this call. I want when the call timeout, as before explain, all other calls in the same profile end too, because the client has expired his credit. Is there a way to achieve this? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

4 6

java code to interact with kamailio
by Rupayan Dutta 13 Mar '13

13 Mar '13

Dear Sir, I want to authenticate users as well as getting buddy list from kamailio using java .Is there any existing library already in place. Rupayan Dutta

3 2

how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

Hi, I want to extract the IP address in the received parameter of the top most via header of a response and use that in a custom header while sending response back to the caller. Can somebody please tell how to implement that in the Kamailio configuration file? I need to handle a NAT traversal scenario without enabling NAT in Kamailio which is used as the proxy server. The caller which is in the same network as Kamailio first sends a OPTIONS request to the callee before sending the INVITE. It uses the public IP address returned in the custom header of the OPTIONS response to build the contact header and SDP of the INVITE request. In fact, it first looks for the IP address in the received parameter of the via header to do that. But Kamailio removes the top most via with received parameter and then sends the OPTIONS response back to the caller. So we are using a custom header for that purpose. I would appreciate other suggestions to achieve this in Kamailio, may be by enabling NAT or some other alternative. Thanks, Ajay

2 1

Malformed PUBLISH from pua_dialoginfo (4.0.0)
by Charles Chance 13 Mar '13

13 Mar '13

Hi all, We are testing 4.0.0 in the lab and since upgrading, PUBLISHes generated by pua_dialoginfo appear to be malformed (content-length mismatch). Please see attached output from Wireshark for both the PUBLISH and the 400 response from Kamailio. PUBLISH is being sent from one Kamailio to another - nothing has changed other than upgrading from 3.3.3 to 4.0.0. Has anyone else noticed the same? Regards, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

2 5

Failover on carrierroute
by Ali Jawad 13 Mar '13

13 Mar '13

Hi I did setup carrierroute with failover using t_on_failure, I do have 4 ".25 ratio" primary gateways for one carrier and 4 ".25 ratio"secondary gateways for a secondary carrier. However, given the current setup, if one gateway is down carrieroute does use the secondary carrier if one gateway of the primary carrier is down. What I need is to only go to the secondary carrier if all the gateways of the primary carrier are down. I thought of setting up 4 t_on_failure branches with carrierroute but that will be troublesome to maintain. Is there another method for this ? Regards

2 2

High availability
by Moacir Ferreira 12 Mar '13

12 Mar '13

Hi, What would be the community's architecture advice for deploying Kamailio in a high availability environment? I am looking for something simple, still efficient. Thanks,

3 3

Porting kamailio4 to OpenWrt
by Jiri Slachta 12 Mar '13

12 Mar '13

Hello everyone, I am new to this list and I am a newbie when it comes to Kamailio details, so sorry for any of my misunderstandings. I am the author of kamailio3 package in OpenWrt and currently I am trying to port recent major release of kamailio to OpenWrt. Let's jump directly to my question. The current state of my Kamailio4 package is that all modules which does not depend on any external libraries, are succesfully built. Any module that depends on external library is not built at all. But if I pass LD="$(TARGET_CC)" to the linker via: make -C/path/to/module LD="$(TARGET_CC)" then the module is succesfully compiled. I build kamailio4 via macro Build/Compile in following Makefile - http://liptel.vsb.cz/svn/besip/Trunk/packages-trunk/net/kamailio4/Makefile My question is - has anything changed in build procedures or variables significantly? It seems that LD variable is not passed into Makefiles of specific modules. Am I doing anything wrong? I am open to any suggestions. Thank you! ~ Jiri Slachta

2 1

Fwd: Re: memcached module problems
by Henning Westerholt 12 Mar '13

12 Mar '13

---------- Forwarded Message ---------- Subject: Re: memcached module problems Date: Dienstag, 12. März 2013, 14:50:41 From: Dragos Oancea <droancea(a)yahoo.com> To: Bis: Henning Westerholt <henning.westerholt(a)1und1.de> Thank you Henning, that was a good hint. I've noticed that kamailio and my perl script (which is using Cache::Memcached) set the value in memcached somehow differently. kamailio: set 888_HA1 1 10800 32. 6fa22552ea04d443ab5864a7b3c2168c. perl script: set 888_HA1 0 0 32. 6fa22552ea04d443ab5864a7b3c2168c. VAR_VAL_STR is 1. I will investigate further. Regards, Dragos ps: I will also subscribe to sr-users . From: Henning Westerholt <henning.westerholt(a)1und1.de> To: Dragos Oancea <droancea(a)yahoo.com> Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 12, 2013 9:35 AM Subject: RE: memcached module problems Hi Dragos, (setting sr-users to CC) Ok, If you’ve set this by other means you need to set the appropriate “Kamailio” type flag to the value as well, that Kamailio can parse it correctly. You can get the appropriate value if you grep for “VAR_VAL_STR” in the source code. Your perl script function call should also export the appropriate flag setting from the memcache library function call, then just add it here. Best regards, Henning From: Dragos Oancea [mailto:droancea@yahoo.com] Sent: Monday, March 11, 2013 8:24 PM To: Henning Westerholt Subject: Re: memcached and other things Hi Henning, Other means. a perl script. Dragos > 0(21130) DEBUG: memcached [mcd_var.c:87]: allocate new memcache request at > 0x7f747fb798f0 > 0(21130) DEBUG: memcached [mcd_var.c:93]: allocate new > memcache result at 0x7f747fb79a70 0(21130) > DEBUG: memcached [mcd_var.c:103]: result: 'just_A_text' for key 888_HA1 with > flag 0 > 0(21130) ERROR: memcached [mcd_var.c:146]: could not convert string > 'just_A_text' to integer value > 0(21130) DEBUG: memcached [mcd_var.c:159]: free memcache request and result > at 0x7f747fb798f0 The issue is that the module somehow does not evaluate the string variable _from memcached_ as a string, it tries to evaluate it as integer. This obviously fails. Has this value set from kamailio or is this from some external application? Then probably this application does not set the proper flags which kamailio expects. Regards, Henning Westerholt -------------------------------------------------------------

1 0

File kamailio_fifo is removed...
by Bruno Bresciani 12 Mar '13

12 Mar '13

Hi, Kamailio is running fine, but when I try run again by command line without killing the previous, the kamailio_fifo is removed and functions fifo in kamctl doen't work anymore. Why when I try running kamailio when already exist one, it remove de file kamailio_fifo? I can't find the part of source code that file kamailio_fifo is removed... Someone knows? Best Regards

2 2

Kamailio MySQL DB Creation
by Amr Elsharqawi 12 Mar '13

12 Mar '13

Hello Kamailio Community!! I am newbie to Kamailio & SIP. And I am interested in Kamailio. I've used asterisk for some time then I heard about FreeSWITCH unfortunately FreeSWITCH was not good enough for me like Asterisk. I've found NAT problem with Asterisk Finally I've decided to work with Kamailio as SIP server & Asterisk as PBX. to be fair I need to learn more about FreeSWICH to test it well. My Question is about Kamailio installation; I am using FreeBSD 9 attached my uname -a FreeBSD PBX.XXXXX 9.1-RELEASE FreeBSD 9.1-RELEASE #0 r243826: Tue Dec 4 06:55:39 UTC 2012 root@obrian.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC i386 I've used gmake for building, compiling and installation. Everything goes smooth EXCEPT mysql db installation when I use sh /usr/local/sbin/kamdbctl create I am getting the following error: /usr/local/lib/kamailio//kamctl/kamdbctl.base: 36: Syntax error: Missing '}' Thanks a lot for your valuable time and that amazing project (Kamailio). Amr

3 6

Strange TLS issue
by Jirka Jirout 12 Mar '13

12 Mar '13

Hello, I have a very strange issue - we are running kamailio 3.3.2 as a part of Sip Provider CE and some of our clients have issues with TLS connectivity. 1. some clients such as Jitsi on Mac OS X and Bria or idoubs on iOS can always connect 2. imsdroid on Android can sometimes connect and works (about 10 % of attempts) When I run tcpudmp on the server during the unsuccessfull attempts, I see something like this: > 1 1 0.0347 (0.0347) C>S SSLv2 compatible client hello > Version 3.1 > cipher suites > TLS_RSA_WITH_AES_128_CBC_SHA > 1 2 0.0360 (0.0013) S>C Handshake > ServerHello > Version 3.1 > session_id[0]= > cipherSuite TLS_RSA_WITH_AES_128_CBC_SHA > compressionMethod NULL > 1 3 0.0361 (0.0001) S>C Handshake > Certificate > 1 4 0.0361 (0.0000) S>C Handshake ServerHelloDone > 1 5 0.0782 (0.0420) C>S Alert > level fatal > value unexpected_message > 1 0.0785 (0.0002) S>C TCP FIN > 1 6 0.1092 (0.0307) C>S Handshake > ClientKeyExchange > 1 7 0.1092 (0.0000) C>S ChangeCipherSpec > 1 8 0.1092 (0.0000) C>S Handshake What I find particularly strange is that the client sends fatal alert and then attempts to continue with handshake. What exactly does this "unexpected_message" mean? Is the client letting the server know the last message was unexpected? Or is this telling me the server had not expected the message? It cannot be a certificate problem, can it? Considering other clients were fine, I blamed our code at first, but I am no longer sure that is the (only) reason for this behavior. I decided to start from scratch and wrote an extremely simple application in plain C, that connects to the TLS port using openssl (which imsdroid also uses), sends some query and then prints response. It works fine on https sites, even on the same server kamailio is running and with the same certificates (our certificates come from Verisign CA G3). When I run ssldump, I can see handshake is fine, socket opened, the client is sending application data. I can see data coming in on the server using tcpdump. However, the data never reaches Kamailio - no sign of the packets in the log and no answer whatsoever (I use a fake REGISTER message so I would expect to get something like "Unauthorized" back). The same happens, when I try to use openssl s_client to connect to the site and send "register" message (because R at the start of packet triggers renegotiation in s_client, uff). My TLS settings in Kamailio are pretty much default, only using my own key and certificate. regards, jjj

4 5

Re: [SR-Users] memcached module problems
by Henning Westerholt 12 Mar '13

12 Mar '13

Hi Dragos, (setting sr-users to CC) Ok, If you’ve set this by other means you need to set the appropriate “Kamailio” type flag to the value as well, that Kamailio can parse it correctly. You can get the appropriate value if you grep for “VAR_VAL_STR” in the source code. Your perl script function call should also export the appropriate flag setting from the memcache library function call, then just add it here. Best regards, Henning From: Dragos Oancea [mailto:droancea@yahoo.com] Sent: Monday, March 11, 2013 8:24 PM To: Henning Westerholt Subject: Re: memcached and other things Hi Henning, Other means. a perl script. Dragos > 0(21130) DEBUG: memcached [mcd_var.c:87]: allocate new memcache request at > 0x7f747fb798f0 > 0(21130) DEBUG: memcached [mcd_var.c:93]: allocate new > memcache result at 0x7f747fb79a70 0(21130) > DEBUG: memcached [mcd_var.c:103]: result: 'just_A_text' for key 888_HA1 with > flag 0 > 0(21130) ERROR: memcached [mcd_var.c:146]: could not convert string > 'just_A_text' to integer value > 0(21130) DEBUG: memcached [mcd_var.c:159]: free memcache request and result > at 0x7f747fb798f0 The issue is that the module somehow does not evaluate the string variable _from memcached_ as a string, it tries to evaluate it as integer. This obviously fails. Has this value set from kamailio or is this from some external application? Then probably this application does not set the proper flags which kamailio expects. Regards, Henning Westerholt

1 0

RPMs for v4.0.0
by Daniel-Constantin Mierla 12 Mar '13

12 Mar '13

Many RPM packages are are now available on OpenSUSE build server (for Centos 5.5 and 6, RedHat 5 and 6, Fedora 16, 17 and 18, OpenSuse 11.4, 12.1, 12.2 and 12.3, on i586 and x86_64 architectures). Links to repositories are at: - http://www.kamailio.org/wiki/packages/rpms There is listed the link to Peter's RPM repository as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Routing calls
by mark＠brightvoip.co.uk 12 Mar '13

12 Mar '13

Hi, I'm looking for pointers on the following scenario. I have two sites (A and B) and sip traffic is sent from A to B. At each site there are a number of SIP (Asterisk) servers and a Kamailio server. The two sites have various network routes (X, Y & Z) between them. I have been asked to load balance the inter-site SIP traffic across the available links with a 20/20/60 traffic split (dispatcher, carrierroute). Each Kamailio server has multiple IP addresses and these are used to direct traffic to specific routes (Kamailio at site A IP 111.111.111.1 to Kamailio site B 111.111.111.2 defines route X for example). For any given call, the complete call (SIP and both RTP streams) should travel wholly over one route. I would welcome any suggestions about how best to route the whole call (sip & rtp) over a specific route, I was working on a solution with multiple RTPProxy instances, but was wondering if this is the best solution. Thanks in advance. Mark

1 0

Kamailio v4.0.0 Released - new major version is out
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Kamailio v4.0.0 is out – a special major release, shifting up first number of versioning string! It is the landmark for the end of Kamailio-SER integration, bringing out as well a very large set of new features and improvements, like websocket support or dozens of IMS extensions. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v4-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.0.0! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

releasing v4.0.0 tomorrow
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Hello, the plan is to release final Kamailio v4.0.0 tomorrow, Monday, March 11, 2013. If any developer/packager has something to add to the release, please commit by 12:00GMT to GIT branch 4.0. After that, announce first on sr-dev and wait for an ack, to avoid packaging inconsistent content. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 1

kamailio is returning 500
by chanea 11 Mar '13

11 Mar '13

Dear all While lots of request come to my Kamailio 3.1.x in the same time , server is returning "500, Server Internal Error" to some of the request. What should i do if i want to reduce the number of "500" response? chanea

3 7

SIP and XMPP server integration
by Vinay V 11 Mar '13

11 Mar '13

Hi All, I am in process of integrating a sip server (Lync) with XMPP server (kamailio). Can a kamailio server be used as a sip to xmpp gateway and vice versa. If i am using any other xmpp server and want to integrate with lync will kamailio can be used as a gateway. This integration is maily for IM and presence. If it can be done what configuration should be done to kamailio.conf file. Please provide the inputs. I am very much need of it. If anybody has done already the above testing please provide DOCS. Thanks in advance, Vinay

2 2

Problem with frequent use of FIFO commands
by Mino Haluz 11 Mar '13

11 Mar '13

Hi, I have some nagios scripts binded to kamctl fifo commands, as well as some web scripts that are directly calling kamctl fifo statements. I do not know why, but it sometimes returns ** ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo [1] => ** ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config [2] => ** ERROR: and also have loaded the mi_fifo module. when it is called with web script with www-data account. From shell it works everytime, but with web 20% of results are errors. Other 80% are sucessful, so no problem with permissions. Does mi_fifo support multiple access, are there any other limits in this regard ? Within OS level? kamailio-3.3.4 Thanks, Mino

2 1

Kamailio 4.0 install guide
by Jesús Pérez Rubio 11 Mar '13

11 Mar '13

Hi, I think there is a little error on this guide about 4.0 version install: http://www.kamailio.org/wiki/install/4.0.x/git First it is said this: "8. Edit configuration file To fit your requirements for the VoIP platform, you have to edit the configuration file. /usr/local/etc/kamailio/kamailio.cfg" At the end it is said that only this line of init.d script have to be changed: DAEMON=/usr/local/sbin/kamailio I would add this one too: CFGFILE=/usr/local/etc/kamailio/kamailio.cfg Another option would be to work directly with "/etc/kamailio/kamailio.cfg" Anyway, I take this oportunity to thank you for your great job and to notice that this version 4.0 seems to work stil better with our QoffeSIP Javascript SIP stack. :) -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com

2 1

no destination sets in dispatcher module
by Ali Jawad 11 Mar '13

11 Mar '13

Hi I am using carrieroute to distribute traffic, I am not sure if there is a way for carrieroute to detect if a destination is down or not, but I read that the dispatcher module does that so I did compile it into the existing kamailio installation and added the following settings : loadmodule "dispatcher.so" modparam("dispatcher","db_url", "mysql://connectionstring") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "force_dst", 1) and under ROUTE if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; xlog("L_ERR","************************AFTER TOO MANY HOPS******************************************* cr\n"); ds_select_dst("1", "0"); forward(); # t_relay(); All I get is too many hops in ngrep and kamailio reports : (26467) ERROR: dispatcher [dispatch.c:1564]: no destination sets In the DB I do have a setid 1 and sip:ip.of.carrer:5060 as destination. Regards

2 1

why use received as RURI for nathelper OPTIONS ping?
by Andrew Pogrebennyk 11 Mar '13

11 Mar '13

Some Fritzbox devices don't accept the OPTIONS ping generated by kamailio with RURI: sip:11.22.3.4:5060. In the location table we have: received: sip:11.22.3.4:5060 contact: sip:user@11.22.3.4:5060;uniq=6633BC1386F4D4CC4EBD64DC7E967 path: <sip:lb@127.0.0.1;lr;received='sip:11.22.3.4:5060'> Kamailio version is 3.3.2 and the nathelper config is nothing fancy: modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sipwise.local") modparam("nathelper", "received_avp", "$avp(s:received)") What I don't understand is why kamailio sets RURI of the OPTIONS message to value of "received" instead of the contact. I suspect a bug in the parser somewhere along these lines: > rval = ul.get_all_ucontacts(buf,cblen,(ping_nated_only?ul.nat_flag:0), > ((unsigned int)(unsigned long)timer_idx)*natping_interval+iteration, > natping_processes*natping_interval); > if (rval != 0) { > pkg_free(buf); > goto done; > } > } > > if (buf == NULL) > goto done; > > cp = buf; > while (1) { > memcpy(&(c.len), cp, sizeof(c.len)); > if (c.len == 0) > break; > c.s = (char*)cp + sizeof(c.len); > cp = (char*)cp + sizeof(c.len) + c.len; > memcpy( &send_sock, cp, sizeof(send_sock)); > cp = (char*)cp + sizeof(send_sock); > memcpy( &flags, cp, sizeof(flags)); > cp = (char*)cp + sizeof(flags); > memcpy( &(path.len), cp, sizeof(path.len)); > path.s = path.len ? ((char*)cp + sizeof(path.len)) : NULL ; > cp = (char*)cp + sizeof(path.len) + path.len; > > /* determin the destination */ > if ( path.len && (flags&sipping_flag)!=0 ) { > /* send to first URI in path */ > if (get_path_dst_uri( &path, &opt) < 0) { > LM_ERR("failed to get dst_uri for Path\n"); > continue; > } > /* send to the contact/received */ > if (parse_uri(opt.s, opt.len, &curi) < 0) { > LM_ERR("can't parse contact dst_uri\n"); > continue; > } > } else {

4 5

scheduling inside SER
by irma zujovic 11 Mar '13

11 Mar '13

Hello, I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. My question is : where inside SER configuration scripts, the *scheduling *and *queueing *of SIP messages is done? I am interested in a special function or part of the written program. I hope that engineers that worked on developing of SER could help me with this issue. Thank you in advance Irma

3 2

INVITE messages not authenticated (default configuration)?
by Paul Belanger 08 Mar '13

08 Mar '13

Greeting, Hopefully, I'm understanding the following default kamailio.cfg[1] file. Over the weekend, I was attached by SipVicious. Following along with the example Daniel[2] create with kamailio and asterisk, I have almost the same setup. Rather then storing my SIP profiles in Asterisk database, I have then in Kamailio. To my point, the attacker was actually able to by pass any sort of authentication, but simply sending an INIVTE message: ./svmap.py -e 18885551234 kamailio.example.org -m INVITE Which kamailio, forwarded to Asterisk and because there is no additional auth within asterisk, was able to hit the asterisk context for getting processed (they did not get out to the real world). However, my question is.... why do we not authenticate INVITE messages? If my understanding is correct, if would require something like the following: if (is_method("INVITE")) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } } If so, why not also do it in the default configuration file? [1] http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=et… [2] http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

6 8

Update existing module or create new?
by Charles Chance 08 Mar '13

08 Mar '13

All, We are working on a memcached module which uses the libmemcached library, as opposed to the (no longer under active development) libmemcache that the existing module depends on. Am I correct in thinking that in this instance, it would be better to create a brand new module (with a different name) rather than modify the existing one? In addition to changing the dependency, we will also be exporting new commands and potentially changing the format of the exported PVs. Regards, Charles -- www.sipcentric.com <http://www.sipcentric.com/> Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

4 8

Asterisk realtime with kamailio Load balancing issue for sip user
by Prakash N 07 Mar '13

07 Mar '13

Hi All, We have finished the Kamailio & Asterisk real time integration and load balancing also done using dispatcher module. Queue and voice mails are load balancing as well.When we are calling extension to extension it is showing in all the servers.It seems extension are not load balancing as per our knowledge. I have attached the kamailio.cfg for your reference,Find my coding below as mentioned. *# -- dispatcher params for DB support --* *modparam("dispatcher","db_url", "mysql:// openser:openserrw@192.168.1.170/openser")* *modparam("dispatcher", "table_name", "dispatcher")* *modparam("dispatcher", "setid_col", "setid")* *modparam("dispatcher", "destination_col", "destination")* *modparam("dispatcher", "flags_col", "flags")* *modparam("dispatcher", "priority_col", "priority")* * * * ----------------------------------------------------------------------------------------- * *# Dispatch requests* *route[DISPATCH] {* *if ( method=="INVITE" ) {* *# dst_select( "GROUP", "HASH METHOD")* * ds_select_dst("1","4");* * sl_send_reply("100","Trying");* * forward();#uri:host, uri:port);* * exit();* *}}* Kindly suggest the solution for the same. Thanks in advance. Regards, N.Prakash

3 10

Re: [SR-Users] scheduling inside SER
by Jiri Kuthan 07 Mar '13

07 Mar '13

not sure if there is a point in doing that, at least for incoming traffic since it already happens in kernel, unless you want to do it in kernel. jiri On 3/7/13 2:25 PM, irma zujovic wrote: > Thank you for your answer. > > So is there any way to produce some kind of algorithm for prioritization of SIP messages inisde FOKUS IMS Core platform by configuring SER? That was my first > goal, when asking the previous question. I tought that scheduling and buffering of messages inside FOKUS IMS is done inside SER, and tought about priorituzation > inside SER. > > If you can help me in any way i would be very greatful. > > Thank you in advance > > Irma > > On Tue, Mar 5, 2013 at 4:13 PM, Jiri Kuthan <jiri(a)iptel.org <mailto:jiri@iptel.org>> wrote: > > On 3/5/13 2:36 PM, irma zujovic wrote: > > Hello, > > I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. > > My question is : where inside SER configuration scripts, the *scheduling * and *queueing *of SIP messages is done? I am interested in a special function > or part > of the written program. > > > this is not done in SER, it is done in kernel. kernel dispatches incoming traffic to > one of multiple processes listening on sockets. Same for outgoing traffic -- SER leaves > it to kernel's decision. > > jiri > > > I hope that engineers that worked on developing of SER could help me with this issue. > > Thank you in advance > > Irma > > > _________________________________________________ > 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 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > >

1 0

Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 07 Mar '13

07 Mar '13

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

2 1

installing v4.0.x from git
by Daniel-Constantin Mierla 07 Mar '13

07 Mar '13

Hello, I created the wiki page with the how-to install kamailio v4.0.x from git. The tutorial is available at: - http://www.kamailio.org/wiki/install/4.0.x/git It should make life easier for those willing to give a try to v4.0.0 these days. Anyhow, a review of the content is very appreciated. Feel free to enhance the content by adding to the wiki page or reply here with your ideas to improve. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

4 7

Fwd: kamailio rls mod
by Aleksandrs Semenenko 06 Mar '13

06 Mar '13

Hello, I'm looking for assistance in RLS module setup for kamailio 3.3.2 The problem is that I need to handle resource-list updates on my kamalio server. For testing purposes I use Jitsi client. Expected: 1) When a new contact is added to Jitsi or an old one is removed, Jitsi sends updated resource-list XML to caps. - This works fine, I can see PUT request in Wireshark and I can see following log from kamailio.cfg: xlog("===== xhttp put: refreshing resource-list for $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); which hopefully means that rls_update_subs function has been called. 2) After this function was called, I expect SUBSCRIBE messages in backend, not in Jitsi, but I still see Jitsi sending SUBSCRIBE for each contact. What am I doing wrong? Any help would be useful! Regards, Alex.

4 3

New developer: Hugh Waite
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Hugh Waite - he is quite old contributor by now, with many patches and activity on mailing lists. He is part of the Crocodile RCS team, which developed very important features for Kamailio so far, like websocket and outbound extensions, Lua API and presence/rls improvements. His git commit id is: hpw My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

early rpm and deb packages from branch 4.0
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, if anyone is preferring to use packages for pre-release testing of v4.0.0, I created rpms for centos, redhat, opensuse and fedors, plus packages for debian squeeze. You can find them on opensuse build service servers: - https://build.opensuse.org/package/show?package=kamailio40&project=home%3Ak… - https://build.opensuse.org/package/show?package=kamailio40-d6.0&project=hom… I made them mainly to be sure there are no issue compiling on common linux OSes. Debian packages should be available at the usual repository for more versions (including ubuntu), courtesy of Jon Bonilla. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio Geoip
by Ian French 06 Mar '13

06 Mar '13

Hey List, I'm looking at an article by Daniel, http://by-miconda.blogspot.ie/2010/10/best-of-new-in-kamailio-310-5-geoip-a…. In it he's uses the below to block source IP addresses by country, How could this be used to block multiple country sources?? I've tried a couple of things aleardy but have failed geoip_match("$si", "src"); if($gip(src=>cc)=="CA") { xlog("SIP message from Canada (ip: $si) - drop it\n"); send_reply("403", "Forbidden"); exit; } Thanks in advance Ian

3 4

Kamailio World Updates - March 06
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, the schedule for Kamailio World Conference has been updated, among topics of the recently added talks: North American VoIP market, Jitsi softphone, usage in embedded systems and enterprises. Check the current schedule at: - http://conference.kamailio.com/k01/schedule/ The group of speakers for the two days is filled up at this moment, as soon as we received the profiles and presentation summaries, the schedule will get final. Regarding Kamailio project, these are the last days before the next major release, version 4.0.0 is planned to be out next week. This week is CeBIT in Hanover, many community members will visit the event in various days (to my knowledge, Thursday and Friday will be the busiest). If you want face to face meetings, announce yourself on our forums. I plan to be there tomorrow. It is just a bit over one month till the event starts, if you haven't registered yet, you should hurry up, there is no similar event in Europe this year that gives the chance to meet and learn from many professionals working with rich communication services and products: - http://conference.kamailio.com/k01/registration/ Also, if you consider sponsoring or participate to surrounding exhibition space, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ Do not forget that there are open positions to work with VoIP/SIP and Kamailio, just in case you look for new challenges in your career, check: - http://conference.kamailio.com/k01/career-jobs/ Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Simple look table
by Gareth Rylance 05 Mar '13

05 Mar '13

Is it possible to have a custom look-up table to map ids to other ids in the configuration file? E.G. TestLookValues key | value 1 |100 2 | 100 3 | 600 lookup("TestLookValues", "3") gives $var(lookupval) equal to 600 It is similar to MTree functionality only I am looking for an exact match not just a prefix TestLookValues prefix | value 1 |100 2 | 100 3 | 600 mt_match("TestLookValues", "3003)", "0") gives $var(mtval) = 600 I would need $var(mtval) = -1 (not found) Thanks Gareth

2 1

Troubles with call between 2 users
by TF Julian Plata 05 Mar '13

05 Mar '13

Hello, I'm Julian Plata and I have some troubles to do a call between 2 users: julianplata(a)iptel.org since Media5 in my phone, and marinastereo(a)iptel.org in a Comrex box. When I try to connect, Media5 shows a message "The caller can not be found". Also, in the Iptel User Management the users julianplata@iptel & marinastereo@iptel are shown as online. Could you help me to fix the issue ? Could you verify if the accounts are OK for establish a connection?? Thanks, Best Regards, Julian Plata -- ------------------------------------------------------------------------------------ La informacion contenida en este E-mail es confidencial y solo puede ser utilizada por el individuo o la compania a la cual esta dirigido. Si no es el receptor autorizado, cualquier retencion, difusion, distribucion o copia de este mensaje es prohibida y sera sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo al remitente y borrar el mensaje recibido inmediatamente. Cualquier punto de vista expresado en este mensaje es unicamente de quien lo envia excepto donde especificamente se indique que es la posicion de la Armada Nacional de Colombia (ARC). ------------------------------------------------------------------------------------ Apaga la pantalla de su computador cuando salgas a almorzar. No malgastemos la Energía. Cuidemos lo que es de Todos. ------------------------------------------------------------------------------------ Antes de imprimir este e-mail piensa bien si es necesario hacerlo: 1 hoja de papel tamaño carta = 2 litros de agua + 10 gramos de materia prima (decolorantes, fijadores, pigmentos) + 1 bombillo de 40 Vatios prendido por 1 hora + 14 gramos de madera pura.

2 1

Kamailio problem with both clients in the same NAT
by mohavere36 04 Mar '13

04 Mar '13

4 4

releasing v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, from my point of view, the last to sort out for v4.0.0 is packaging (I just sent a dedicated email to decide on that). Personally I have been testing v4.0.0 thoroughly in various cases, no critical issue being on my list at this moment. Also, v4.0 is running for quite some time on voipuser.org (our usual life testing platform). I propose releasing v4.0.0 next Monday, on March 11, 2013. The second option, if more people prefer it, would be Thursday, March 14, 2013. Opinions? If anybody has other dates in mine, let us know here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

packaging v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, (cross-posting as this decision may be done properly with feedback from both developers and users community). Now that we have some band new modules but also several that were not packaged by Kamailio, just by ser (due to single modules directory), I was thinking we may want to reconsider a bit the packages we do for v4.0.0. 1) A new package is related to the bunch of IMS modules, that is pretty much obvious. 2) There are few modules targeting development samples or troubleshooting: print, print_lib and malloc_test - I think they should not be packaged. 3) I would create a special package for the modules using uid-based database schema: db2_ops uid_auth_db uid_avp_db uid_domain uid_gflags uid_uri_db uri_db 4) Maybe we should merge some packages by dependencies: - a) some depend on libxml2: cpl-c xhttp_pi xmlrpc xmlops - b) some depend on libssl: tls auth_identity outbound 5) Some experimental modules with external deps will stay unpackaged like so far: db_oracle Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

When to rewrite o= in SDP ?
by Khoa Pham 04 Mar '13

04 Mar '13

Hi, I read in http://kamailio.org/docs/modules/1.4.x/nathelper.html that Kamailio might rewrite ip address in o= in SDP to facilitate NAT traversal. It may decide to use to original IP or use IP of RTP server. What factors affect its decision ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

UnixODBC cant insert sip_capture record on sql server
by Grant Bagdasarian 03 Mar '13

03 Mar '13

Hello, We are using SQL Server 2008 as our database. When the sip_capture module tries to insert a row, the following error is given: db_unixodbc [con.c:220]: unixodbc:SQLExecDirect=42000:1:156:[FreeTDS][SQL Server]Incorrect syntax near the keyword 'authorization'. Using Kamailio 3.3.4. Is there a way to force brackets around the column values? Regards, Grant

2 3

uac_replace_from / New URI shorter than old URI
by Ronald Voermans 03 Mar '13

03 Mar '13

Hi, this last weekend we went live from the old 3.0 version to the latest stable release of Kamailio (3,3,0). All went well, except for one issue. We make use of uac_replace_from in our config, in case a client needs CLIR: uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); Since the upgrade we receive the following error: uac [replace.c:521]: new URI shorter than old URI Which causes the call to be disconnected after about 20/30 seconds. This, because the ACK is received by Kamailio, but cannot be 'linked' to the transaction i guess. What is exactly causing this issue and, more importantly, is there any way to resolve this? Thanks in advance, Regards, Ronald Voermans

2 4

Which is first, t_on_reply or callbacks in C ?
by David 03 Mar '13

03 Mar '13

Hello, Can someone tell me which is executed first ? the on reply route setup with t_on_reply() or the callbacks in the code that allows the module dialog to insert the confirmed dialog into the dialog table ? Thanks, David

2 1

not able to install Kamailio
by Julián Quintero 03 Mar '13

03 Mar '13

Hello thereI hope that you can help me, I have an issue installing Kamailio, when I issue th command "make all" I receive the following error resolve.h:471:4: error: #error neither gethostbyname2 or getipnodebyname present make: *** [action.o] Error 1 I am using debian squeeze, please let me know if there is something missing in my system or what is not allowing to install the SIP Server THanks Julián

3 3

t_lookup.c: WARNING: script writer didn't release transaction - problem
by Efelin Novak 03 Mar '13

03 Mar '13

Hi folks, we have a serious issue with tm module. In Fail route we would like to send a reply 404 when a cr_route function fails(no destination found). We use t_reply("404", "No route to destination"). However after the call is disconnected with a 404 reply the tm module writes a following warning WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction Very soon after this error message is shown a segfault occurs. Log is as follows: Mar 3 15:26:46 hostname kamailio[791]: ERROR: carrierroute [cr_func.c:594]: desired routing domain doesn't exist, prefix 0039333333333, carrier 14, domain 2 Mar 3 15:26:46 hostname kamailio[791]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction //ACK is received Mar 3 15:26:52 hostname kernel: [771134.093250] kamailio[796]: segfault at 4 ip 0815c182 sp bf8ace00 error 6 in kamailio[8048000+1ea000] Mar 3 15:26:53 hostname kamailio[803]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:785]: child process 796 exited by a signal 11 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:788]: core was generated Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Mar 3 15:26:53 hostname kamailio[802]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[799]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[801]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[798]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[800]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[797]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[792]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[803]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[795]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[791]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[794]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[793]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x886b3ad8/0x886b3ae0) - ignore Mar 3 15:26:53 hostname kernel: [771135.681720] kamailio[790]: segfault at 7dd193e0 ip 0816f5cd sp bf8acc20 error 4 in kamailio[8048000+1ea000] Currently we have fixed it using a custom route route(T_REPLY); . . . route[T_REPLY] { sl_send_reply("404","No route to destination"); } This works correctly with no error or a warning message or a segfault. We are using kamilio version: kamailio 3.3.4 (i386/linux) f8c8f2 Is this behavior a bug or should I dig deeper into the tm documentation? Kind regards Efelin

2 1

Karl Smith
by Karl Smith 01 Mar '13

01 Mar '13

witwcvrf ictniydtoz.vsd/http://www.loudounwalkin.com/sq/f7nax0z0n2ouq77adgkgejjvnz.g… yklxx.vsd/knc hj.vsd/

1 0

RPM packages for v3.3.4
by Daniel-Constantin Mierla 01 Mar '13

01 Mar '13

Hello, the RPM packages for latest minor release v3.3.4 were created on openSuse build service. More details and download links at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Karl Smith
by Karl Smith 01 Mar '13

01 Mar '13

http://www.louiselemelin.com/sinkzm/hhir5ew9drwu4s54d429jdnj.d?a6dfinu09fqc… Smith3/1/2013 3:00:01 PM

1 0

How to let Kamailio to always use RTP proxy
by Khoa Pham 01 Mar '13

01 Mar '13

Hi, Supposed A call B in a non-NAT environment. How to let Kamailio to always force A, B to use RTP proxy ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

4 4

Help with Asterisk RT integration
by Jon Morby 28 Feb '13

28 Feb '13

Hi I'm trying to integrate a (K) front end cluster with an Asterisk back end cluster and Asterisk RT (legacy system) I've followed the recipe at http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb (with one minor exception …) if (from_uri!=myself && uri!=myself) became if (uri!=myself) as with the original line in place we were able to spoof traffic from 3rd party sites and route out onto the PSTN (which I thought was bad) … anyway … The problem I'm seeing currently is that when a call is passed down a SIP trunk to an end user on the (K) platform we're losing the DNID Asterisk delivers the call to SIP/account/DNID (K) however just tries to deliver the call to DNID@domain which comes back as not found. I've had limited success hand coding aliases into the alias_db however we're still missing the DNID info and having to scrape it from the SIP To field (with a) limited success and b) fears of a support nightmare if we try and move existing customers onto the new platform). If we were to do this for real I'd either have to modify usrloc or try some perl_exec magic I'm guessing … (although it might be possible with a view on the existing sql database … but I'd rather not have to do that if there is a simpler way) I confess I'm struggling to get my head around the config and docs and hoped someone could point me in the right direction? For legacy reasons Asterisk needs to be in the critical path on this particular build … what I'm looking for is a simple recipe and some helpful pointers on how to implement it that will allow enable me to swing (K) into the path between our end user SIP devices and the existing asterisk back ends without losing the ability to deliver hundreds of numbers down a single SIP trunk to a subscriber, and that doesn't require them to make any changes on their end as they will still see the equivalent of SIP:${DNID}@example.com arriving on their PBX This should be simple, but I'm obviously missing something :) Help and pointers gratefully received -- Jon Morby FidoNet - the internet made simple! 10 - 16 Tiller Road, London, E14 8PX tel: 0845 004 3050 / fax: 0845 004 3051

3 4

Re: [SR-Users] SBC
by Keith 28 Feb '13

28 Feb '13

Hi All, So following a guide I have successfully setup Kamailio and FreeSwitch together working as an SBC. SIP registration with Kamailio and using FreeSwitch as the media server which in turn authenticates with users in Kamailio. I have managed to pass calls between users etc. So the next challenge is bolting 2 Asterisks servers behind this SBC and load balancing the media. As the SBC will be public facing I also have to consider NAT which I believe will be configured in FreeSwitch. I have had some fantastic advice in the past so just interested to get peoples views again. Thanks. -- Keith

1 0

Re: [SR-Users] [sr-dev] openser db on an external mysql server
by Prakash N 28 Feb '13

28 Feb '13

Hi Govoiper Free Memory on my server 2309 MB total used free shared buffers cached Mem: 2795 486 2309 0 70 180 -/+ buffers/cache: 234 2561 Swap: 2927 0 2927 *Kamailio error:* * * ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed *tail -f /var/log/messages:* Feb 12 23:41:28 localhost /usr/local/sbin/kamailio[4280]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: usrloc [ul_mod.c:407]: child(1): failed to connect to database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module usrloc (/usr/local/lib64/kamailio/modules_k/usrloc.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: auth_db [authdb_mod.c:174]: unable to connect to the database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [pt.c:350]: ERROR: fork_process(): init_child failed for process 3, pid 4283, "udp receiver child=2 sock=127.0.0.1:5060" Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: : <core> [main.c:1625]: main_loop: Cannot fork With Regards N.Prakash ---------- Forwarded message ---------- > From: SamyGo <govoiper(a)gmail.com> > Date: Tue, Feb 12, 2013 at 5:49 PM > Subject: Re: [SR-Users] [sr-dev] openser db on an external mysql server > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: Muhammad Shahzad <shaheryarkh(a)gmail.com>, sr-dev(a)lists.sip-router.org, > Prabhakaran B <prabhakaran(a)tevatel.com>, Indu Priya <indu(a)tevatel.com> > > > Hi Prakash, > > Please paste All ERROR appearing and Warnings when you start your > kamailio. What is the free memory situation on your server ? > > Regards, > Sammy > > > > > On Tue, Feb 12, 2013 at 5:12 PM, Prakash N <prakash.n(a)tevatel.com> wrote: > >> Hi Muhammad, >> >> Yes, I have define Mysql on top in Kamailio.cfg >> >> DBURL "mysql://openser:openserrw@192.168.1.170/openser" >> >> >> I have attached Kamailio.cfg ,please advice >> >> With Regards >> >> N.Prakash >> >> >> On Tue, Feb 12, 2013 at 5:37 PM, Muhammad Shahzad <shaheryarkh(a)gmail.com>wrote: >> >>> What is your connection string in kamailio i.e. DBURL? Also make sure >>> you have defined WITH_MYSQL on the top of kamailio.cfg to load necessary >>> drivers. >>> >>> #!define WITH_MYSQL >>> >>> Thank you. >>> >>> >>> On Tue, Feb 12, 2013 at 12:56 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>> >>>> Hi Shahzad , >>>> >>>> It is not ,We can able to connect remote mysql ( please find >>>> the detail below),moreover we can able create openser db from Kamailio >>>> server to remote mysql server >>>> >>>> *Mysql server : 192.168.1.170* >>>> *Kamailio server : 192.168.1.101* >>>> >>>> >>>> *Mysql output :* >>>> * >>>> * >>>> *mysql -u openser -h 192.168.1.170 -p* >>>> Enter password: >>>> Welcome to the MySQL monitor. Commands end with ; or \g. >>>> Your MySQL connection id is 68 >>>> Server version: 5.1.67 Source distribution >>>> >>>> Copyright (c) 2000, 2012, Oracle and/or its affiliates. All rights >>>> reserved. >>>> >>>> Oracle is a registered trademark of Oracle Corporation and/or its >>>> affiliates. Other names may be trademarks of their respective >>>> owners. >>>> >>>> Type 'help;' or '\h' for help. Type '\c' to clear the current input >>>> statement. >>>> >>>> mysql> >>>> >>>> With Regards >>>> >>>> N.Prakash >>>> >>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Muhammad Shahzad <shaheryarkh(a)gmail.com> >>>>> Date: Tue, Feb 12, 2013 at 5:16 PM >>>>> Subject: Re: [sr-dev] [SR-Users] openser db on an external mysql server >>>>> To: Development mailing list of the sip-router project < >>>>> sr-dev(a)lists.sip-router.org> >>>>> Cc: Daniel-Constantin Mierla <miconda(a)gmail.com>, >>>>> sr-users(a)lists.sip-router.org, Indu Priya <indu(a)tevatel.com>, >>>>> Prabhakaran B <prabhakaran(a)tevatel.com> >>>>> >>>>> >>>>> Its not a Kamailio error, it is problem with connecting to mysql >>>>> server. Error 4 means No MySQL Server Running On Specified Host. >>>>> >>>>> May be MySQL server is down or there is some firewall running or MySQL >>>>> not listening on IP you specified. >>>>> >>>>> Thank you. >>>>> >>>>> >>>>> On Tue, Feb 12, 2013 at 12:11 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>>>> >>>>>> Dear Daniel, >>>>>> >>>>>> Thanks for your quick response >>>>>> >>>>>> We can able to create operseb db on remote mysql server ,but >>>>>> Kamailio not getting start >>>>>> >>>>>> Error detail: >>>>>> >>>>>> ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to >>>>>> MySQL server on '192.168.1.170' (4) >>>>>> ERROR: <core> [db.c:32 0]: could not add connection to the pool >>>>>> ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database >>>>>> ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while >>>>>> initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) >>>>>> >>>>>> >>>>>> I have attached my kamailo.cfg ( we followed Kamailio and >>>>>> Asterisk real time installation) ,please advice >>>>>> >>>>>> With Regards >>>>>> >>>>>> N.Prakash >>>>>> >>>>>>> >>>>>>> ---------- Forwarded message ---------- >>>>>>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>>>>>> Date: Tue, Feb 12, 2013 at 2:41 PM >>>>>>> Subject: Re: [SR-Users] openser db on an external mysql server >>>>>>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>>>>>> Users Mailing List" <sr-users(a)lists.sip-router.org>, >>>>>>> sr-dev(a)lists.sip-router.org, miconda(a)gmail.com >>>>>>> Cc: Prabhakaran B <prabhakaran(a)tevatel.com> >>>>>>> >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> kamailio database is created with: >>>>>>> >>>>>>> kamdbctl create >>>>>>> >>>>>>> the host of mysql server has to be specified in kamctlrc file via >>>>>>> DBHOST variable - kamctlrc is located in same directory as kamailio.cfg. >>>>>>> >>>>>>> In kamailio.cfg, update the db_url value (or DBURL define) to >>>>>>> reflect the mysql server host. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> On 2/12/13 4:16 AM, Prakash N wrote: >>>>>>> >>>>>>> Hi All, >>>>>>> >>>>>>> I am new this Kamailo ,ther is any step by step document to set >>>>>>> up openser db on an external mysql server ? >>>>>>> >>>>>>> With Regards >>>>>>> >>>>>>> N.Prakash >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> Kamailio World Conference, April 16-17, 2013, Berlin >>>>>>> - http://conference.kamailio.com - >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> sr-dev mailing list >>>>>> sr-dev(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Muhammad Shahzad >>>>> ----------------------------------- >>>>> CISCO Rich Media Communication Specialist (CRMCS) >>>>> CISCO Certified Network Associate (CCNA) >>>>> Cell: +49 176 99 83 10 85 >>>>> MSN: shari_786pk(a)hotmail.com >>>>> Email: shaheryarkh(a)googlemail.com >>>>> >>>>> >>>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.com >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > >

4 18

GIT master branch is open for development of v4.1.x
by Daniel-Constantin Mierla 27 Feb '13

27 Feb '13

Hello, the GIT master branch is again open for new development. Common testing period of releasing v4.0.0 has ended, if you have pending commits or personal branches to merge, you can go ahead. v4.0.0 will be released out of newly created branch 4.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

How to remove custom header after adding one
by Mino Haluz 27 Feb '13

27 Feb '13

Hi, if I use append_hf() for adding my custom header to initial request, how I can remove this header later ? What I'm doing: 1) initial request: remove RPID header with remove_hf 2) initial request: add custom RPID header append_hf 3) after rerouting: in failure_route remove RPID header remove_hf 4) after rerouting: in failure_route add custom RPID header with append_hf The result of this, are 2 RPID headers, because remove_hf did not catch RPID header added in step 2, but I think the original one that was removed already in step 1. Thanks, Mino

3 3

GIT branch 4.0
by Daniel-Constantin Mierla 27 Feb '13

27 Feb '13

Hello, the GIT branch 4.0 has been created to host the release series for v4.0.x. It must take only bug fixes and documentation improvements related to the features existing at this moment in the branch. To get the branch locally, use: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 4.0 origin/4.0 To push the local commits on remote repository: git push origin 4.0:4.0 Any fixes that have to get in the branch 4.0 should be committed first in master branch and then cherry-picked in this branch, in order to be able to track forgotten backports easier. Testing for 4.0 went very good so far, in few days we will probably sketch the path to the release day. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

siremis CDR gneration procedure
by Klaus Darilion 27 Feb '13

27 Feb '13

Hi! IMO the siremis CDR generation procedure is buggy: CREATE PROCEDURE `kamailio_cdrs`() DECLARE inv_cursor CURSOR FOR SELECT src_user, src_domain, dst_user, dst_domain, dst_ouser, time, callid,from_tag, to_tag, src_ip FROM acc where method='INVITE' and cdr_id='0'; ... REPEAT FETCH inv_cursor INTO v_src_user, v_src_domain, v_dst_user, v_dst_domain, v_dst_ouser, v_inv_time, v_callid, v_from_tag, v_to_tag, v_src_ip; IF NOT done THEN SET bye_record = 0; SELECT 1, time INTO bye_record, v_bye_time FROM acc WHERE method='BYE' AND callid=v_callid AND ((from_tag=v_from_tag AND to_tag=v_to_tag) OR (from_tag=v_to_tag AND to_tag=v_from_tag)) ORDER BY time ASC LIMIT 1; IF bye_record = 1 THEN INSERT INTO cdrs (src_username,src_domain,dst_username, dst_domain,dst_ousername,call_start_time, duration,sip_call_id, sip_from_tag,sip_to_tag,src_ip,created) VALUES (v_src_user, v_src_domain,v_dst_user, v_dst_domain,v_dst_ouser,v_inv_time, UNIX_TIMESTAMP(v_bye_time)-UNIX_TIMESTAMP(v_inv_time), v_callid,v_from_tag,v_to_tag,v_src_ip,NOW()); UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid AND from_tag=v_from_tag AND to_tag=v_to_tag; END IF; SET done = 0; END IF; UNTIL done END REPEAT; ... 1. The UPDATE query will not UPDATE BYE records which were sent from CALLEE to CALLER. This can be easily fixed with: UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid AND ( (from_tag=v_from_tag AND to_tag=v_to_tag) OR (to_tag=v_from_tag AND from_tag=v_to_tag) ); 2. (minor) It does not consider reINVITEs. I know the default Kamailio config does not account reINVITEs, but I think when reINVITEs are accounted too, this can mess up the CDR generation. regards Klaus

3 13

kamailio call transfer
by kiran bhosale 27 Feb '13

27 Feb '13

hello, we want to implement call transfer with opneser. but the fact is we have not implemented any database and implemented own module for contact storage. i tried using avpops module. but couldn't do please suggest

1 0

Kamailio and Geoip
by Ian French 26 Feb '13

26 Feb '13

Hi list, I've been trying to get Kamailio using the geoip module to deny traffic from countries not matching certain sources, but I'm unsure of how to achieve this I've tried the below which failed. I'm trying to drop requests outside say 2 countries "AA" or "BB" geoip_match("$si", "src"); if($gip(src=>cc)!="AA" || $gip(src=>cc)!="BB") { xlog("L_INFO","SIP message from prohibited source $gip(src=>cc) (ip: $si)\n"); send_reply("403", "Forbidden"); exit; } else { xlog("L_INFO","SIP message from unprohibited source $gip(src=>cc) (ip: $si)\n"); } Can you tell me where I am wrong or how I can achieve this Thanks in advance Ian

1 0

announcing DruCall WebRTC for Drupal
by Daniel Pocock 26 Feb '13

26 Feb '13

I've just released DruCall, the WebRTC module for Drupal. http://danielpocock.com/announcing-drucall-webrtc-for-drupal Making WebRTC available as a Drupal module hopefully means thousands of blogs and small business web sites will start picking this up and hopefully Kamailio will share in some of the demand for WebRTC capable server infrastructure.

1 0

Re-registration
by Abdullah-al-mahmud Al-amin 26 Feb '13

26 Feb '13

Hi, I'd created an account with my domain shaluavita. My user id was alamin(a)shaluavita.tk. Unfortunatly, I canceled the account. It war about 2 moth ago. I tried to re-register many times but failed for remaining the domain in the database. Now I want to use my account. Please re-activate it. Thanks

1 0

Stale dialogs in database - stale reference
by David 26 Feb '13

26 Feb '13

Hey all, I am dealing with a possible race condition in the dialog module. About one call in 50 isn't cleared from the database. I catch the calls by joining the dialog table to my cdr table ( which is generated from acc table ). The call remains in the dialog table in the database. I tested, and I am not hitting the destroy_dlg() function which is called from the unref_dlg_unsafe macro. destroy_dlg() calls the remove_dialog_from_db(). unref_dlg_unsafe is called but the ref counter is still at one. I can see this from my logs : 11:22:55 [20195]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 1 to state 3, due event 3 (ref 2) <== de ringing a repondu 11:22:55 [20172]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 3 to state 4, due event 6 (ref 3) <== answer 11:22:55 [20202]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 3) <== answer 11:22:55 [20177]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 4) <== answer 11:23:05 [20190]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 5, due event 7 (ref 3) <== BYE, hangup 11:23:05 [20183]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 5 to state 5, due event 7 (ref 5) <== confirm hangup The problem seems to be happening randomly and I can't find any explanation for it. So the number of references never reaches 0, because it never reaches 0, the dialog is not unreferenced : Here is the output from syslog ( grepped with the call hash ) : Feb 26 11:21:41 kamailio-1 /usr/sbin/kamailio[20164]: ERROR: dialog [dlg_hash.c:717]: ref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:21:41 kamailio-1 /usr/sbin/kamailio[20164]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 5, due event 7 (ref 3) Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 2 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20170]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20170]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 2 -> 0 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:757]: ref <=0 for dialog 0x7fab27f4aa88 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:317]: destroying dialog 0x7fab27f4aa88 (ref 0) Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:333]: removed timer for dlg 0x7fab27f4aa88 [7298:10958] with clid '13b951b67f2704d51d25cf643b3469d1(a)pbx.omnity.biz' and tags 'as3a8e0df2' 'as4c79d466' Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:717]: ref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_cb.c:230]: dialog=0x7fab27f4aa88 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 1 to state 3, due event 3 (ref 2) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_handlers.c:484]: dialog 0x7fab27f4aa88 confirmed (ACK pending) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=8 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20174]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20174]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20178]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20178]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 3 to state 4, due event 6 (ref 3) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 3) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=32 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 4) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=32 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 5, due event 7 (ref 3) Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=64 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 5 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 5 to state 5, due event 7 (ref 5) Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 5 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 2 -> 2 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 You can see the last refence is never removed, so the dialog isn't destroyed. Here is the end of a healthy dialog : Feb 26 08:03:00 kamailio-1 /usr/sbin/kamailio[4668]: ERROR: dialog [dlg_hash.c:753]: unref dlg 0x7f74db910e60 with 2 -> 0 Feb 26 08:03:00 kamailio-1 /usr/sbin/kamailio[4668]: ERROR: dialog [dlg_hash.c:753]: ref <=0 for dialog 0x7f74db910e60 Feb 26 08:03:00 kamailio-1 /usr/sbin/kamailio[4668]: ERROR: dialog [dlg_hash.c:315]: destroying dialog 0x7f74db910e60 (ref 0) You see it falls to 0 and the dialog is destroyed. So, how do I debug this ? Should I maybe be posting to sr-dev ? How do I determine who is incrementing the counter without decrementing it? What information do I need to continue with this problem? Thanks! David ps I used perl pie to replace LM_DBG with LM_ERROR in modules_k/dialog/* version: kamailio 3.3.2 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 09:17:03 Oct 18 2012 with gcc 4.4.3

1 1

Bug in AVPOps?
by Carsten Bock 26 Feb '13

26 Feb '13

Hi, i use the AVPops module to store information about Call-Forwardings in the database: # Store new CF-Target in Database: $avp(s:usr_cfnr) = $rU; avp_db_store("$avp(s:caller_userid)", "$avp(s:usr_cfnr)/usr_preferences"); $avp(s:usr_cfnr_delay) = 20; avp_db_store("$avp(s:caller_userid)", "$avp(s:usr_cfnr_delay)/usr_preferences"); (please not, that $avp(s:caller_userid) was not changed) however this results into two different inserts, one for user 7736 (correct) T 2013/02/25 14:25:11.704628 192.168.0.2:51460 -> 192.168.0.20:3306 [AP] {....insert into usr_preferences (uuid,attribute,value,type,username,domain ) values ('7736','usr_cfnr','491792021244',0,'','') and one for user 7720 (20???, as in the delay, incorrect) T 2013/02/25 14:25:11.705572 192.168.0.2:51460 -> 192.168.0.20:3306 [AP] w....insert into usr_preferences (uuid,attribute,value,type,username,domain ) values ('7720','usr_cfnr_delay','20',1,'','') I am using Kamailio 3.3.2 from GIT-Trunk (nightly, checked out on December 14th): root@proxy1:/# kamailio -V version: kamailio 3.3.2 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 01:57:11 Dec 14 2012 with gcc 4.4.5 Before i dig deeper into the code of the avpops module, has anyone seen / fixed this issue? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ============================================================= Visit us on the Kamailio World 2013, 16./17. April 2013 => http://conference.kamailio.com/k01/ See you there! ============================================================= ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

2 2

Errors leading to kamailio proc crash
by Grant Bagdasarian 26 Feb '13

26 Feb '13

Hello, Last Thursday our kamailio process crashed. There are a lot of memory related errors in there. See attachment for details. Did the machine run out of memory? How do I prevent this from happening again? The machine has 4GB of memory. Regards, Grant

2 4

Loop detection in Kamailio
by Olle E. Johansson 26 Feb '13

26 Feb '13

Hi! I can see that there's been some discussion about SIP loop detection on the OpenSER and SER mailing lists, but I must have missed the conclusion. Apart from setting a max-forwards, how would you configure your Kamailio server to prevent looping? Regards, /O

3 2

Kamailio multiple asterisk boxes
by imcdona＠gmail.com 26 Feb '13

26 Feb '13

Hello, I've been having some issues with the syntax of Kamailio's config file. I've been following the turorial here: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb I'd like to setup Kamailio to act as an SBC providing NAT traversal, and security for multiple asterisk boxes on the inside. Currently kamailio has two NIC's, one public and one private. All traffic hits the external interface and is then forwarded to an Asterisk box on the inside. I'd like to modify the config from the tutorial above to support multiple asterisk boxes. Authentication will be done via MySQL and depending on the user domain, the request get's forwarded to the proper Asterisk box. Here is what I've got so far: if (is_method("REGISTER") && from_uri=~.*(a)10.55.55.5) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); route(REGFWD); exit; } else if (is_method("REGISTER") && from_uri=~".*asterisk.domain.tld") { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); route(REGFWDCLOUD); exit; } } Based on the domain I route the SIP message to another route block which is esentally the same route block as the origional route[FROMASTERISK] route[TOASTERISK] route[REGFWD]. I beleive there's an easier way of doing this though. Creating separate route blocks per asterisk box would mean creating a new route blocks for each asterisk box. Here's an example: route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $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: " + $hdr(Expires) + "\r\n"; uac_req_send(); } route[FROMCLOUD] { if($si==$sel(cfg_get.cloud.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } route[TOCLOUD] { $du = "sip:" + $sel(cfg_get.cloud.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } route[REGFWDCLOUD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.cloud.bindip); $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(); } I'm sure their is any easier way of doing this alhthough I'm not sure how. Can anyone point me in the right direction? Lastly, is there a way to modify the DBURL string on the fly so based on the domain for example, I coudl query a different DB for user credentials? Thanks!

2 1

Kamailio v3.3.4 Released
by Daniel-Constantin Mierla 26 Feb '13

26 Feb '13

Hello, Kamailio SIP Server v3.3.4 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1, v3.3.2 or v3.3.3. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.4. For more details about version 3.3.4 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/02/kamailio-v3-3-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

Re: [SR-Users] SBC
by Keith 26 Feb '13

26 Feb '13

Hi, I am looking to build an SBC/SIP router made up from Kamailio and FreePBX. Calls will then need to be passed to multiple Asterisk media gateways. Can anyone point me to some documentation or help on how to do this? -- Keith

6 7

planning to release Kamailio v3.3.4
by Daniel-Constantin Mierla 25 Feb '13

25 Feb '13

Hello, I am considering packaging a new minor release for branch 3.3 next Monday, on February 25. Please backport any patches that you have to be part of v3.3.4 Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 4

Changing outgoing/source port
by Efelin Novak 25 Feb '13

25 Feb '13

Hi, I'd like to ask whether I am able to change an outgoing port. I'm using t_relay for outgoing INVITEs & Kamailio uses 5060 port defined in listen. When the call fails I am rerouting it using append_branch and I'd like to change an outgoing port to 5061. The reason I'd like to do this is a Loop detection of my buggy VoIP/analog gateway. Am I able to specify an outgoing interface/port for t_relay in Kamailio or should I find an another gateway? :) Thanks for help Kind regards Efelin

2 1

kamailio
by Suhaib Mehyar 25 Feb '13

25 Feb '13

Dears, My name is suhaib mehyar, I tried to install kamailio server, and I successfully set it up, it is locally running fine, both audio and video are very good locally but when I call a client that is connected to the server that is exist after my cisco firewall with private ip and the client is exist after his home router both audio and video are not working. Kindly any advice please??? Best Regards, Suhaib Mehyar Skype : suhaib.mehyar Mob :+962 795 161311

4 3

Kamailio World Updates
by Daniel-Constantin Mierla 25 Feb '13

25 Feb '13

Hello, the next group of accepted talks at Kamailio World Conference will be published in few days, keep an eye on: - http://conference.kamailio.com/k01/schedule/ Till then, here is what happened lately around the project. Olle E. Johansson spent the week at SIPit, making sure the upcoming version 4.0.0 is ready for release: - http://www.kamailio.org/w/2013/02/sipit30/ I conducted the Kamailio Development Workshop, with a great group of participants, first results being visible as contribution to app_lua for review on tracker. Today is schedule to release Kamailio v3.3.4, a new minor version out of latest stable branch 3.3. This week on Wednesday, development branch will be unfrozen, lots of new code is already waiting to be merged there (to be part of future version 4.1.0). If you are in Berlin this week, participating at Call Center World (http://www.callcenterworld.de/en/), ping me to meet, I will be glad to catch up with what is happening on real time communications market. Do not forget that there are open positions to work with VoIP/SIP and Kamailio, just in case you look for new challenges in your career, check: - http://conference.kamailio.com/k01/career-jobs/ It is less than two months till Kamailio World Conference, the schedule is almost filled up, if you haven't registered yet, it is time to hurry up, don't miss the chance to meet many professionals working with large real time communications systems or solutions, from Kamailio or different other projects: - http://conference.kamailio.com/k01/registration/ At the end, many thanks to those sponsoring the event, helping to cover the logistics to ensure a great event. If you consider sponsoring, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)
by Rowie 24 Feb '13

24 Feb '13

Hi, We are having an issue where a phone (snom in particular) cannot make a call through Asterisk. It just hangup and does not allow the call to go through. I am including a a sip trace on this thread to show what is happening within the call. Please see below: Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 3

SOS!! Kamailio 3.1.x Load Balancer doesn't work
by chanea 24 Feb '13

24 Feb '13

Dear All, I am trying to use Kamailio 3.1.x server (called LB here) as the Load Balancer for two other Kamailio server (called server01 and server02 here) . Client A -----> Kamailio LB -----> Kamailio server01 Client B -----> Kamailio LB -----> Kamailio server01 If I dont use LB here , and letting Client A and B connect to Kamailio server01 directly , all the communication is fine. While using Kamailio LB ,without luck, although the sip client A and B get 200 from server01 for Register, they cannot reach each other for "Message" and "Invite". It will be great appreciated if anyone can give a hint. my config for Kamailio LB is as below : ========================================== .... debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 # for more info: sip_router -h # ------------------ module loading ---------------------------------- mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "dispatcher.so" loadmodule "db_mysql.so" # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("dispatcher", "db_url","mysql://.........@........") modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; # if (!method=="REGISTER") # record_route(); ds_select_dst("1","4"); forward(); exit; } .. ========================================== best regards, chanea

2 3

SIPit 30 summary
by Olle E. Johansson 22 Feb '13

22 Feb '13

Hi! I've spent five days at SIPit 30 testing Kamailio on a tiny Asus EEE-netbook. It's been great, and apart from the bug hitting me while using outbound, we've done great in all tests. During the proxy doom test today, the netbook processed way over 50.000 messages in a single INVITE transaction that spiralled and looped and behaved in funny ways, and the CPU still had a lot of power to do other things. The hub we used was the limiting factor. Kamailio 4.0.0 beta was involved in all multiparty tests, from TLS and RTCweb to forking, spiralling and proxy doom. We proved that we have a solid product that can handle all kind of scenarious, even TCP overflow attacks. Thanks for all the help over jabber, IRC and other means! Special thank you to Fred Posner who contributed a Big Fred Cookie! Have a great weekend! /O

3 2

Path and DNS SRV
by Charles Chance 22 Feb '13

22 Feb '13

Hi All, Given a situation where there are multiple registrars, behind a single load balancer, sharing the same location table in DB, I understand that Path can be used to ensure everything works nicely with clients behind NAT. So, if UA1 is registered on P1, UA2 on P2, and UA1 calls UA2... P1 performs lookup, sees path and sends via P2. What about situations where the distribution between registrars is being handled client-side through DNS SRV, not a load balancer? Path cannot be used then, right? As I understand it, in this instance add_path() will do nothing, is this correct? 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: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

3 4

Weird out-of-memory message on start/stop
by Andreas Granig 21 Feb '13

21 Feb '13

Hi, Maybe some of you have encountered this funny error message when starting and even killing kamailio on a Debian 6 with 2.6.x and 3.x kernels and 2.11.3-4: #------------------------- # /usr/sbin/kamailio -f /etc/kamailio/proxy/kamailio.cfg -P /var/run/kamailio/kamailio.proxy.pid -m 4 -M 16 -u kamailio -g kamailio loading modules under /usr/lib/kamailio/modules_k:/usr/lib/kamailio/modules Listening on udp: 127.0.0.1:5062 tcp: 127.0.0.1:5062 Aliases: Message from syslogd@sp2 at Feb 19 11:15:42 ... t of memory [6720] # killall kamailio Message from syslogd@sp2 at Feb 19 11:17:51 ... t of memory [6720] #------------------------- The PID reported here is the kamailio attendant process. It seems to be somehow related to the shared memory. The interesting part about that is that kamailio really starts just fine, and there is plenty (>5GB) of free memory available. It doesn't seem to influence the behaviour of kamailio at all as it's working perfectly fine, so for now it's just annoying, but I'm really curious I've tried to fiddle with mem_join, but it doesn't matter whether it's on or off. Kamailio version is this: # kamailio -V version: kamailio 3.3.2 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 19:09:17 Jan 21 2013 with gcc 4.4.5 Has anyone seen this too? What's the cause of that? Andreas

4 7

SDP c= address as a key for a key-value lookup?
by Hegner, Travis 21 Feb '13

21 Feb '13

Hi All, I am working on a kamailio proxy with a very specific goal I need to accomplish. What I would like to do is a key-value lookup, utilizing the SDP's media contact (c= line) IP as a key, and have a function return a value for that key. It appears that I can possibly accomplish this with the AVPops module, combined with DB_TEXT (maybe DB_MYSQL in the future) and a custom table with a custom query. The only thing I seem to be missing is the ability to extract the c= IP from the SDP, and store it as some kind of variable with which to do the lookup. I've looked into the iptrtpproxy module as it seems to have a select: @iptrtpproxy.sdp_ip that could work, but that module is extreme overkill for my purposes, and I am running into some issues getting it built for my platform (Ubuntu server 12.04). Also the documentation for that select states: "Return first rewritten IP provided at SDP c= line." However, I'm looking for the original sdp_ip, and have some concern that this won't return an IP that has not yet been re-written. My environment has a very specific 1:1 ratio of private network, to static public IP mappings. I currently have a successful OpenSER (1.3.2) deployment within this environment (running 5 years), which uses a hacked version of the old "mangler" module, with the function sdp_mangle_ip(<network>, <ip>). I could do something similar using textops->search_body() and nathelper->fix_nated_sdp(), but the risk of a typo or syntax error in the config file every time I have to make a change or add a network is fairly high. Less importantly, there is some extra overhead as the proxy will have to test for every network on every SDP looking for a match (this is what the current deployment does). Eventually I'd like to add a simple frontend to add/edit networks on the fly, so that no one is ever touching the config. What would you folks recommend for this? Should I look at building a custom module which only grabs the c= IP and stores it into an AVP? Am I not seeing another module that will expose that information for me already? Would it simply be easier to work on getting iptrtpproxy built, and utilizing it only for the purpose of extracting the information I need from the SDP? Any insight would be most greatly appreciated. Thanks, Travis Hegner ________________________________ The information contained in this communication is confidential and is intended only for the use of the named recipient. Unauthorized use, disclosure, or copying is strictly prohibited and may be unlawful. If you have received this communication in error, you should know that you are bound to confidentiality, and should please immediately notify the sender or our IT Department at 866.459.4599.

5 6

Difference between shared and script variable
by Mino Haluz 21 Feb '13

21 Feb '13

Hi, what is the difference between shared and script variable ? Thanks Mino

2 3

Re building command
by Techie Sup 20 Feb '13

20 Feb '13

Hi all, Sorry for this simple question. 1) I have made some changes to main.c file and want rebuild kamailio so that changes take Éffect, what is the command to rebuild? Thanks Selvam

2 1

using dlg_manage before or after?
by Uri Shacked 20 Feb '13

20 Feb '13

Hi, I am trying to change the timeout of a dialog based on the $avp() I set on the modeparam. While testing, I notice that the change does not apply after I am using dlg_manage, only if i changed it before. Can anyone explain why is that? and is it possible to change it after doing dkg_manage? Thanks, URi

2 1

mojtaba jahanbakhsh
by mojtaba jahanbakhsh 20 Feb '13

20 Feb '13

http://www.firstchoiceforyou.com/ecj/xnyfbj4f=4dbl9cu52nkdz

1 0

Strip All but numbers/*/#
by Ali Jawad 20 Feb '13

20 Feb '13

Hi I am using the below to strip "-" from destination numbers, it does work just fine "I replace - with nothing", what i need to do now is to strip all characters except 0-9 * and #. Is there anyway to do it using the following configuration part if($rU =~ ".*-.*" ) { $rU = $(rU{re.subst,/-//g}); } Regards

4 5

Re: [SR-Users] kamctl restart and mediaproxy
by Victor Seva 19 Feb '13

19 Feb '13

We use mediaproxy and kamailio and using the init.d script from the Debian package to restart kamailio has no effect on mediaproxy process. 2013/2/8 Eduardo Lejarreta <lejarreta.e(a)sarenet.es>: > Good morning Daniel. > > I send you the "messages" log which is a bit filtered. The flow is: > > Start a call, engage_mediaproxy on ports (50156, 50157, 50158, 50159) and > then "kamctl restart". As you see it seems that an instruction from kamailio > tells mediaproxy to drop the ports. > We have tried to script around "kamctl" to avoid sending orders to > mediaproxy, killing the "media-dispatcher" before Kamailio and then > restarting again (media-dispatcher) and this works OK, but seems not be a > reasonable way to fix it. > > ...... > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50156 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50157 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50158 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50159 > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3461]: NOTICE: <core> > [main.c:706]: Thank you for flying kamailio!!! > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3561]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3558]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3560]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3553]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3557]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3549]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3552]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3547]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3545]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3546]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3541]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3538]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3536]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3526]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3492]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3528]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3521]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3530]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3495]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3532]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3493]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3539]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3523]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3534]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3489]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3496]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3488]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3525]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3487]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3498]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3484]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3500]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3482]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3501]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3479]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3499]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3474]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3476]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3475]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3477]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3491]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3480]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3490]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3486]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3483]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50156 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50157 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50158 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50159 Closed) > Feb 8 11:05:08 kamailio-dev kamailio: INFO: <core> [tcp_main.c:4727]: > init_tcp: using epoll_lt as the io watch method (auto detected) > Feb 8 11:05:08 kamailio-dev /sbin/kamailio[3761]: INFO: usrloc > [hslot.c:53]: locks array size 512 > ...... > > Thanks and regards. > -- > Eduardo Lejarreta. > > -----Mensaje original----- > De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Enviado el: viernes, 08 de febrero de 2013 9:38 > Para: lejarreta.e(a)sarenet.es; SIP Router - Kamailio (OpenSER) and SIP > Express Router (SER) - Users Mailing List > Asunto: Re: [SR-Users] kamctl restart and mediaproxy > > Hello, > > not using mediaproxy, but have you looked at its logs to see if prints some > messages? > > Cheers, > Daniel > > On 2/7/13 5:03 PM, Eduardo Lejarreta Langarica SARENET wrote: >> Good evening again. >> >> Kamailio 3.2.2 and mediaproxy 2.5.2 >> >> An long-time issue for us: >> >> kamctl restart stops mediaproxy audio. >> >> is there anyway to avoid this? >> >> Any idea will be helpful. >> >> Thanks and best regards. >> ---------------------------------------------------------------------- >> ------------------------------------------------------------- >> Eduardo Lejarreta >> >> >> >> >> _______________________________________________ >> 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 > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 5

branching for v4.4.x and unfreezing development
by Daniel-Constantin Mierla 19 Feb '13

19 Feb '13

Hello, it is time to sketch the road to the next major release, my plan is: - create branch 4.4 on Wednesday, February 27, 2013 - that will unfreeze development, so personal branches and contributions kept in external repositories can be merged in master branch - consider to release v4.4.0 in first or second week of March Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

Nonce secret and expiry value
by Andreas Granig 19 Feb '13

19 Feb '13

Hi guys, Nonces have a default expiry value of 300 seconds. Does this still apply if the "secret" parameter is set? So can you use both "secret" and "nonce_expire" value together, taking for granted that the time is in sync over multiple proxies? Andreas

2 2

2 kamailio servers as proxy
by Andrei Rotaras 19 Feb '13

19 Feb '13

Hello, We have this situation: 2 servers, each with his own "location". First user is registered on one server and second user on other server. If we receive an invite for a user that is on other server's location, second server doesn't react on hostrewrite(dynamic routing,forward, append_branch, t_replicate, etc) from first server. Can you please advise us in what direction to look? Our kamailio version: kamailio-3.3.1-1.2.x86_64. (installed from rpm) Thank you! Best regards, Andrei Rotaras NOC Systems Engineer StarNet Moldova Phone: +(373-22) 844-770 Fax: +(373-22) 844-445 Mob: +(373-79) 700-552 E-mail: arotaras(a)starnet.md<mailto:arotaras@starnet.md> Address: 4th Floor, Calea Iesilor 10, Chisinau, Republic of Moldova Web: http://www.starnet.md<http://www.starnet.md/> [Description: Description: Description: starnet] This email message is confidential and may contain privileged information. If you are not the above named addressee, it may be unlawful for you to read, copy, distribute, disclose or otherwise use the information in this email message. If you are not the intended recipient of this email message please telephone one of the above indicated numbers immediately and delete this message.

2 1

Any substitute or replace function for the request-/response line of SIP messages available?
by Klaus Feichtinger 18 Feb '13

18 Feb '13

Hi, I have a very specific situation that the implementation of a customer SIP user agent is absolutely not standards conform. Therefore, I use Kamailio as a SIP protocol adapter between the non-standard conform user agent(s) and the typical RFC3261 compliant user agents. Most conversions are working fine. But in a very specific scenario I have to change/convert a SIP response message to a SIP request message and change the SIP method´s name. This conversion is no problem for the SIP header fields. However, I haven't found a way for adapting the response-/request line of a SIP message. This line seems to be protected. The described situation of the perverse conversion I try to realize is following (please do not ask "why?" or think I'm mad...): (mad) UA1 == INVITE ==> SIP-Prot-Adapter(Kamailio) == INVITE ==> (typ.)UA2 UA1 <== BYE == SIP-Prot-Adapter(Kamailio) <== 488 == (typ.)UA2 UA1 == BYE ==> SIP-Prot-Adapter(Kamailio) == ACK ==> (typ.)UA2 I use the "textops" module for substitution and/or replacement in SIP header fields. But the "subst" and "replace" functions cannot be used for the request-/response line on top of the SIP messages. I was a little bit irritated, as the "search" function is still working on these lines, but not the subst/replace function. Creating a new SIP message with the UAC module (as alternative) is not okay, as this new message does not use the same values for Call-ID, From-tag a.s.o. and these values can't be changed before sending the message. Does anybody know a way / function / method for manipulating the request-/response line of a message? It would be perfect using a function á la "sed"/subst. Thanks Klaus P.S. I tested with kamailio-3.3.3

2 3

Kamailio and Ubuntu Upstart script
by Moacir Ferreira 17 Feb '13

17 Feb '13

As I posted here before, Ubuntu has moved to Upstart as the main engine to start daemons. As a result, Kamailio will fail to start on boot time because it starts before MySQL that has already been moved to the new Upstart daemon starting fashion. I am struggling to get some help from the Ubuntu community explaining how we can match the previous init script behavior when using Upstart but I got no success till now. Anyway, I have created an Upstart script for Kamailio that does most everything that the old script use to do. However, in this script I am making sure MySQL starts before Kamailio. If you have another SQL engine you must change the script to match your SQL engine by changing the “start on started” statement on the script. To use it you have to: A – Create a kamailio.conf file at “/etc/init/” with the contents of the kamailio.conf script in attach, that is the new Upstart stile init script. Please notice that I just adapted the existing script, no news… B – Create a symbolic link at /etc/init.d/ pointing to this kamailio.conf at /etc/init/. C – If you want, you can still have a /etc/default/kamailio file like it used to be and the script will behave like it used to behave when using the default file. Hopefully someone more “influent” at the Ubuntu community may get some help on how to adapt their “start on started” Upstart stanza to behave like we need for multiple possible SQL engines. Cheers! Moacir

3 2

app_python rewrite_ruri - BUG: do_action bad set*()
by V Tone 15 Feb '13

15 Feb '13

Hi, I'm having problems getting started with the app_python module. I'm getting the following in the logs: 5(8890) : <core> [action.c:785]: BUG: do_action: bad set*() type 14 5(8890) ERROR: <core> [action.c:1568]: run action error at: :0 5(8890) ERROR: app_python [python_msgobj.c:115]: Error in do_action 5(8890) ERROR: app_python [python_support.c:38]: python_exec2: Unhandled exception in the Python code: 5(8890) ERROR: app_python [python_support.c:78]: RuntimeError: Error in do_action My Python module is very basic: class Router(object): def child_init(self, child_id): return 0 def route(self, msg): msg.rewrite_ruri('sip:1000@1.2.3.4:5060') return 0 def mod_init(): instance = Router() return instance I've tested this in Kamailio 3.2 and 3.3. Can someone point me to what I'm doing wrong? Many thanks in advance, vtone

2 4

siptrace/sipcapture and MSRP
by Peter Dunkley 15 Feb '13

15 Feb '13

Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter

3 7

Warning - What does this mean?
by Krishna Kurapati 15 Feb '13

15 Feb '13

I am seeing this error lately. /usr/local/sbin/kamailio[28559]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction What does this mean? Krish Kura

1 0

Dialog is not cleaned up after failed calls and BYEs.
by Aft nix 15 Feb '13

15 Feb '13

Hi, I'm trying to limit concurrent calls through kamailio using following script: #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$fd", "$var(100)"); if ($var(100) >= 100 ) { xdbg("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); } #!endif But the dialogs are not cleaned up after the call ends or a failed call happens. For example after two failed calls : [root@new tls]# kamctl fifo profile_get_size callquota profile:: name=callquota value= count=4 and the dialog list : [root@new tls]# kamctl fifo dlg_list dialog:: hash=896:11654 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11755 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=896:11655 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11755 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=896:11657 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11756 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=1532:8089 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: povqQaC.sRhKvmUaLzzR83bRXxouoNy8 from_uri:: sip:123@205.164.40.150 from_tag:: qXQll3QXrff.t27FHY2hucPfJ7K8Pfrs caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 9232 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: You can see there are multiple entries. And these dialog does not get cleaned up. The data is collected when there was no call(pending or connected). My kamailio version is : [root@new tls]# kamailio -V version: kamailio 4.0.0-pre0 (x86_64/linux) 83eafc flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_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: 83eafc compiled on 18:34:58 Jan 17 2013 with gcc 4.4.6 Any suggestion?(It seems dialog module is not working as it should be. I've also tried calling unset_profile() forcefully in failure route with no luck) -- -aft

1 0

Registration and TCP connection
by Krishna Kurapati 15 Feb '13

15 Feb '13

Hi, When client connects to server over TCP/TLS and REGISTERs two things that can happen. 1. Client can disconnect the connection when the App is killed abruptly and the client did not get a chance to send unregiter. 2. CRLF KA did not arrive due to network failures in the middle. Is there a way to remove contact from usrloc table. This can spare server from trying to send message to a client who is no longer connected to the server. It would be ideal if this can be done automatically with a module param in tcp or registrar. Any suggestion on how to tie TCP connection with usrloc? Thanks Krish Kura

2 1

kamailio - simple presence over port 5061
by David Thomson 15 Feb '13

15 Feb '13

I was wondering if simple presence can work over port 5061 or if it always goes over 5060. my client (jitsi) is configured to work over port 5061, xcap is configured to work over port 5061 but I am still seeing some messages flowing over port 5060. any ideas as to what is going on? ttyl,Dave

2 1

[Fwd: Re: Deleting the expired contacts using the Rtimer module]
by kiran bhosale 15 Feb '13

15 Feb '13

he array values that we are modifying and the function that we call periodically are from our module . to exemplify i have put the code below .our requirement is that we want to delete the expired contacts without relying on usrloc module as "we have not used any DB for contact storage" but developed our own module for storing them in a file.even we had installed the SIGALRM handler in our module for decrementing the contacts but the values of array (that stores the expires) elements were zero. we thought of workaround to store the expires values to file and decrement them periodically. but frequent I/O is very consuming. whats the solution for deleting the expired contacts . #define DB_PATH "/usr/local/etc/kamailio/dbtext/syntel" #define DB_PATH1 "/usr/local/etc/kamailio/dbtext/temp" #define MAX_BUFFER 128 #define MAX_USERS 128 typedef struct { int max_bindings; int syntel_index; int expires[8]; char ip_addr[8][16]; } Syntel_users; Syntel_users syntel_database[16]={{0}}; //this is function used for storing the contacts int write_to_file (char *uri, char *user, int locator,int expires) { int Retval = 0; int fd; int bytes; char ip[13]; strncpy (ip, &uri[9], 12); ip[12] = '\0'; fd = open (DB_PATH, O_CREAT | O_RDWR | O_APPEND, 0777); if (syntel_database[locator].max_bindings < 8) { strcpy (syntel_database[locator].ip_addr[syntel_database[locator].max_bindings], uri); syntel_database[locator].expires[syntel_database[locator].max_bindings]=expires; syntel_database[locator].max_bindings++; bytes = write (fd, uri, strlen (uri)); Retval = 1; } return Retval; } //AND THIS IS FUNCTION WE CALL FROM kamailio.conf WITH RTIMER MODULE AFTER INTERVAL OF 1 SECOND. int syntel_delete (void) { int counter; int j; for (counter = 0; counter < 16; counter++) { for (j = 0; j < syntel_database[counter].max_bindings; j++) { if(syntel_database[counter].expires[j]<=0) { memset(syntel_database[counter].ip_addr[j],0,strlen(syntel_database[counter].ip_addr[j])); syntel_database[counter].max_bindings--; } else syntel_database[counter].expires[j]--; } } return 0; } BUT WHAT THIS FUNCTION GETS IS ZERO VALUES FOR EXPIRES AND NOT MODIFIED IN SYNTEL_WRITE_TO_FILE HOW TO DECREMENT THESE VALUES PERIODICALLY THEN ??

2 3

Fwd:
by Maurice Vistan 14 Feb '13

14 Feb '13

http://www.hotelgloriachiancianoterme.it/ako7ed.php?s=lf

1 0

How to rewrite "tu" in CANCEL message ?
by Julia 14 Feb '13

14 Feb '13

Hello all, Is there any way to rewrite "tu" in CANCEL message in 3.2.1? We use remove_hf / insert_hf for INVITE's "tu" modification before t_relay. It works fine, but it doesn't work for CANCEL (INVITE re-parsing is disabled). Ths,

8 19

Openfire and lync 2010 integration
by Vinay V 13 Feb '13

13 Feb '13

Hi All, I am in process of integrating lync 2010 with openfire xmpp server. Reading some of the blogs on kamailio found that it can be done using kamailio. Hence request if some body has done this please your inputs. Regards, Vinay

1 1

Re: [SR-Users] kamctl restart and mediaproxy
by Mino Haluz 13 Feb '13

13 Feb '13

You have to figure it somehow with adjusting the python code of media-dispatcher. We have developed special "/etc/init.d/kamailio grace" command, which waits for the moment, when there are no calls, so it can be restarted. It queries mediaproxy every second. In future we will not use mediaproxy anymore - it is not supported in latest version either. rtpproxy gives us better performance results. On Tue, Feb 12, 2013 at 2:44 PM, Eduardo Lejarreta <lejarreta.e(a)sarenet.es>wrote: > Good evening. > > Launching "mediaproxy-dispatcher" and "mediaproxy-relay" on the foreground > and with "DEBUG" we can see: > > mediaproxy-dispatcher log: > ------------------------------------------------------------------> Kamctl > restart > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Issuing "remove" command to relay at A.B.C.D > debug: Got statistics: {xxxxxxxxxxxxxxxxxxx} > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > > mediaproxy-relay log: > ------------------------------------------------------------------> Kamctl > restart > debug: removing session xxxxxxxxxxxxxxxxxxx > (Port 50000 Closed) > (Port 50001 Closed) > (Port 50002 Closed) > (Port 50003 Closed) > > So "mediaproxy-dispatcher" doesn't receive any command from Kamailio, he > just detects " Connection was closed cleanly" and then issues a "remove" > command to "mediaproxy-relay". > > Does anybody know any way to avoid this? > > Thanks and regards. > -- > Eduardo Lejarreta. > > > _______________________________________________ > 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

hadnling the IP extension calls freeing the PBX channel
by kiran bhosale 13 Feb '13

13 Feb '13

hello , w have PBX gateway along with kamailio. all calls are routed to pbx extensions. but if it happens to be an IP extrension we want the kamailio to handle the speech without keeping the PBX channel busy. if its an analog extension then this channel can be busy but if its IP extension kamailio should handle the speech itself. how is this possible ?? thanks in advance.

1 0

pseudovariable values written from app_lua sometimes appear incorrect
by elactrum＠jamailca.com 12 Feb '13

12 Feb '13

Hi, I am using app_lua to frequently store some data in htable and avp entries, using sr.pv.sets. When I read the values using sr.pv.get immediately after writing them, they seem correct. However, when I read them later from Lua, I have noticed that sometimes the value is incorrect or nil. Am I doing something wrong? The following code uses rtimer to duplicate the problem on startup, printing the errors to syslog. Tested on version: kamailio 3.3.1 (i386/linux) 2d0f6f ############### # kamailio.cfg ############### loadmodule "pv.so" loadmodule "xlog.so" loadmodule "htable.so" loadmodule "rtimer.so" loadmodule "app_lua" modparam("htable", "htable", "status=>size=14;autoexpire=0;") modparam("rtimer", "timer", "name=ta;interval=10;mode=1") modparam("rtimer", "exec", "timer=ta;route=LUA_TEST") # the following path may need to change depending on filesystem setup modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") route { route(LUA_TEST); } route[LUA_TEST] { $var(lua_test_key) = $(pp{s.toupper}) + "::"; lua_runstring("htable_test([[status=>$var(lua_test_key)]])"); lua_runstring("avp_test([[status=>$var(lua_test_key)]])"); } ---------------- -- kamailio.lua ---------------- -- Test of sr.pv operations in Lua function htable_test (namekey) local pvar_prefix = "$sht("..namekey pvar_test(pvar_prefix) end function avp_test ( name ) local pvar_prefix = "$avp("..name pvar_test(pvar_prefix) end function pvar_test ( pvar_prefix ) for value = 10000, 10050 do local pvar = pvar_prefix..value..")" --sr.log("info", "Lua attempting to set '"..pvar.."' to value '"..value.."'") sr.pv.sets(pvar, value) end for value = 10000, 10050 do -- now we check what was stored local pvar = pvar_prefix..value..")" local checkvar = sr.pv.get(pvar) if ( tostring(checkvar) ~= tostring(value) ) then sr.err(string.format("CHECK: Lua checking pvar '%s' and got value '%s' which is NOT '%s'", tostring(pvar), tostring(checkvar), tostring(value) )) end end end -- end of kamailio.lua Thanks, -Andrew

2 6

DNS error
by Vitaliy Aleksandrov 12 Feb '13

12 Feb '13

Hi, all Have a question about failure_routes. As I remember if kamailio statefuly sent a request and didn't receive a reply script writer can handle this situation at a failure_route[] block. But if kamailio can't resolve a destination from R-URI it sends back 478 reply and finishes current transaction without reaching a failure_route[]. Is it possible to process such a request in a failure_route[] ? Additional details: version: kamailio-3.2.4 t_on_failure() is called before t_relay() I have checked with REGISTER and INVITE transactions.

1 0

Re: [SR-Users] sr-users Digest, Vol 93, Issue 13
by madhumanjusha＠integramicro.com 12 Feb '13

12 Feb '13

Yes,Yes I did it the way tm and mediaproxy modules are added... when i check /usr/local/lib/kamailio/modules I found my module and $stat module time is effected. 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 >> >> >> > seems that you want to create your custom module. have you checked the > makefile of your module. in order to develop the custom module just copy > the existing module and make relevant changes there. this is genuine way . > for this u need to include makefile.lst ,Makefile in your module. so best > way is to copy the smallest module (e.g.) pdb module and make changes. > On Fri, February 8, 2013 3:36 am, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Reg. shared library linking during compiling time > (madhumanjusha(a)integramicro.com) > 2. Re: Reg. shared library linking during compiling time > (kiran bhosale) > 3. Re: Multiple radius responses on a missed call > (Daniel-Constantin Mierla) > 4. Re: Reg. shared library linking during compiling time > (Daniel-Constantin Mierla) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Fri, 8 Feb 2013 01:46:58 -0500 (EST) > From: madhumanjusha(a)integramicro.com > Subject: [SR-Users] Reg. shared library linking during compiling time > To: sr-users(a)lists.sip-router.org > Message-ID: > <38400.61.8.152.138.1360306018.squirrel(a)mail.integramicro.com> > Content-Type: text/plain;charset=iso-8859-1 > > > > Hello All, > Please find below steps...and kindly let me know the error solution. > > > STEP 1:: > Under modules I created a directory name b2bentities > /root/kamailio-3.3.2/modules/b2bentities > $ls > b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o > dialog.h librpath.lst makecfg.lst Makefile > > In b2b_server.h > #ifndef SERVER_H > #define SERVER_H > #endif > > > int b2b_dlg_reply(); > > > > In b2b_server.c I wrote following code :: > > > int b2b_dlg_reply() { > LOG(L_INFO,"Hello am in b2bua module\n"); > return 0; } > > > Also,I added below code to the same file as per steps given in kamailio > server's developer's guide. In developer's guide,they told to initialize > parameters as per module's requirement but i made them all empty and > null... for example static int mod_init(void) { > LOG(L_INFO,"IN mod_init\n"); > return 0; } > > > static cmd_export_t commands[] = { }; > > > > static param_export_t params[]={ }; > > > struct module_exports exports = { "b2bentities", // module name > NULL, // dlopen flags > commands, // exported functions params, // exported parameters > NULL, // exported statistics > NULL, // exported MI functions > NULL, // exported pseudo-variables > NULL, // extra processes > mod_init, // module init function (before fork. kids will inherit) > NULL, // reply processing function > NULL, // destroy function > child_init // child init function }; > > > > STEP 2:: > In kamailio core > file : In receive.c Method Name : In receive_msg > > > i did this way... > > In receive.c add follow piece of code:: > > > #include "modules/b2bentities/b2b_server.h" > > > int receive_msg(....) { > /*my code*/ > int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; > if(msg->REQ_METHOD!=METHOD_REGISTER) { > > > b2b_ret=(*ptr)(); if(b2b_ret!=0) { > LOG(L_INFO,"B2B Dialogue reply function failed\n"); > exit(1); } > else { > LOG(L_INFO,"B2B Dialogue reply function success\n"); > exit(1); } > > > } > > > > } > > > STEP 3:: > I even added kamailio.cfg load module "b2bentities.so" > > > > > STEP 4:: > Final step,when I do make all..... > its giving me the following error :: [root@kamserver kamailio-3.3.2]# make > all CC (gcc) [kamailio] receive.o > LD (gcc) [kamailio] kamailio > receive.o: In function `receive_msg': > /root/kamailio-3.3.2/receive.c:163: undefined reference to `b2b_dlg_reply' > collect2: ld returned 1 exit status > make: *** [kamailio] Error 1 > > > > > My doubt: > Where should my b2bentities.so should be linked during compiling time? > Please tell me the steps for it. > > > Thanks & Regards, > Manjusha A. > Integra Micro Software Services (P) Ltd. > > > > > > > > > > > ------------------------------ > > > Message: 2 > Date: Fri, 08 Feb 2013 12:36:36 +0530 > From: kiran bhosale <kbhosale(a)synteltelecom.com> > Subject: Re: [SR-Users] Reg. shared library linking during compiling > time To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114A3FC.9000902(a)synteltelecom.com> > Content-Type: text/plain; charset=us-ascii; format=flowed > > > madhumanjusha(a)integramicro.com wrote: >> Hello All, >> Please find below steps...and kindly let me know the error solution. >> >> >> STEP 1:: >> Under modules I created a directory name b2bentities >> /root/kamailio-3.3.2/modules/b2bentities >> $ls >> b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o >> dialog.h librpath.lst makecfg.lst Makefile >> >> In b2b_server.h >> #ifndef SERVER_H >> #define SERVER_H >> #endif >> >> >> int b2b_dlg_reply(); >> >> >> >> In b2b_server.c I wrote following code :: >> >> >> int b2b_dlg_reply() { >> LOG(L_INFO,"Hello am in b2bua module\n"); >> return 0; } >> >> >> Also,I added below code to the same file as per steps given in kamailio >> server's developer's guide. In developer's guide,they told to >> initialize parameters as per module's requirement but i made them all >> empty and null... for example static int mod_init(void) { >> LOG(L_INFO,"IN mod_init\n"); >> return 0; } >> >> >> static cmd_export_t commands[] = { }; >> >> >> >> static param_export_t params[]={ }; >> >> >> struct module_exports exports = { "b2bentities", // module name >> NULL, // dlopen flags >> commands, // exported functions params, // exported >> parameters NULL, // exported statistics >> NULL, // exported MI functions >> NULL, // exported pseudo-variables >> NULL, // extra processes >> mod_init, // module init function (before fork. kids will >> inherit) NULL, // reply processing function >> NULL, // destroy function >> child_init // child init function }; >> >> >> >> STEP 2:: >> In kamailio core >> file : In receive.c Method Name : In receive_msg >> >> >> i did this way... >> >> In receive.c add follow piece of code:: >> >> >> #include "modules/b2bentities/b2b_server.h" >> >> >> int receive_msg(....) { >> /*my code*/ >> int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; >> if(msg->REQ_METHOD!=METHOD_REGISTER) { >> >> >> b2b_ret=(*ptr)(); if(b2b_ret!=0) { >> LOG(L_INFO,"B2B Dialogue reply function failed\n"); >> exit(1); } >> else { >> LOG(L_INFO,"B2B Dialogue reply function success\n"); >> exit(1); } >> >> >> } >> >> >> >> } >> >> >> STEP 3:: >> I even added kamailio.cfg load module "b2bentities.so" >> >> >> >> >> STEP 4:: >> Final step,when I do make all..... >> its giving me the following error :: [root@kamserver kamailio-3.3.2]# >> make all CC (gcc) [kamailio] receive.o >> LD (gcc) [kamailio] kamailio >> receive.o: In function `receive_msg': >> /root/kamailio-3.3.2/receive.c:163: undefined reference to >> `b2b_dlg_reply' >> collect2: ld returned 1 exit status >> make: *** [kamailio] Error 1 >> >> >> >> >> My doubt: >> Where should my b2bentities.so should be linked during compiling time? >> Please tell me the steps for it. >> >> >> Thanks & Regards, >> Manjusha A. >> Integra Micro Software Services (P) Ltd. >> >> >> >> >> >> >> >> >> _______________________________________________ >> 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 >> >> >> > seems that you want to create your custom module. have you checked the > makefile of your module. in order to develop the custom module just copy > the existing module and make relevant changes there. this is genuine way . > for this u need to include makefile.lst ,Makefile in your module. so best > way is to copy the smallest module (e.g.) pdb module and make changes. > > > > > ------------------------------ > > > Message: 3 > Date: Fri, 08 Feb 2013 09:27:07 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Multiple radius responses on a missed call > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114B6DB.10605(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Hello, > > > there are different event logged for accounting: - missed call event which > is sent from the point of view of callee - transaction answered which is > sent from the point of view of caller > > In db (eg mysql) accounting, each event in stored in a different table. > I don't know how they are in radius. You probably get two missed call > events and one transaction answered. > > If you want only one, do not set the flag for missed calls. > > > Cheers, > Daniel > > > On 2/7/13 9:23 PM, Efelin Novak wrote: > >> Hi folks, >> >> >> I'd like to ask you how can I solve an issue when I'm receiving three >> RADIUS requests on one missed call. My scenario is as follows: >> >> >> A calls to B. B returns 503 error message, INIVITE is then sent to C >> and C sends 486. >> >> After first 503 I receive a first RADIUS request. This is correct. >> >> >> However after 486 I receive three RADIUS requests with different IDs. >> As you can see in a DEBUG one RADIUS is send after the SIP reply >> message is received and second and third RADIUS messages are sent after >> the SIP reply message is forwarded. >> >> I'd like to receive only two RADIUS messages. One after 503 and second >> after 486. >> >> >> FLT_ACC is defined as 1. My config is as follows: >> modparam("acc", "failed_transaction_flag", FLT_ACC) modparam("acc", >> "report_cancels", 1) >> modparam("acc", "report_ack", 0) modparam("acc", >> "detect_direction", 0) >> modparam("acc", "early_media", 0) modparam("acc", "log_level", >> 5) >> modparam("acc", "log_flag", 1) modparam("acc", >> "log_missed_flag", 1) >> modparam("acc_radius", "radius_config", >> "/etc/radiusclient-ng/radiusclient.conf") # This is the >> location of the configuration file of radius client >> modparam("acc_radius", "radius_flag", FLT_ACC) >> modparam("acc_radius", "radius_missed_flag", FLT_ACC) >> >> In a route I set setflag("FLT_ACC") only once per call. >> >> >> Here is the debug for debug level 7: >> 0(5890) DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): >> 0(5890) DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> >> 0(5890) DEBUG: <core> [parser/msg_parser.c:638]: status: <486> >> 0(5890) DEBUG: <core> [parser/msg_parser.c:640]: reason: <Busy here> >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 232, <branch> = <z9hG4bK8d9c.4438d232.1>; state=6 >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 234, <received> = <192.168.21.101>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=2 0(5890) DEBUG: <core> [parser/msg_parser.c:513]: >> parse_headers: this >> is the first via 0(5890) DEBUG: <core> [receive.c:149]: After >> parse_msg... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: >> msg id=6 global id=5 T start=0xffffffff 0(5890) DEBUG: <core> >> [parser/parse_via.c:1286]: Found param type >> 232, <branch> = <z9hG4bK8d9c.e7b650e1.0>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=22 0(5890) DEBUG: <core> [parser/msg_parser.c:524]: >> parse_headers: this >> is the second via 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found >> param type 232, <branch> = <z9hG4bK463ae086>; state=6 >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 235, <rport> = <5060>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=22 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: >> add_param: tag=as493d6134 >> 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, >> state=29 0(5890) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: >> get_hdr_field: <To> [42]; uri=[sip:5556003900000@dr.vm] >> 0(5890) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body >> [<sip:5556003900000@dr.vm>] >> 0(5890) DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq >> <CSeq>: <102> <INVITE> >> 0(5890) DEBUG: tm [t_lookup.c:965]: DEBUG: t_reply_matching: hash >> 51672 label 590185284 branch 1 >> 0(5890) DEBUG: tm [t_lookup.c:1016]: DEBUG: t_reply_matching: reply >> matched (T=0x871afc98)! 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: >> trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: acc >> [acc_logic.c:537]: acc callback called for >> t(0x871afc98) event type 2, reply code 486 0(5890) DEBUG: tm >> [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 2, id 0 entered 0(5890) DEBUG: <core> [parser/parse_to.c:178]: >> DEBUG: add_param: tag=as5c1d222e >> 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, >> state=29 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... >> 0(5890) DEBUG: tm [t_lookup.c:1148]: DEBUG: t_check_msg: msg id=6 >> global id=6 T end=0x871afc98 0(5890) DEBUG: tm [t_reply.c:2081]: DEBUG: >> reply_received: org. >> status uas=180, uac[1]=180 local=0 is_invite=1) 0(5890) DEBUG: tm >> [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 4194304, id 0 entered 0(5890) DEBUG: siptrace [siptrace.c:794]: >> storing info... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: >> msg id=6 global id=6 T start=0x871afc98 0(5890) DEBUG: tm >> [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! >> 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> >> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [REPLY_TWO] Incoming reply code: 486 0(5890) DEBUG: tm >> [t_reply.c:1204]: ->>>>>>>>> T_code=180, new_code=486 >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 128, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback >> called for t(0x871afc98) event type 128, reply code 486 0(5890) acc >> [acc.c:275]: ACC: call missed: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius >> 0(5890) DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: >> 6e1b9f35a8b3e65859e29aefc1c26fc3 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=262 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_TYPE=application/sdp >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_SUPPORTED=replaces >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_ALLOW=INVITE, ACK, >> CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_DATE=Thu, 07 Feb 2013 >> 19:35:53 GMT >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_MAX_FORWARDS=68 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_USER_AGENT=Asterisk PBX >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=102 INVITE >> 0(5890) DEBUG: exec [exec_hf.c:267]: >> SIP_HF_CALLID=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> 0(5890) DEBUG: exec [exec_hf.c:267]: >> SIP_HF_CONTACT=<sip:0221028400@192.168.21.106> >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<sip:5556003900000@dr.vm> >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM="Janko" >> <sip:0221028400@dr.vm>;tag=as5c1d222e >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP >> 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0,SIP/2.0/UDP >> 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 >> 0(5890) DEBUG: exec [exec_mod.c:175]: executing >> [/opt/kamrouter/scripts/notify_reroute.sh 503 >> sip:3333003900000@ast-car.vm sip:0221028400@dr.vm >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm] >> sh: /opt/kamrouter/scripts/notify_reroute.sh: not found >> 0(5890) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on >> 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=0 >> global id=0 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: >> DEBUG: t_check_msg: T already found! >> 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> >> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [FAIL_TWO] Rerouted gw returned 486 0(5890) INFO: <script>: XLOG: >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [aaa] VYMAZAVAM3 0(5890) DEBUG: >> tm [t_reply.c:1702]: DEBUG: relay_reply: branch=1, save=0, relay=1 >> 0(5890) DEBUG: <core> [msg_translator.c:1933]: old size: 572, new >> size: 485 >> 0(5890) DEBUG: <core> [msg_translator.c:1951]: >> build_res_from_sip_res: copied size: orig:110, new: 23, rest: 462 msg= >> SIP/2.0 486 Busy here >> Via: SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0 >> Via: SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 >> From: "Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e >> To: <sip:5556003900000@dr.vm>;tag=as493d6134 >> Call-ID: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> CSeq: 102 INVITE >> Server: Asterisk PBX 11.0.2 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO, PUBLISH >> Supported: replaces, timer >> Content-Length: 0 >> >> >> >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 1048576, id 0 entered 0(5890) DEBUG: dialog [dlg_hash.c:588]: ref >> dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:590]: >> dialog id=3894 found on entry 2825 0(5890) DEBUG: dialog >> [dlg_hash.c:940]: dialog 0x871af3f0 changed >> from state 2 to state 5, due event 4 (ref 2) 0(5890) DEBUG: dialog >> [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 3 >> 0(5890) DEBUG: dialog [dlg_handlers.c:1469]: executing event_route 4 on >> state 5 0(5890) INFO: <script>: XLOG: >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [dialog:failed] Deleting 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm from >> dialogs and dstprefix_limit 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref >> dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_handlers.c:543]: >> dialog 0x871af3f0 failed (negative reply) >> 0(5890) DEBUG: dialog [dlg_cb.c:256]: dialog=0x871af3f0, type=4 >> 0(5890) DEBUG: dialog [dlg_hash.c:643]: ref dlg 0x871af3f0 with 1 -> 3 >> 0(5890) DEBUG: dialog [dlg_hash.c:646]: dialog >> callid='38e422e10af08b2e34d58e5b765e81d3(a)dr.vm' found on entry 2825, >> dir=1 0(5890) DEBUG: dialog [dlg_var.c:203]: Internal var-list ((nil)): >> 0(5890) DEBUG: dialog [dlg_var.c:213]: Dialog var-list (0x871af6f8): >> 0(5890) DEBUG: dialog [dlg_var.c:218]: "ru"=sip:003900000@cr1.vm:5060 >> (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: d=0 (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: et=1360265682.889 (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: st=1360265682.889 (flags 1) >> 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 >> 0(5890) acc [acc_cdr.c:192]: st=1360265682.889; et=1360265682.889; >> d=0; c1= 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 >> with 1 -> 1 0(5890) DEBUG: dialog [dlg_handlers.c:937]: registering TMCB >> to wait for negative ACK 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg >> 0x871af3f0 with 1 -> 2 >> 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 512, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback >> called for t(0x871afc98) event type 512, reply code 486 0(5890) acc >> [acc.c:275]: ACC: call missed: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius >> 0(5890) acc [acc.c:275]: ACC: transaction answered: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:867]: acc event for engine: radius >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 8388608, id 0 entered >> >> Thanks for a help. >> >> >> Efelin >> >> >> _______________________________________________ >> 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 > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > > ------------------------------ > > > Message: 4 > Date: Fri, 08 Feb 2013 09:36:01 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Reg. shared library linking during compiling > time To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114B8F1.8020807(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Hello, > > > some guidelines to write a new module are available at: - > http://www.asipto.com/pub/kamailio-devel-guide/#c16_devel_new_module > > > Cheers, > Daniel > > > On 2/8/13 8:06 AM, kiran bhosale wrote: > >> madhumanjusha(a)integramicro.com wrote: >>> Hello All, >>> Please find below steps...and kindly let me know the error solution. >>> >>> >>> STEP 1:: >>> Under modules I created a directory name b2bentities >>> /root/kamailio-3.3.2/modules/b2bentities >>> $ls >>> b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o >>> dialog.h librpath.lst makecfg.lst Makefile >>> >>> In b2b_server.h >>> #ifndef SERVER_H >>> #define SERVER_H >>> #endif >>> >>> >>> int b2b_dlg_reply(); >>> >>> >>> >>> In b2b_server.c I wrote following code :: >>> >>> >>> int b2b_dlg_reply() { >>> LOG(L_INFO,"Hello am in b2bua module\n"); >>> return 0; } >>> >>> >>> Also,I added below code to the same file as per steps given in >>> kamailio server's developer's guide. In developer's guide,they told to >>> initialize parameters as per module's requirement but i made them all >>> empty and null... for example static int mod_init(void) { >>> LOG(L_INFO,"IN mod_init\n"); >>> return 0; } >>> >>> >>> static cmd_export_t commands[] = { }; >>> >>> >>> >>> static param_export_t params[]={ }; >>> >>> >>> struct module_exports exports = { "b2bentities", // module name >>> NULL, // dlopen flags >>> commands, // exported functions params, // exported >>> parameters NULL, // exported statistics >>> NULL, // exported MI functions >>> NULL, // exported pseudo-variables >>> NULL, // extra processes >>> mod_init, // module init function (before fork. kids will >>> inherit) NULL, // reply processing function >>> NULL, // destroy function >>> child_init // child init function }; >>> >>> >>> >>> STEP 2:: >>> In kamailio core >>> file : In receive.c Method Name : In receive_msg >>> >>> >>> i did this way... >>> >>> In receive.c add follow piece of code:: >>> >>> >>> #include "modules/b2bentities/b2b_server.h" >>> >>> >>> int receive_msg(....) { >>> /*my code*/ >>> int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; >>> if(msg->REQ_METHOD!=METHOD_REGISTER) { >>> >>> >>> b2b_ret=(*ptr)(); if(b2b_ret!=0) { >>> LOG(L_INFO,"B2B Dialogue reply function >>> failed\n"); exit(1); } >>> else { >>> LOG(L_INFO,"B2B Dialogue reply function >>> success\n"); exit(1); } >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> STEP 3:: >>> I even added kamailio.cfg load module "b2bentities.so" >>> >>> >>> >>> >>> STEP 4:: >>> Final step,when I do make all..... >>> its giving me the following error :: [root@kamserver kamailio-3.3.2]# >>> make all CC (gcc) [kamailio] receive.o >>> LD (gcc) [kamailio] kamailio >>> receive.o: In function `receive_msg': >>> /root/kamailio-3.3.2/receive.c:163: undefined reference to >>> `b2b_dlg_reply' >>> collect2: ld returned 1 exit status >>> make: *** [kamailio] Error 1 >>> >>> >>> >>> >>> My doubt: >>> Where should my b2bentities.so should be linked during compiling time? >>> Please tell me the steps for it. >>> >>> >>> Thanks & Regards, >>> Manjusha A. >>> Integra Micro Software Services (P) Ltd. >>> >>> >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> seems that you want to create your custom module. have you checked the >> makefile of your module. in order to develop the custom module just copy >> the existing module and make relevant changes there. this is genuine way >> . for this u need to include makefile.lst ,Makefile in >> your module. so best way is to copy the smallest module (e.g.) pdb module >> and make changes. >> >> >> _______________________________________________ >> 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 > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > > ------------------------------ > > > _______________________________________________ > sr-users mailing list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > End of sr-users Digest, Vol 93, Issue 13 > **************************************** > > Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 1

Re: [SR-Users] [sr-dev] openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Hi Shahzad , It is not ,We can able to connect remote mysql ( please find the detail below),moreover we can able create openser db from Kamailio server to remote mysql server *Mysql server : 192.168.1.170* *Kamailio server : 192.168.1.101* *Mysql output :* * * *mysql -u openser -h 192.168.1.170 -p* Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 68 Server version: 5.1.67 Source distribution Copyright (c) 2000, 2012, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql> With Regards N.Prakash > > ---------- Forwarded message ---------- > From: Muhammad Shahzad <shaheryarkh(a)gmail.com> > Date: Tue, Feb 12, 2013 at 5:16 PM > Subject: Re: [sr-dev] [SR-Users] openser db on an external mysql server > To: Development mailing list of the sip-router project < > sr-dev(a)lists.sip-router.org> > Cc: Daniel-Constantin Mierla <miconda(a)gmail.com>, > sr-users(a)lists.sip-router.org, Indu Priya <indu(a)tevatel.com>, Prabhakaran > B <prabhakaran(a)tevatel.com> > > > Its not a Kamailio error, it is problem with connecting to mysql server. > Error 4 means No MySQL Server Running On Specified Host. > > May be MySQL server is down or there is some firewall running or MySQL not > listening on IP you specified. > > Thank you. > > > On Tue, Feb 12, 2013 at 12:11 PM, Prakash N <prakash.n(a)tevatel.com> wrote: > >> Dear Daniel, >> >> Thanks for your quick response >> >> We can able to create operseb db on remote mysql server ,but >> Kamailio not getting start >> >> Error detail: >> >> ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL >> server on '192.168.1.170' (4) >> ERROR: <core> [db.c:32 0]: could not add connection to the pool >> ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database >> ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while >> initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) >> >> >> I have attached my kamailo.cfg ( we followed Kamailio and Asterisk real >> time installation) ,please advice >> >> With Regards >> >> N.Prakash >> >>> >>> ---------- Forwarded message ---------- >>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>> Date: Tue, Feb 12, 2013 at 2:41 PM >>> Subject: Re: [SR-Users] openser db on an external mysql server >>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>> Users Mailing List" <sr-users(a)lists.sip-router.org>, >>> sr-dev(a)lists.sip-router.org, miconda(a)gmail.com >>> Cc: Prabhakaran B <prabhakaran(a)tevatel.com> >>> >>> >>> Hello, >>> >>> kamailio database is created with: >>> >>> kamdbctl create >>> >>> the host of mysql server has to be specified in kamctlrc file via DBHOST >>> variable - kamctlrc is located in same directory as kamailio.cfg. >>> >>> In kamailio.cfg, update the db_url value (or DBURL define) to reflect >>> the mysql server host. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 2/12/13 4:16 AM, Prakash N wrote: >>> >>> Hi All, >>> >>> I am new this Kamailo ,ther is any step by step document to set >>> up openser db on an external mysql server ? >>> >>> With Regards >>> >>> N.Prakash >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio World Conference, April 16-17, 2013, Berlin >>> - http://conference.kamailio.com - >>> >>> >>> >> >> _______________________________________________ >> sr-dev mailing list >> sr-dev(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk(a)hotmail.com > Email: shaheryarkh(a)googlemail.com > >

3 3

Re: [SR-Users] openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Dear Daniel, Thanks for your quick response We can able to create operseb db on remote mysql server ,but Kamailio not getting start Error detail: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) ERROR: <core> [db.c:32 0]: could not add connection to the pool ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) I have attached my kamailo.cfg ( we followed Kamailio and Asterisk real time installation) ,please advice With Regards N.Prakash > > ---------- Forwarded message ---------- > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Date: Tue, Feb 12, 2013 at 2:41 PM > Subject: Re: [SR-Users] openser db on an external mysql server > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users > Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev(a)lists.sip-router.org, > miconda(a)gmail.com > Cc: Prabhakaran B <prabhakaran(a)tevatel.com> > > > Hello, > > kamailio database is created with: > > kamdbctl create > > the host of mysql server has to be specified in kamctlrc file via DBHOST > variable - kamctlrc is located in same directory as kamailio.cfg. > > In kamailio.cfg, update the db_url value (or DBURL define) to reflect the > mysql server host. > > Cheers, > Daniel > > > On 2/12/13 4:16 AM, Prakash N wrote: > > Hi All, > > I am new this Kamailo ,ther is any step by step document to set > up openser db on an external mysql server ? > > With Regards > > N.Prakash > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > >

2 1

Fwd: INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers
by Bayram Karagoz 12 Feb '13

12 Feb '13

---------- Forwarded message ---------- From: Bayram Karagoz <karagoz.bayram(a)gmail.com> Date: 2013/2/12 Subject: Re: [SR-Users] INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers To: miconda(a)gmail.com Cc: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> recheck config and saw that I used dlg_manage() most before route(QUOTA) block. does it sits in the wrong place? you can find my kamailio.cfg file in attachment also. # account only INVITEs * dlg_manage();* if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } ###Display Name manipulation### if (is_method("INVITE") && ($fn != $fU)) { uac_replace_from("$fU",""); } ### incoming call from SBC### route(INCOMINGSBC); ###Suspend User### route(SUSPEND); ### Concurrent Call Limitation Routing Block ### * route(QUOTA);* 2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com> > Where do you use dlg_manage()? It is not in the route you pasted below. > > Cheers, > Daniel > > > On 2/12/13 10:12 AM, Bayram Karagoz wrote: > > Thanks for quick response Daniel, > > My kamailio version is 3.2.3 (x86_64/linux) . > I want to use dlg_manage() before but not sure which side of my call > limitation route code block. > can you advise me where to use dlg_manage() in my routing? > here is my route block defined below; > > > route[QUOTA] { > > if (is_method("INVITE") && $fU == $avp(s:username)) { > > avp_db_query("select quota from subscriber where > username='$avp(s:username)'", "$avp(s:quota)"); > > $var(SIZE) = 0; > get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); > xlog("L_INFO", "CALL_KAMAILIO:Call originated from = > $fd\n"); > xlog("L_INFO", "Active call amount for $fU is $var(SIZE) , > call-id=$ci, quota=$avp(s:quota)\n"); > if( $var(SIZE) >= $avp(s:quota) ){ > sl_send_reply("403", "Simultaneous calls limit reached"); > xlog("L_INFO", "QUOTA:Limit of $var(SIZE) active calls has > been reached for user=$avp(s:username), call-id=$ci, > quota=$avp(s:quota)\n"); > exit; > } > set_dlg_profile("callquota","$avp(s:username)"); > } > } > > > 2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com> > >> Hello, >> >> what version of kamailio are you using? >> >> State 1 is for dialogs that were not relayed (or they haven't received >> any reply yet). That means you create the dialog and don't relay the >> INVITE. Try to us dlg_manage() just before relaying the INVITE. >> >> Cheers, >> Daniel >> >> >> On 2/12/13 9:37 AM, Bayram Karagoz wrote: >> >> Hi, >> >> is there any advise about my problem? >> I'm using kamailio in my production environment and this issue started to >> reproduce a lot. >> is there any way to discard retransmitted INVITE messages from counting >> dialog? >> >> >> 2013/2/9 Bayram Karagoz <karagoz.bayram(a)gmail.com> >> >>> thanks for the response Daniel, >>> >>> has_totag not solved my problem. >>> >>> in a deep search I realized that these retransmission INVITE dialogs >>> are waiting in dialog list as state = 1. one example is below; >>> is there any way to clear these dialog which state is waiting for state >>> = 1 or any other solution is welcome for me? >>> >>> dialog:: hash=246:499719317 >>> state:: 1 >>> ref_count:: 1 >>> timestart:: 0 >>> timeout:: 0 >>> callid:: 1356619-3569384775-474862(a)GBSBC-1.mydomain.com >>> from_uri:: sip:02165151057@xxx.xxx.xxx.xxx >>> from_tag:: 3569384775-474868 >>> caller_contact:: sip:02165151057@xxx.xxx.xxx.xxx:5060 >>> caller_cseq:: 2 >>> caller_route_set:: >>> caller_bind_addr:: udp:xxx.xxx.xxx.xxx:5060 >>> callee_bind_addr:: >>> to_uri:: sip:4115667@xxx.xxx.xxx.xxx >>> to_tag:: >>> callee_contact:: >>> callee_cseq:: >>> callee_route_set:: >>> >>> >>> >>> >>> 2013/2/4 Daniel Tryba <daniel(a)pocos.nl> >>> >>>> On Sunday 03 February 2013 12:58:02 Bayram Karagoz wrote: >>>> > I'm trying to limit all subscriber's active call according to setting >>>> their >>>> > simultaneous use setting defined in database. >>>> > But having problem when one subscriber's INVITE message goes to >>>> > retransmission. When retranmission occures INVITE messages don't >>>> complete >>>> > with BYE and subscriber simultaneus use amount will hung. >>>> > My example code block is below. >>>> > Does anyone advice me for overcome INVITE retransmissions? >>>> >>>> I haven't implemented it myself yet, but all examples I've seen check >>>> for >>>> !has_totag() in the INVITE when doing call limit accounting. So >>>> if (is_method("INVITE") && $fU == $avp(s:username) && !has_totag()) >>>> in your case. But that only works if the device received the initial 100 >>>> Trying with a tag from Kamailio. >>>> >>>> But might it not be better to just do the count in the INVITE and >>>> adding/deleting of dialogs in the dialog event routes? >>>> http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2556156 >>>> Problem here is that if the limit is 5 and and there are more than 5 >>>> concurrent INVITES you'd have to drop calls with a 200 OK response. >>>> >>>> -- >>>> >>>> POCOS B.V. - Croy 9c - 5653 LC Eindhoven >>>> Telefoon: 040 293 8661 - Fax: 040 293 8658 >>>> http://www.pocos.nl/ - http://www.sipo.nl/ >>>> K.v.K. Eindhoven 17097024 >>>> >>> >>> >>> >>> -- >>> *Bayram KARAGÖZ* >>> >>> *NGN Engineer* >>> >>> *-----------------------------------------* >>> >>> *@* bayram(a)bayramkaragoz.org >>> >>> *:** *www.bayramkaragoz.org >>> >>> *-----------------------------------------* >>> >> >> >> >> -- >> *Bayram KARAGÖZ* >> >> *NGN Engineer* >> >> *-----------------------------------------* >> >> *@* bayram(a)bayramkaragoz.org >> >> *:** *www.bayramkaragoz.org >> >> *-----------------------------------------* >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio World Conference, April 16-17, 2013, Berlin >> - http://conference.kamailio.com - >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > -- > *Bayram KARAGÖZ* > > *NGN Engineer* > > *-----------------------------------------* > > *@* bayram(a)bayramkaragoz.org > > *:** *www.bayramkaragoz.org > > *-----------------------------------------* > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > -- *Bayram KARAGÖZ* *NGN Engineer* *-----------------------------------------* *@* bayram(a)bayramkaragoz.org *:** *www.bayramkaragoz.org *-----------------------------------------* -- *Bayram KARAGÖZ* *NGN Engineer* *-----------------------------------------* *@* bayram(a)bayramkaragoz.org *:** *www.bayramkaragoz.org *-----------------------------------------*

2 2

Kamailio World Updates
by Daniel-Constantin Mierla 12 Feb '13

12 Feb '13

Hello, we have published the first set of accepted speeches at Kamailio World Conference, you can view the current draft of the schedule at: - http://conference.kamailio.com/k01/schedule/ There are already about 15 speakers and presentations in the agenda for the two days of conference, other proposals being in the review queue. Another good news is that business environment around Kamailio project is growing, the sponsors of Kamailio World Conference, Sipwise, Sipgate and Asipto are looking to hire or subcontract new talents, check the Career & Jobs page at: - http://conference.kamailio.com/k01/career-jobs/ Greetings again to our community members that participated at Fosdem, we were quite a big group there, unfortunately not much time to catch up in detail with everyone. Looking forward to see many of you in Berlin as well, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Handling 302 redirect on group ringing
by Mark Boyce 12 Feb '13

12 Feb '13

Hi All Got a problem handling redirects when more than one phone is ringing. The setup is this; Phone1 - Extension 100 Phone 2 - Extension 101 Phone 3 - Extension 102 & 101 When Phone 1 rings Extension 101 Phones 2 & 3 ring correctly as you would expect. If Phone 2 now sets the on phone forward/redirect we see the following; Phone 1 - Invite sent to 101 Phone 2 & 3 Receive Invites Phone 2 responds with a 302 Redirect Phone 3 Rings. Which again is as you would expect. The problem is that the 302 Message we're getting back is being seen in onreply_route where I cannot add a new branch to handle it. (Error - misused command append_branch / Command cannot be used in the block) If Phone 3 is unavailable / times out / busy / or otherwise does not take the call we then see the 302 from Phone 2 arrive in failure_route where it is handled as expected. The call flow we're trying to achieve is that the redirect is followed whilst the other available phone carries on ringing. I've tried with failure_reply_mode as the default 0 and as 3, both do the same thing. Currently working with kamailio version 3.4.0-dev6 Can anyone spot what I'm doing wrong ? Thanks Mark

2 1

How to detect there are no matching subscriber for uri
by Krishna Kurapati 12 Feb '13

12 Feb '13

Hi, When we receive an request, we do " lookup("location") call to find if the uri is there or not. There are two scenarios: 1. The aor is in the subscriber table but not in usrloc table 2. The aor is not in subscriber table I would like to respond differently to the sender in each of these cases. Right now, the lookup returns -1 for either case. Is there any other way other than dipping into database each time? Thanks Krish Kura

2 1

INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers
by Bayram Karagoz 12 Feb '13

12 Feb '13

Hello, I'm trying to limit all subscriber's active call according to setting their simultaneous use setting defined in database. But having problem when one subscriber's INVITE message goes to retransmission. When retranmission occures INVITE messages don't complete with BYE and subscriber simultaneus use amount will hung. My example code block is below. Does anyone advice me for overcome INVITE retransmissions? Thanks. modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) modparam("dialog","profiles_with_value","callquota") modparam("dialog", "default_timeout", 3600) modparam("dialog", "dlg_match_mode",0) modparam("dialog", "detect_spirals", 1) route[QUOTA] { if (is_method("INVITE") && $fU == $avp(s:username)) { avp_db_query("select quota from subscriber where username='$avp(s:username)'", "$avp(s:quota)"); $var(SIZE) = 0; get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); xlog("L_INFO", "CALL_OPENSER:Call originated from = $fd\n"); xlog("L_INFO", "Active call amount for $fU is $var(SIZE) , quota=$avp(s:quota)\n"); if( $var(SIZE) >= $avp(s:quota) ){ sl_send_reply("403", "Simultaneous calls limit reached"); xlog("L_INFO", "QUOTA:Limit of $var(SIZE) active calls has been reached for user=$avp(s:username), quota=$avp(s:quota)\n"); exit; } set_dlg_profile("callquota","$avp(s:username)"); get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); } } -- *Bayram KARAGÖZ*

3 6

Video Conferencing
by Mohakem Khan 12 Feb '13

12 Feb '13

Does kamailio support video endpoint/codecs? How about video conference? Is there MCD support?

3 2

openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Hi All, I am new this Kamailo ,ther is any step by step document to set up openser db on an external mysql server ? With Regards N.Prakash

2 1

mysql: driver_error
by Uri Shacked 11 Feb '13

11 Feb '13

Hi, I am doing some testing and i have trouble understanding when does the "mysql:driver_error" counter/statistics increase.... I stopped the mysqld process on the DB server and used sqlops. the driver_error did still is on 0. I use version 3.3.2. Thanks, Uri

2 3

Re: [SR-Users] dlg_manage and engage_media_proxy
by Daniel-Constantin Mierla 11 Feb '13

11 Feb '13

Hello, On 2/11/13 5:17 PM, Eduardo Lejarreta wrote: > Good evening Daniel. > > You are right, calling engage_media_proxy() prior to dlg_manage() solves the > issue (just before t_relay on INVITEs). > > I've done several test changing the order of things and now everything works > fine. great, thanks for reporting back. Perhaps dlg flag vs dlg_manage has to be clarified a bit more in the next major release (4.1). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio multiple registration issue
by Ramaseshi reddy kolli 11 Feb '13

11 Feb '13

Hi, i am using kamailio 3.3 version please help me with this issue. 1) Registered from SIP end point, kamailio added row in location table 2) Now using same SIP account details, pushed registration entry using *kamctl ul add , * instead of overwriting the previous contact by new one it inserts another new row in the location table 3) Total now 2 rows with same username are present in DB. I am using save("location", "0x04") so before inserting new row it should check whether already entry is there or not, if there should update the same row. I look forward to your reply. Thanks Ram

2 2

Re: [SR-Users] dlg_manage and engage_media_proxy
by Daniel-Constantin Mierla 11 Feb '13

11 Feb '13

On 2/8/13 12:15 PM, Eduardo Lejarreta wrote: > Yes it's the correct order. We call dlg_manage() prior to > engage_media_proxy(): The correct order could be the other one, as mediaproxy might need some callbacks from dialog module. Usually the dlg_manage() should be executed just before relaying, when is no other chance that the invite will get a negative reply from the config. Cheers, Daniel > > route[INVITE_MANAGE] { > > if (is_method("INVITE")) { > > $dlg_var(si) = $si; > $dlg_var(tU) = $tU; > $dlg_ctx(timeout_bye) = 1; > > dlg_manage(); > ........ > ........ > } > } > ....... > ....... > route[RELAY] { > > if (is_method("INVITE")) { > ........ > ........ > route(MEDIAPROXY); # route MEDIAPROXY > t_on_failure(FAIL_ONE); > t_on_reply(REPLY_ONE); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > ........ > ........ > route[MEDIAPROXY] { > > xlog("L_INFO", "Engage Media Proxy - M=$rm R=$ru F=$fu T=$tu > IP=$si:$sp ID=$ci\n"); > engage_media_proxy(); > return; > } > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

UC Expo London
by Peter Dunkley 11 Feb '13

11 Feb '13

Hello, Are there any plans for a Kamailio meet-up on one of the evenings this year? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Kamailio-Java Integration
by ron.kamailio＠mcleodnet.com 10 Feb '13

10 Feb '13

I am looking for a way to integrate Java business logic with Kamailio to influence routing decisions, add/remove/modify headers, record accounting-related information, etc. A while back there was a project named WeSIP, but it doesn't appear to active anymore. Is anyone doing something like this now, or any ideas or suggestions? Thanks, Ron

8 23

Multiple radius responses on a missed call
by Efelin Novak 08 Feb '13

08 Feb '13

Hi folks, I'd like to ask you how can I solve an issue when I'm receiving three RADIUS requests on one missed call. My scenario is as follows: A calls to B. B returns 503 error message, INIVITE is then sent to C and C sends 486. After first 503 I receive a first RADIUS request. This is correct. However after 486 I receive three RADIUS requests with different IDs. As you can see in a DEBUG one RADIUS is send after the SIP reply message is received and second and third RADIUS messages are sent after the SIP reply message is forwarded. I'd like to receive only two RADIUS messages. One after 503 and second after 486. FLT_ACC is defined as 1. My config is as follows: modparam("acc", "failed_transaction_flag", FLT_ACC) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) modparam("acc", "detect_direction", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 5) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # This is the location of the configuration file of radius client modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACC) In a route I set setflag("FLT_ACC") only once per call. Here is the debug for debug level 7: 0(5890) DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): 0(5890) DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> 0(5890) DEBUG: <core> [parser/msg_parser.c:638]: status: <486> 0(5890) DEBUG: <core> [parser/msg_parser.c:640]: reason: <Busy here> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK8d9c.4438d232.1>; state=6 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 234, <received> = <192.168.21.101>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 0(5890) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via 0(5890) DEBUG: <core> [receive.c:149]: After parse_msg... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6 global id=5 T start=0xffffffff 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK8d9c.e7b650e1.0>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=22 0(5890) DEBUG: <core> [parser/msg_parser.c:524]: parse_headers: this is the second via 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK463ae086>; state=6 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 235, <rport> = <5060>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=22 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=as493d6134 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 0(5890) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_field: <To> [42]; uri=[sip:5556003900000@dr.vm] 0(5890) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body [<sip:5556003900000@dr.vm>] 0(5890) DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq <CSeq>: <102> <INVITE> 0(5890) DEBUG: tm [t_lookup.c:965]: DEBUG: t_reply_matching: hash 51672 label 590185284 branch 1 0(5890) DEBUG: tm [t_lookup.c:1016]: DEBUG: t_reply_matching: reply matched (T=0x871afc98)! 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 2, reply code 486 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=as5c1d222e 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... 0(5890) DEBUG: tm [t_lookup.c:1148]: DEBUG: t_check_msg: msg id=6 global id=6 T end=0x871afc98 0(5890) DEBUG: tm [t_reply.c:2081]: DEBUG: reply_received: org. status uas=180, uac[1]=180 local=0 is_invite=1) 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 4194304, id 0 entered 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6 global id=6 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [REPLY_TWO] Incoming reply code: 486 0(5890) DEBUG: tm [t_reply.c:1204]: ->>>>>>>>> T_code=180, new_code=486 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 128, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 128, reply code 486 0(5890) acc [acc.c:275]: ACC: call missed: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius 0(5890) DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: 6e1b9f35a8b3e65859e29aefc1c26fc3 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=262 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_TYPE=application/sdp 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_SUPPORTED=replaces 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_ALLOW=INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_DATE=Thu, 07 Feb 2013 19:35:53 GMT 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_MAX_FORWARDS=68 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_USER_AGENT=Asterisk PBX 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=102 INVITE 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CALLID=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTACT=<sip:0221028400@192.168.21.106> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<sip:5556003900000@dr.vm> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM="Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0,SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 0(5890) DEBUG: exec [exec_mod.c:175]: executing [/opt/kamrouter/scripts/notify_reroute.sh 503 sip:3333003900000@ast-car.vm sip:0221028400@dr.vm 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm] sh: /opt/kamrouter/scripts/notify_reroute.sh: not found 0(5890) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=0 global id=0 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [FAIL_TWO] Rerouted gw returned 486 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [aaa] VYMAZAVAM3 0(5890) DEBUG: tm [t_reply.c:1702]: DEBUG: relay_reply: branch=1, save=0, relay=1 0(5890) DEBUG: <core> [msg_translator.c:1933]: old size: 572, new size: 485 0(5890) DEBUG: <core> [msg_translator.c:1951]: build_res_from_sip_res: copied size: orig:110, new: 23, rest: 462 msg= SIP/2.0 486 Busy here Via: SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0 Via: SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 From: "Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e To: <sip:5556003900000@dr.vm>;tag=as493d6134 Call-ID: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm CSeq: 102 INVITE Server: Asterisk PBX 11.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 1048576, id 0 entered 0(5890) DEBUG: dialog [dlg_hash.c:588]: ref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:590]: dialog id=3894 found on entry 2825 0(5890) DEBUG: dialog [dlg_hash.c:940]: dialog 0x871af3f0 changed from state 2 to state 5, due event 4 (ref 2) 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 3 0(5890) DEBUG: dialog [dlg_handlers.c:1469]: executing event_route 4 on state 5 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [dialog:failed] Deleting 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm from dialogs and dstprefix_limit 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_handlers.c:543]: dialog 0x871af3f0 failed (negative reply) 0(5890) DEBUG: dialog [dlg_cb.c:256]: dialog=0x871af3f0, type=4 0(5890) DEBUG: dialog [dlg_hash.c:643]: ref dlg 0x871af3f0 with 1 -> 3 0(5890) DEBUG: dialog [dlg_hash.c:646]: dialog callid='38e422e10af08b2e34d58e5b765e81d3(a)dr.vm' found on entry 2825, dir=1 0(5890) DEBUG: dialog [dlg_var.c:203]: Internal var-list ((nil)): 0(5890) DEBUG: dialog [dlg_var.c:213]: Dialog var-list (0x871af6f8): 0(5890) DEBUG: dialog [dlg_var.c:218]: "ru"=sip:003900000@cr1.vm:5060 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: d=0 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: et=1360265682.889 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: st=1360265682.889 (flags 1) 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 0(5890) acc [acc_cdr.c:192]: st=1360265682.889; et=1360265682.889; d=0; c1= 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 0(5890) DEBUG: dialog [dlg_handlers.c:937]: registering TMCB to wait for negative ACK 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 512, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 512, reply code 486 0(5890) acc [acc.c:275]: ACC: call missed: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius 0(5890) acc [acc.c:275]: ACC: transaction answered: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:867]: acc event for engine: radius 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 8388608, id 0 entered Thanks for a help. Efelin

2 4

Reg. functions in custom modules
by madhumanjusha＠integramicro.com 08 Feb '13

08 Feb '13

Hello All, Is it possible to call functions in custom modules say modules/my_module/my_file.c if my_file.c has a function and want to access/call it from kamailio core file ie., which is under /root/kamailio-3.3.2/receive.c ?? If not possible,what is the way to access functions in custom modules. Please do reply. Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 2

dlg_manage and engage_media_proxy
by Eduardo Lejarreta Langarica SARENET 08 Feb '13

08 Feb '13

Good evening. Kamailio 3.2.2 and mediaproxy 2.5.2 We find a problem when using engage_media_proxy and dlg_manage on the same kamailio.cfg (we loose one way audio) is ther any issue with regard to this? may be anything we're doing wrong? any other easy way to control concurrent calls? If 1.- we avoid dlg_manage with engage_media_proxy or 2.- we use dlg_manage with "use_media_proxy/end_media_session" every thing works fine. Any idea will be helpful. Thanks and best regards. ----------------------------------------------------------------------------------------------------------------------------------- Eduardo Lejarreta

3 2

kamctl restart and mediaproxy
by Eduardo Lejarreta Langarica SARENET 08 Feb '13

08 Feb '13

Good evening again. Kamailio 3.2.2 and mediaproxy 2.5.2 An long-time issue for us: kamctl restart stops mediaproxy audio. is there anyway to avoid this? Any idea will be helpful. Thanks and best regards. ----------------------------------------------------------------------------------------------------------------------------------- Eduardo Lejarreta

3 2

Somewhat OT: Specific SIP packets can cause ethernet controller reset
by Kristian Kielhofner 08 Feb '13

08 Feb '13

While not strictly Kamailio related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html I'm hearing initial reports that this issue is fairly widespread. -- Kristian Kielhofner

2 1

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Modify IP address in SDP (Austin Einter)
by madhumanjusha＠integramicro.com 08 Feb '13

08 Feb '13

On Tue, January 22, 2013 1:58 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. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Tue, 22 Jan 2013 18:37:13 +0530 > From: Austin Einter <austin.einter(a)gmail.com> > Subject: [SR-Users] Modify IP address in SDP > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANXt1k--VtC6aYHGTBZ5GqF8n=66g5PVwG4YhOC_uefxpVYy=g(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Dear All > When 200 OK to INVITE passes through Kamailio proxy , I want to change the > IP address in SDP to a specific IP address. > > > Can this be done by just modifying configuration files or do I need to > modify in source code. > > In case I need to modify source code, which files I need to look at. > > > Kindly provide pointers for said problem. > > > Thanks > Austin Hello Austin, 1...I was doing similar one for my task. Files : /root/kamailio-3.3.2/parser/sdp/sdp.h sdp.h file has struct that holds ip address and port numbers of audio and video. Stucture: sdp_session_cell struct has member named str ip_addr will give you ip address in sdp. 2...Yes you can edit existing code or else you can create your own 200ok packet.I created my own 200OK and invite packets with sdp. 3...You can grab below ip address when kamailio proxy is about to forward 200OK to end client.File where you can find 200OK is (though not sure but you can try this file edit existing 200OK): /root/kamailio-3.3.2/modules/tm/t_reply.c has below function static int _reply( struct cell *trans, struct sip_msg* p_msg, unsigned int code, char * text, int lock ) { } 4...I did the following way to get ipaddr...and below piece of code you can use it in the above function's body to edit 200OK's ip address:: char my_src_rtp_addr[8]; int my_src_rtp_addr_len=0; struct sdp_session_cell *sdp_session; if(0 != parse_sdp(p_msg)) { LOG(L_INFO,"Unable to parse sdp\n"); } sdp_session=get_sdp_session(p_msg,0); if(sdp_session!=0){ my_src_rtp_addr_len=strlen(my_src_rtp_addr); my_src_rtp_addr[my_src_rtp_addr_len]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+1]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+2]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+3]='\0'; strncpy(my_src_rtp_addr,sdp_session->ip_addr.s,sdp_session->ip_addr.len); } If any questions,please let me know. Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

3 3

Need help in rewriting the Request URI
by Ajay Sabat 07 Feb '13

07 Feb '13

Hi, I use Kamailio for basic call setup with one exception where it has to route call to another server by seeing the domain name in the Request URI. If the Request uri in the received INVITE message looks like "sip12345@video-conf", then it should route the INVITE request to the video-conf server by resolving the domain name. I changed the config file kamailio.cfg as given below. route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); # AKS: call routing if (uri=~".*@video-conf") { rewritehost("video-conf"); } # AKS route(RELAY); } } It resolves the domain name and routes the INVITE message to the resolved IP address. However it uses the domain name in the Request line and To header and I need to replace domain with the resolved IP address in the Request URI and To header. Now Request URI and To header look like "INVITE sip:12345@video-conf" and "To: <sip:12345:video-conf>" respectively and I need them as "INVITE sip:12345@video-conf" and "To: <sip:12345:video-conf>". I don't have any knowledge in tweaking kamailio. I tried some tricks like the following by going through documents and pervious emails in the forum. But it didn't work. Some of them have spoken about avp and pseudo variables. But I am getting confused as I have just begun using Kamailio and it is vast. # $ru = "sip:" + $rU + "@" + to_ip; I will appreciate if you could help me or guide me in right direction. Thanks, Ajay

2 2

Problems with MSILO module can't receive messages with m_dump.
by Nord7 07 Feb '13

07 Feb '13

Hello i have two problems with module MSILO and text messages from SIP client BRIA. 1. I can't send messages with m_dump. I have installed Kamailio 3.3 in default configuration. I'm added "m_dump("$fu")" to my config in route[REGISTRAR]. If i use client CSipSimple for sending messages to offline user "333". i'm receive notification message "*** User 333 offline!". Next i connect to Kamailio with user "333" and i see in kamailio log what "m_dumt" trying to send messages but can't because "request authorisation". How to disable authorisation request, without leak security. route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); if (m_dump("$fu")) { log("MSILO: offline messages dumped - if they were\n"); } else { log("MSILO: no offline messages dumped\n"); } exit; } }

3 2

msilo patch
by Stoyan Mihaylov 07 Feb '13

07 Feb '13

When I used, last versions from git, I got error and crash of kamailo using m_dump. I got latest version from git, and I made one patch: diff --git a/modules/msilo/msilo.c b/modules/msilo/msilo.c index 209b309..4bda8fa 100644 --- a/modules/msilo/msilo.c +++ b/modules/msilo/msilo.c @@ -1125,8 +1125,16 @@ static int m_dump(struct sip_msg* msg, str* owner_s) memcpy(tmp_extra_hdrs.s, extra_hdrs_str.s, extra_hdrs_str.len); memcpy(tmp_extra_hdrs.s+extra_hdrs_str.len, str_vals[4].s, str_vals[4].len); } else { - tmp_extra_hdrs.len = 0; - tmp_extra_hdrs.s = ""; + tmp_extra_hdrs.len = 1; + if ((tmp_extra_hdrs.s = pkg_malloc(1)) == NULL) + { + LM_ERR("Out of pkg memory"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + LM_ERR("failed to free the query result\n"); + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } + tmp_extra_hdrs.s[0]='\x0'; } hdr_str.len = 1024; if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, The problem is: tmp_extra_hdrs.s = ""; and consequent pkg_free(tmp_extra_hdrs.s); As I see, there are couple of ways to solve problem, one of them is setting tmp_extra_hdrs.s = 0; but then we have warnings, or setting tmp_extra_hdrs.s = 0; and check before pkg_free, but pkg_free is used on couple of places. I am not sure what is the best way to report bugs and offer patches.

2 1

Deleting the expired contacts using the Rtimer module
by kiran bhosale 06 Feb '13

06 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

2 1

401 message forwarding
by Алексей Ёлохов 06 Feb '13

06 Feb '13

Hi! I am setting up Kamailio as SBC with upper registration. I can forward REGISTER messages to my softswitch, but 401 UNAUTHORIZED from it is not coming to my UAC. Config section for registration route: route[REGISTER_MY] { rewritehost("3.3.3.3"); t_relay(); } What I see with ngrep (1.1.1.1 - my external UAC ip, 2.2.2.2 - Kamailio ip, 3.3.3.3 - mera softswitch ip): U 1.1.1.1:5060 -> 2.2.2.2:5060 REGISTER sip:2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;rport;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. Max-Forwards: 70. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. User-Agent: SFLphone. Contact: <sip:101@10.2.1.35:5060>. Expires: 600. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. . U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.2.1.35:5060 ;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d;received=1.1.1.1. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=2fb3028f366408eaa444e96dfd845447.8865. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. Contact: <sip:101@10.2.1.35:5060>;expires=600. Server: kamailio (3.3.3 (x86_64/linux)). Content-Length: 0. . U 2.2.2.2:5060 -> 3.3.3.3:5060 REGISTER sip:3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK929f.8339e446.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. Max-Forwards: 69. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. User-Agent: SFLphone. Contact: <sip:101@10.2.1.35:5060>. Expires: 600. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. . U 3.3.3.3:5060 -> 2.2.2.2:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK929f.8339e446.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=d67d5778704611e2a3c600505691470a. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. WWW-Authenticate: Digest realm="SIP-REGISTRAR", nonce="475a20c7". Server: MERA MVTS3G v.4.3.0-38t. Content-Length: 0. . U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=d67d5778704611e2a3c600505691470a. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. WWW-Authenticate: Digest realm="SIP-REGISTRAR", nonce="475a20c7". Server: MERA MVTS3G v.4.3.0-38t. Content-Length: 0. . Thanks in advance! --- Alex Elokhov

1 0

[Fwd: Deleting the expired contacts using the Rtimer module]
by kiran bhosale 06 Feb '13

06 Feb '13

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users