sr-users August 2007

sr-users@lists.kamailio.org

169 participants
286 discussions

by ram

Hi iam installing openser SVN and Freeradius 1.16 followed the link as mentioned in document when i run radiusd -X iam getting below problem 1. rlm_sql (sql): Connected new DB handle, #4 2. Module: Instantiated sql (sql) 3. Module: Loaded Acct-Unique-Session-Id 4. acct_unique: key = "User-Name, Acct-Session-Id, NAS-IP-Address, Client-IP-Address, NAS-Port" 5. rlm_acct_unique: Cannot find attribute 'NAS-IP-Address' in dictionary 6. radiusd.conf[1159]: acct_unique: Module instantiation failed. 7. radiusd.conf[1753] Unknown module "acct_unique". 8. radiusd.conf[1747] Failed to parse preacct section. any suggestiong ram

17 years

[OpenSER-Users] PRACK handling

by Zahid Mehmood

Hello everyone, What is the preferred way of handling PRACKS? simply relay them if they are in loose_route? should they be authenticated? What, if any, are the potential risks of not authenticating PRACKS? Thanks in advance. -- Zahid

17 years

[OpenSER-Users] disable media/nathelper

by Aymeric Moizard

Hi, I'm facing an issue with openser. I checked the latest SVN code and it seems my issue is not solved even in the svn version. My UA is sending an SDP with 2 m lines: one for audio and one for video. In a re-INVITE, I want to disable video and I'm sending m=video 0 RTP/AVP.. My openser is currently modifying the port number for the video line even if it's 0 which means it's disabled... This behavior would of course happen even for audio lines that ones wants to disable. I was not able to find a clean fix in nathelper.c and hope to get some help for proposing one... I guess it would be something close to: if (oldport==0) skip altering ip/port for media and continue; but I don't want to insert any mem leak or lose a session... tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

17 years

[OpenSER-Users] how does OpenSER run on IPV6?

by null_neo＠sina.com

Dear sir: First,I am a Chinese,and my English is poor.^_^ We are testing kinds of applications on IPV6 including VOIP. And we use Linphone as our UA. If  I don't use domain name,how can I add a account? On IPV4,I add account " # openserctl add test test test@ipv4_address" .And in the file "openser.cfg" ,I directly write my SIP server ipv4_address . Can I directly replace "ipv4_address" by "ipv6_address"?  I has tried,and the resault is bad. I am very sorry that I am amateurish for VOIP ,because Our work is only to test. Thank you!                                                                                                 Li  Zhonglei ------------------------------------------------------------------- 吉贝克 ——商业智能的领军者( http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… ) =================================================================== 注册新浪2G免费邮箱（ http://mail.sina.com.cn/chooseMode.html ）

17 years

[OpenSER-Users] about openSER and SIPp

by bozhang＠nec-labs.com

Hi there, I'm doing sipp performance measurements with openser 1.2. I run OpenSER on machine A, then run one SIPp on machine B, one SIPp on machine C. For the SIPp on machine C, I start it using UAS mode, ie., ./sipp -sn uas For the SIPp on machine B, if I start is using the following command: ./sipp -sn uac C -rsa A -m 1 -r 1 Everything is correct. The OpenSER can relay messages correctly. But if I start SIPp on machine B using the following command: ./sipp -sn uac -rsa A -m 1 -r 1 -d 5000 Then the SIPp client on machine B will crash because of receiving unexpected "ringing 180" message. Note: the only difference is I add a pause 5 seconds using "-d 5000" option. What I found is: the OpenSER server will keep sending "INVITE" to C during the '5 second pause', so C will keep reponding with "180 Ringing" to B. Does anybody know why OpenSER keeps sending "INVITE" to C while B only issues one "INVITE" request? BTW, the configuration file for OpenSER is very simple: # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=32 log_facility=LOG_LOCAL0 disable_tcp=yes disable_dns_blacklist=yes disable_dns_failover=yes # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # listen=udp:192.168.2.102:5060 # ------------------ module loading ---------------------------------- loadmodule "modules/tm/tm.so" modparam("tm", "wt_timer", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ t_relay(); } Thanks a lot! -Bo Zhang

17 years

Re: [OpenSER-Users] Dispatcher

by Daniel-Constantin Mierla

OK, so you didn't get in the condition of the failure route. Daniel On 08/02/07 12:58, Thomas Deillon wrote: > Hello, > > It's ok, I finally find the solution with raviprakash sunkara. > It was the parameter of tm. fw_timer. > > Cheers, > > Thomas > > On 02/08/07, *Daniel-Constantin Mierla * <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > can you run in debug=7 and log_stderror=yes and watch the debug > messages? If you have issues understanding what is happening, then > just > forward it to me along with dispatcher.list file. > > Daniel > > On 08/01/07 15:56, Thomas Deillon wrote: > > Hi, > > > > I'm using OpenSER as PROXY and REGISTRAR server, a Cisco as Gateway > > and 2 Asterisks (using dispatcher) as Voicemail and conference > Servers. > > > > It works perfectly on load-balancing, but not in failover because I > > never go on the failure route. > > Do you have a idea (a part of my config file below) ? > > rq: Just for information, the 12 is the milliwatt() function on > Asterisks. > > > > my openser.cfg > > > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > modparam("usrloc", "db_url", "mysql://openser:openser@localhost > /openser") > > modparam("usrloc", "db_mode", 1) > > > > > modparam("dispatcher","list_file","/usr/local/etc/openser/dispatcher.list") > > modparam("dispatcher","flags",2) > > modparam("dispatcher", "dst_avp", "$avp(i:271)") > > modparam("dispatcher", "grp_avp", "$avp(i:272)") > > modparam("dispatcher", "cnt_avp", "$avp(i:273)") > > modparam("dispatcher", "force_dst", 1) > > > > route { > > [...] > > > > if (method=="INVITE") > > { > > if (uri=~"sip:12@.*") > > { > > xlog( "L_ERR", "A: $rm $fu:$op -> $du AVP: $avp(i:271), > > $avp(i:272), $avp(i:273)\n" ); > > ds_select_dst("1", "4"); > > #t_relay(); > > xlog( "L_ERR", "B: $rm $fu:$op -> $du AVP: $avp(i:271), > > $avp(i:272), $avp(i:273)\n" ); > > t_on_failure("1"); > > } > > } > > [...] > > } > > > > failure_route[1] > > { > > xlog( "L_ERR", "C: $rm $fu:$op -> $du\n" ); > > if(t_check_status("408")) > > { > > ds_mark_dst(); > > ds_next_dst(); > > } > > else { t_reply("501", "Not Implemented"); } > > > > xlog( "L_ERR", "D: $rm $fu:$op -> $du\n" ); > > } > > > > Cheers, > > > > Thomas > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

17 years

[OpenSER-Users] Dispatcher

by Thomas Deillon

Hi, I'm using OpenSER as PROXY and REGISTRAR server, a Cisco as Gateway and 2 Asterisks (using dispatcher) as Voicemail and conference Servers. It works perfectly on load-balancing, but not in failover because I never go on the failure route. Do you have a idea (a part of my config file below) ? rq: Just for information, the 12 is the milliwatt() function on Asterisks. my openser.cfg modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_url", "mysql://openser:openser@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("dispatcher","list_file","/usr/local/etc/openser/dispatcher.list") modparam("dispatcher","flags",2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "force_dst", 1) route { [...] if (method=="INVITE") { if (uri=~"sip:12@.*") { xlog( "L_ERR", "A: $rm $fu:$op -> $du AVP: $avp(i:271), $avp(i:272), $avp(i:273)\n" ); ds_select_dst("1", "4"); #t_relay(); xlog( "L_ERR", "B: $rm $fu:$op -> $du AVP: $avp(i:271), $avp(i:272), $avp(i:273)\n" ); t_on_failure("1"); } } [...] } failure_route[1] { xlog( "L_ERR", "C: $rm $fu:$op -> $du\n" ); if(t_check_status("408")) { ds_mark_dst(); ds_next_dst(); } else { t_reply("501", "Not Implemented"); } xlog( "L_ERR", "D: $rm $fu:$op -> $du\n" ); } Cheers, Thomas

17 years

[OpenSER-Users] Header Mangling

by Peter Gradwell

hi ya, When we send phone call out to our PSTN gateways and carriers, we often have to modify the To and From headers to put them in the right syntax and to copy the caller id up from the remote-party-id header. We think the uac module will allow us to do this on the From header, but we also need to add/replace digits in the To header, e.g. changing it into INTL format, from 01225 to 441225... If we do this using TextOps then the problem is that it only changes the string sent to the Telco switch, and it is not therefore changed on the reply, and so the end user agent does not then handle the replies properly. Does anyone know how we can mangle the To header? Is there a similar module to uac for To headers? cheers peter

17 years

[OpenSER-Users] Attribue User-name was not found in request, unique ID MAY be inconsistent

by zze-WU Yin RD-MAPS-LAN

Hi, everybody Another probleme with Openser and Freeradius. When I tried to use the acc module, i recieved: rlm_acct_unique: WARNING Attribue User-name was not found in request, unique ID MAY be inconsistent And in the Accounting-Request packet, i can not find User-name. Do anybody know why? Thank you very much!!

17 years

[OpenSER-Users] Pb with log_fmt

by zze-WU Yin RD-MAPS-LAN

Hi, thank you very much. I know what the problem is now. Because I use a openser-1.2.1-notls. So the "log_fmt" is obsolete. In the http://www.openser.org/docs/modules/1.2.x/acc.htm it wrote that: The accouting module will log by default a fixed set of attributes for the transaction - if you customize you accouting by adding more information to be logged, please see the next chapter about extra accouting - Section 1.2. The fixed minimal accouting information is: Request Method name >From header TAG parameter To header TAG parameter Call-Id 3-digit Status code from final reply Reason phrase from final reply Time stamp when transaction was completed If a value is not present in request, the empty string is accounted instead. I want have a User-Name in the accounting information. So I saw in the Section 1.2 Selection of extra information is done via xxx_extra parameters by specifying the names of additional information you want to log. This information is defined via pseudo-variables and may include headers or AVPs values or other message or system values. The syntax of the parameter is: xxx_extra = extra_definition (';'extra_definition)* extra_definition = log_name '=' pseudo_variable And the pseudo_variable I need is '$au' for User-Name. But I don't understand how to use this paramater in the openser.cfg. Do you have some ideas??

17 years

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sr-users August 2007