sr-users

sr-users@lists.kamailio.org

4 participants
37456 discussions

[Serusers] Radius Accounting
by Caxton 11 Jul '06

11 Jul '06

Hi all: I have installed radius server, and I also configure the "ser.cfg" for the radius accounting. However, there is a problem confues me. While the sip session have set up, the radiusclient send a Request-Account packet to radius server immediately, than radius server can save the start-up time of the session into database. But after the session terminated, radiusclient do nothing(it should send a Accounting-Stop packet to radius server)!! Therefore, I can not have a complete time record. I try to look up the solution in seruser mailing list, but I can not find what I want. Please help me, thanks. Sincerely Caxton -----ser.cfg----- [cut] # -- acc params -- modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "log_level", 2) modparam("acc", "radius_flag", 1) modparam("acc", "report_ack", 0) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { log(1, "REGISTER: Authenticating user\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("pc80.s206.tj.mcu.edu.tw", "subscriber")) { log(1, "REGISTER: Chanllenging user\n"); www_challenge(" pc80.s206.tj.mcu.edu.tw", "0"); break; }; save("location"); break; }; if(method=="INVITE") { log(1, "INVITE\n"); setflag(1); } if(method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); } if(method=="BYE"||method== "CANCEL") { log(1, "BYE or CANCEL\n"); setflag(1); } [cut]

2 1

[Users] Ack is not received the openser, when the phone lefted up.
by raviprakash sunkara 11 Jul '06

11 Jul '06

Hi Users, I installed Openser server and Using the Nathelper with RtpProxy . The openser server and RtpProxy are located in the Same system with ip add (192.168.2.2) , And Public IP address is xx.xxx.xxx.xxx., openSER sip_doamin name is xx.xxx.xxx.xxx Ours Network connect for openser System is given below modem------xx.xxx.xxx.xxx.-------------> router(firewall) ---------------------------------> server 1 |________________ > server 2 | |_________________> server 3 ( openser and rtp ) For Testing i'm X-lite softphones with different network og openser system When made the call from one softphone to other softphones.. When caller calls to callee , after the phone rings , -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

2 1

[Users] Problem with module dispatcher funtion ds_next_dst()
by Jose Gil Navarrete 11 Jul '06

11 Jul '06

Hi, somebody can help me?? I am trying to set up OpenSer 1.0.1 like dispatcher to loadbalance and failover but when I use ds_next_dst() for failover I obtain the next error: 0(19024) parse error (128,25-26): unknown command, missing loadmodule? ERROR: bad config file (1 errors) My config file is the next: loadmodule "/lib/openser/modules/sl.so" loadmodule "/lib/openser/modules/tm.so" loadmodule "/lib/openser/modules/rr.so" loadmodule "/lib/openser/modules/maxfwd.so" loadmodule "/lib/openser/modules/usrloc.so" loadmodule "/lib/openser/modules/registrar.so" loadmodule "/lib/openser/modules/textops.so" loadmodule "/lib/openser/modules/dispatcher.so" loadmodule "/lib/openser/modules/avpops.so" modparam("dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 2) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("2","0"); sl_send_reply("100","Trying"); forward(uri:host, uri:port); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); return; }; } failure_route[1] { if (t_check_status("408")) { if (ds_next_dst()) { # On Failure: Choose next Proxy t_on_failure("1"); # Relay to Proxy t_relay(); } else { t_reply("503", "Service not available"); return; } } } I try to use modparam("dispatcher", "dst_avp_id", 100) but I obtain: 0(19260) set_mod_param_regex: parameter <dst_avp_id> not found in module <dispatcher> 0(19260) parse error (86,21-22): Can't set module parameter 0(19260) parse error (128,25-26): unknown command, missing loadmodule? ERROR: bad config file (2 errors) --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

4 3

[Serusers] print avps in cfg file
by Luis Silva 11 Jul '06

11 Jul '06

Hi all, it's just a simple question. How can I print avps in the cfg file? tks in advance ps - I searched in the avp and xlog modules and couldn't find anything.

5 6

Fwd: [Serusers] Termination from SER to Asterisk
by CM0002＠aol.com 11 Jul '06

11 Jul '06

1 0

[Serusers] Cisco Gateway Disconnects the call
by Hamid Ali Asgari 10 Jul '06

10 Jul '06

Hi, I am stuck with the Cisco gateway (SIP-PSTN) disconnecting the call. The scenario is that a SIP client makes a call which is forwarded to a Cisco gateway, the call gets established but is disconnected after 20-30 seconds. Bellow is the call trace from ethereal, the problem is that the Cisco gateway (for some reason I don't know) sends a BYE. Does this have anything to do with the ACKS? I have tested the same scenario with the UA being Windows Messenger and the Cisco Gateway, and there are no problems. Any advice on troubleshooting this? Thanks in advance, Hamid PS: 192.168.11.102 = UA 10.30.30.18 = OpenSER 10.30.30.19 = Cisco Gateway ---------------------------------------------------------------------------- |Time | 192.168.11.102 | 10.30.30.18 | 10.30.30.19 | ---------------------------------------------------------------------------- |9.739 | INVITE SDP ( g729 g711U g711A telephone-event) | |SIP From: sip:102@mydomain.com To:sip:0098146@mydomain.com | |(5060) ------------------> (5060) | | |9.743 | | INVITE SDP ( g729 g711U g711A telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |9.743 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |9.752 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |9.932 | 100 Trying| | |SIP Status | |(5060) <------------------ (5060) | | |11.668 | | 183 Session Progress SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |11.670 | 183 Session Progress SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |11.671 | | 200 OK SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |11.673 | 200 OK SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |11.687 | RTP (g729) | |RTP Num packets:1634 Duration:19.469s ssrc:126289683 | |(6000) <-------------------------------------- (16400) | |11.735 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |12.003 | RTP (g729) | |RTP Num packets:18 Duration:0.161s ssrc:3634 | |(6000) --------------------------------------> (16400) | |12.169 | | 200 OK SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |12.171 | 200 OK SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |12.184 | RTP (g729) | |RTP Num packets:100 Duration:0.979s ssrc:3634 | |(6000) --------------------------------------> (16400) | |12.254 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |13.169 | | 200 OK SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |13.172 | 200 OK SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |13.183 | RTP (g729) | |RTP Num packets:200 Duration:1.980s ssrc:3634 | |(6000) --------------------------------------> (16400) | |13.240 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |15.169 | | 200 OK SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |15.171 | 200 OK SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |15.183 | RTP (g729) | |RTP Num packets:400 Duration:3.980s ssrc:3634 | |(6000) --------------------------------------> (16400) | |15.235 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |19.169 | | 200 OK SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |19.172 | 200 OK SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |19.183 | RTP (g729) | |RTP Num packets:400 Duration:3.980s ssrc:3634 | |(6000) --------------------------------------> (16400) | |19.245 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |23.169 | | 200 OK SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |23.172 | 200 OK SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |23.183 | RTP (g729) | |RTP Num packets:400 Duration:3.980s ssrc:3634 | |(6000) --------------------------------------> (16400) | |23.256 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |27.169 | | 200 OK SDP ( g729 telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |27.172 | 200 OK SDP ( g729 telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |27.211 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |27.212 | RTP (g729) | |RTP Num packets:402 Duration:3.971s ssrc:3634 | |(6000) --------------------------------------> (16400) | |31.170 | | BYE | |SIP Request | | |(5060) <------------------ (55584) | |31.174 | BYE | | |SIP Request | |(5060) <------------------ (5060) | | |31.221 | 200 OK | | |SIP Status | |(5060) ------------------> (5060) | | |31.223 | | 200 OK | |SIP Status | | |(5060) ------------------> (5060) | |39.683 | 408 Request Timeout | |SIP Status | |(5060) <------------------ (5060) | | |39.751 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |39.752 | | ACK | |SIP Request | | |(5060) ------------------> (5060) |

1 0

Re: [Serusers] Implementing a black list.
by CM0002＠aol.com 10 Jul '06

10 Jul '06

You do not need hundreds of numbers, just carefully read hundreds of ITU documents ;-)..... In fact blocking for instance 00xx7, 00xx8, 00xx9 and 00xx18 helps in many countries. "Suspiciouse" countries like bahamas should be blocked anyway because there is always somebody running a nice "money maker machine". And don't forget to block satellite gateways because the often are used ffom cruzer phones to make expensive calls (20 USD/min upwards) However dial plans changing on daily base. Subscribe to the ITU page to get latest informations. Regards Christian

2 1

[Users] OpenSER v1.1.0 Released
by Daniel-Constantin Mierla 10 Jul '06

10 Jul '06

Hello everybody, OpenSER v1.1.0 is out. After about eight months of development including two months testing phase, another cycle has ended in version 1.1.0. This version introduced a lot of new features as well as enhancements in the existing components. dialog, path, siptrace, statistics, osp and unixodbc are new modules added in this version. NAPTR, serial forking in core, TCP/TLS improvements, statistics are some of the core news. Cacheless usrloc, load balancing with failover support, path support in registrar, raw SQL queries in config file with result available in AVPS are amoung modules' news. Read full release notes at: http://www.openser.org/index.php?option=com_content&task=view&id=47&Itemid=… With this release, the OpenSER web site has a new look, hope it will ease user experience in finding the information quickly. http://www.openser.org/ You have to update the bookmarks inside the site, some pages have new links. A branch has been created: rel_1_1_0. Getting the v1.1.0 sources from CVS is now: "cvs co -r rel_1_1_0". The tarballs with sources are available at: http://www.openser.org/pub/openser/latest/src/ http://www.openser.org/index.php?option=com_content&task=view&id=29&Itemid=… Binaries and packages for different distributions will be uploaded in the next days as they become available. Contributors are welcome to submit packages for preferred OS and architecture. We will make them available on OpenSER download site. The documentation for modules is posted at: http://openser.org/docs/modules/1.1.x/ Documentation repository: http://www.openser.org/index.php?option=com_content&task=view&id=34&Itemid=… We would like to thank to all developers and contributors for their work on this new release. Cheers, Daniel

2 1

RE: [Serusers] Open source GSM/CDMA network
by Aimable Habiyakare 10 Jul '06

10 Jul '06

Check this one out http://www.wireless3g4free.com/ ________________________________ From: Magnus Kelly [mailto:magnus.kelly@mcomwifi.net] Sent: Monday, July 10, 2006 4:26 PM To: Aimable Habiyakare; CM0002(a)aol.com Cc: serusers(a)iptel.org Subject: RE: [Serusers] Open source GSM/CDMA network As one of the UK gsm guard band license holders this thread is exactly what I am hoping for, currently the only SDR GSM bts I am aware of is from www.vanu.com <http://www.vanu.com/> who in discussion, confirm that they have it running under Linux with a wide band final output transceiver + and as such the radio can be changed on the fly between different technologies - meaning can start with gsm and then potentially change by software to other technology as new standards emerge. Perhaps you could share where the info on the open source WCDMA bts can be located? Magnus ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Aimable Habiyakare Sent: Monday, July 10, 2006 2:30 PM To: CM0002(a)aol.com Cc: serusers(a)iptel.org Subject: RE: [Serusers] Open source GSM/CDMA network When you put into consideration Software Defined Radio, whereby software and (general hardware) is used to generate radio signals which were being generated by analog integrated circuits, maybe there is a possibility. Look at what GNU Radio is trying to do. Together with SER/Asterisk, maybe something can be built. BTW there is an open source WCDMA software base station already around. ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of CM0002(a)aol.com Sent: Monday, July 10, 2006 2:46 PM To: serusers(a)iptel.org Subject: [Serusers] Open source GSM/CDMA network GSM suppliers working on that already and implementing softswitches in order to increase capacity and lower cost. However GSM is circuit switched - not packet switched (except GPRS) and here is the point cost can't really dropped. The way it will be more probably is that GSM CDMA might be used in the furture in the suburbans and in metropolitan areas we will see a WIMAX/VOIP scenario. This will result in a hybrid network for at least a transitional period. We already experienced it with AMPS/TDMA to GSM transition. CDMA (spread spectrum) is almost dead because high cost and leck of roaming capacity. WCDMA/UMTS (spread spectrum 3G) is very expencive and hard to manage (in Europe the launch was delayed by 2 years because hard and software problems). I recently started with a project (small scale) to setup such WIMAX based networks. The aim is to develop a low cost internet/phone network for rural communities in third world countries. Anybody with routing experience who wants to join me on a professional base is welcome. Regards Christian

1 0

[Serusers] Echo Cancellation
by ravi reddy 10 Jul '06

10 Jul '06

Hi users I noticed some strange thing yesterday when i made a SIP-SIP call everytthing was O.K , when I made SIP-Pstn local call its pretty good BUT When I made SIP-PSTN (international) I am Hearing my Voice back at Some stages , I tried for lot of calls but i dont understand where actually the problem is creating , Can any body please solve this "ECHO" problem Thanks. Regards, Ravi.

4 6

Jump to page:

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