- sr-users - kamailio.org

[Kamailio-Users] Fax Detection
by Abdul Hakeem 04 Sep '08

04 Sep '08

Hello, Is there a way to identify in incoming call as a fax (perhaps by detecting CNG tone ). Regards, Abdul Hakeem

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[Kamailio-Users] ERROR:mysql:db_mysql_new_connection: driver error: Too many connections
by Samael - 04 Sep '08

04 Sep '08

Hi, We are deploying OpenSER in a large infrastructure together with Asterisk to support thousands of current transactions. We are using udp listener and mysql module in openser. We had to increase the children parameter of the openser.cfg to 200 because Openser couldn't handle the high amount of concurrent SIP REQUESTs from ASTERISK (Not only INVITES, but also a lot of SIP OPTIONs), that means that Openser couldn't handle more than 'children' REQUEST loosing the rest of the concurrent REQUEST. children=200 But then, the following error appeared in the openser: ERROR:mysql:db_mysql_new_connection: driver error: Too many connections This is because the Standard parameter of mysql "max_connections" is set to 100. In mysql CLI, we can see that every child process of openser has a mysql connection. (see in Mysql> SHOW PROCESSLIST) The recommended max_connections parameter is 500-1000 => we have a limit of 500-1000 concurrent SIP REQUEST to be processed by the openser. Why there is a process for every child? Why is neccesary a mysql connection for every child? Can the openser have a pool of mysql connection to be used by thousands of children? Regards

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[Kamailio-Users] OpenSER reply 513 when register use FQDN
by Steven Wu 04 Sep '08

04 Sep '08

Hi, I just want use FQDN in my networking environment. My SJPhone use FQDN register to OpenSER 1.2.2 but was rejected by 513 Message to big. However it can be registed use the same profile just replace FQDN to IP address. The DNS query process worked fine and got the right IP address. My environment elments looks like: OpenSER 1.2.2 at host 192.168.1.120 / ipcsip.org DNS server at host 10.57.39.53 SJPhone at host 192.168.1.105 REGISTER package looks like: 192.168.1.105 -> 192.168.1.120 REGISTER sip:ipcsip.org:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062> Contact: <sip:1000@192.168.1.105> Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 192.168.1.120 -> 192.168.1.105 SIP/2.0 513 Message too big Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport=5060 From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.b7d8 Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 I got same result evenif enabled dns & rev_dns in openser.cfg (dns=yes rev_dns=yes). So, What wrong with me? Coudle some give me some instructions? Best regards, Steven Wu

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[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 04 Sep '08

04 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Serusers] Which kind of SIP message can I use to exchange Data out of session?
by Peter Cai 04 Sep '08

04 Sep '08

Hi All, If a SIP client has to exchange some data out of a Session with the proxy server, such as get his "buddy list", what SIP Message can I use? I didn't find any thing suit for this purpose. Should I use something beside SIP such as HTTP.

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[Serusers] cannot load module acc.so
by 静于 04 Sep '08

04 Sep '08

hello, I cannot load the acc.so module.The config file is below: # ------------- version 0.8.11-0 # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=10.1.2.3 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=bigu.edu alias=10.1.2.3 # dns - Uses dns to check if it is necessary to add a "received=" field # to a via. Default is no. # rev_dns - Same as dns but use reverse DNS. dns=no rev_dns=no port=5060 children=4 # check_via - Turn on or off Via host checking when forwarding replies. # Default is no. arcane. looks for discrepancy between name and # ip address when forwarding replies. check_via=yes # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # user | uid - uid to be used by the server. 99 = nobody. uid="nobody" # group | gid - gid to be used by the server. 99 = nobody. gid="nobody" # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. mhomed=0 # ------------- external module loading loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # set ";lr" tag to ?�?;lr=true?�? modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:>password>@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:>password>@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------- routing logic route { # ------------- routine checks # stop forwarding at 10 hops to prevent infinite loops if (!mf_process_maxfwd_header("10")) { log(1, "LOG: Too many hops\n"); sl_send_reply("483", "Too many hops"); break; }; # prevents private ip space from being used if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { if (method=="REGISTER") { log(1, "LOG: Someone trying to register from private IP\n"); sl_send_reply("479", "Please don't use private IP addresses" ); break; }; }; # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); break; }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(10\.1\.2\.3|(proxy\.)?bigu\.edu)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # challenge/response if (!www_authorize("bigu.edu", "subscriber")) { www_challenge("bigu.edu", "0"); break; }; # only registered users are allowed if (!is_user("replicator") & !check_to()) { log(1, "LOG: unregistered user registration attempt\n"); sl_send_reply("403", "Only registered users are allowed"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # process traffic local to BigU and the PSTN # Find the canonical username lookup("aliases"); # check domain again, if it is not still local after the alias # table lookup, just send it on its way. We do not authenticate # traffic we forward if (!(uri=~"^sip:(.+@)?(10\.1\.2\.3|(proxy\.)?bigu\.edu)([:;\?].*)?$")) { route(5); break; }; # now check for destinations through the gateway. 911 and 9911 # are always sent to the gateway. The assumption is that other all # numeric usernames between 5 and 20 digits are really pstn numbers # and so they are routed to the gateway if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{5,20}@.*") ) { route(3); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # handle local SIP destinations not found in usrloc db # mostly offline or non-existent users if (!lookup("location")) { route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@10\.1\.2\.5([;:].*)*" ) { log(1, "LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ # ------------- process traffic leaving BigU for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==10.1.2.3) & !(proxy_authorize("bigu.edu", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log(1, "LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log(1, "LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("bigu.edu", "0"); break; }; }; # to maintain credibility of our proxy, we check >From in INVITEs if (!src_ip==10.1.2.3 & method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") ) { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # five digit numeric addresses are internal freebies sent to the pbx # without authentication if (uri=~"^sip:[0-9]{5}@(10.1.2.3|(proxy\,)?\.bigu\.edu)") { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(10.1.2.3|(proxy\.)?bigu\.edu)") { if (!(src_ip==10.1.2.3 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("bigu.edu", "subscriber")) { proxy_challenge( "bigu.edu","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance if (uri=~"sip:91[0-9]*@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "Local calls only"); break; }; }; }; # authenticated and authorized, now accounting is set setflag(1); }; rewritehostport("10.1.2.5:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process calls for users offline route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-BigU domain # are not authenticated by BigU route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } Can anyone help BR Ivy ___________________________________________________________ �Ż��䣬��䣡 http://cn.mail.yahoo.com/

1 0

Re: [Kamailio-Users] svn diff
by Vadim Berezniker 04 Sep '08

04 Sep '08

Oops hit Ctrl+Enter svn diff http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/ http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.3/ I've been trying to run this against the public SVN but it never completes.... I hate sourceforge. From: Vadim Berezniker Sent: Wednesday, September 03, 2008 1:57 PM To: 'users(a)lists.kamailio.org' Subject: svn diff Can someone with access to the dev svn run this for me?

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[Kamailio-Users] Could you please help me how to integrate RLS AS and PS AS with SERIMS?
by mahesh.peddi＠info-spectrum.com 04 Sep '08

04 Sep '08

Hi, I am working on openser 1.3.2 as RLS application server and presence application server. I want to integrate this two application servers with SERIMS. Could you please help me how to integrate and make run this. To do this I have run SERIMS components on two servers? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

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[Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Kristian Kielhofner 03 Sep '08

03 Sep '08

Hello everyone, Feel free to set $OSSPROXY to whatever you like = Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). $OSSPROXY is my favorite SIP toolkit. In addition to being an excellent proxy and registrar it serves as a platform to do SIP mangling and wrangling. With that in mind let me propose a crazy idea... How hard would it be to get $OSSPROXY to be able to parse what is commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has no business getting too involved in the SDP but I feel it makes sense in some instances (like this one). What if I want to make a routing decision based on an encapsulated ISUP parameter of some sort? Who knows what it could be!?!? Maybe there are some other uses but that makes the most sense and seems to be the most practical. It should be possible to develop an $OSSPROXY module to: - Use the SDP parser to find the ISUP SDP part: http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing - Pass that part into a "real" ISUP parser (if necessary). I couldn't seem to find the perfect library but perhaps something like this could work: http://www.openss7.org/isup.html - Make the ISUP parameters available via AVPs, pseudo variables, etc for use in the $OSSPROXY script. I don't really have a need for this but it seems like it could be cool. Am I nuts? -- Kristian Kielhofner http://blog.krisk.org

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[Serusers] Is setdsturi supported in SER
by Steven C. Blair 03 Sep '08

03 Sep '08

I am wondering if setdsturi is supported in SER 0.9.7-pre3 Thanks

1 0

[Kamailio-Users] svn diff
by Vadim Berezniker 03 Sep '08

03 Sep '08

Can someone with access to the dev svn run this for me?

1 0

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 03 Sep '08

03 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Kamailio-Users] VQ_MALLOC memory manager
by Henning Westerholt 03 Sep '08

03 Sep '08

Hi all, does somebody uses the VQ_MALLOC memory manager? I thought about removing this one, as it don't work correctly at the moment (it lacks a realloc implementation). This would be a good cleanup in the core. If somebody still depends on this memory manager, perhaps has also some fixes created for this, please reply to this thread. If nobody speaks up i'll remove the VQ_MALLOC implementation at the end of this month. Cheers, Henning

1 0

[Kamailio-Users] NAT transversal without mediaproxy or rtpproxy
by BERGANZ François 03 Sep '08

03 Sep '08

Hello, I am looking for to do nat transversal without mediaproxy or rtpproxy! I cant find any example without mediaproxy or rtpproxy. Have you an idea? I need to register my users, and contact them after Thank you

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[Kamailio-Users] t_relay to two diffrent proxy
by Asim Riaz 03 Sep '08

03 Sep '08

Hi , I am trying to relay invite from Kamailio to one of two proxies, as long as it receive the reply from one it should not forward to second but incase if the proxy1 does not reply then relay to proxy2. here is my config route(1); if((src_ip != proxy1_ip_address) || (src_ip !=proxy2_ip_address) ) { if(!t_relay_to_udp("proxy1_ip_address", "5060")) || (!t_relay_to_udp("proxy2_ip_address","5060")) { end_media_session(); sl_reply_error(); } } I am also allowing only my proxies can see the location tables; f ((src_ip== Proxy1_ip_address) || (src_ip= Proxy2_ip_address)) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; for some reason incoming calls from my proxies are looping means Kamailio sending back to one of my proxyies, i dont know if there is somthing missing in my configuration or somthing else is causing this problem. Regards Asim Riaz

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[Kamailio-Users] Kamailio dispatcher Module redirection
by Asim Riaz 03 Sep '08

03 Sep '08

Hi All I am trying to configure Openser as loadblancer for two more Kamailio. I wan loardbalancer not stay in the middle once routed to other kamilio servers. my first question is, dispathcer list always take IPs or I can configure proxy names as well. at the moment I am checking the to_uri and then redirecting the traffic to my one of Kamilio on my network, it seems to work fine with Linksys but not other phones e.g polycom xlite or even sjphone. if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") { ds_select_domain("1", "4"); sl_send_reply("300","Redirect"); exit; } any suggestion. Thanks in advance Regards Asim Riaz

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[Kamailio-Users] uac_redirect question
by Zahid Mehmood 03 Sep '08

03 Sep '08

Hi, Need a little help with get_redirects(), using openser 1.2. I am using the following in failure_route to capture contacts from a "multiple choices" reply from cisco media gateway. Following code sets the ruri to the last contact, however, the $ds show more information about other contacts. Is there a way to reset this so that only ruri information is used? Cisco response and xlog information is also given below. if(!get_redirects("*:*")) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci \n"); acc_db_request("480", "acc"); t_reply("480", "Temporarily Unavailable"); exit; } # get last URI from destination-set and set it as R-URI xlog("L_INFO", "Redirect from UAC intercepted 1 - M= $rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:tmp)/g"); $avp(s:tmp) = $ds; avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); avp_pushto("$ru", "$avp(s:tmp)"); setflag(29); xlog("L_INFO", "Redirect from UAC intercepted 2 - M= $rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci \n"); append_branch(); route(17); # process exit; } Thanks in advance for your advice. -- Zahid The response from gateway: U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP 10.10.12.140;branch=z9hG4bK612a4f20B9293665. From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F. To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC. Date: Tue, 26 Aug 2008 12:19:04 GMT. Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1. Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>. Content-Length: 0. xlog entries: Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 - M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32 >;q=0.01, <sip:10512@10.10.0.98>;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01, sip:10512@10.10.0.98 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140

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[Kamailio-Users] t_relay
by sabino frisardi 03 Sep '08

03 Sep '08

Hi all, I'm using openser 1.4. Can anyone tell me how can i call t_relay() from another module. Can anyone help me? any suggestion? Thanks for your help. Regards -- Sabino Frisardi

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[Serusers] how can i get the IP?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? 2008-09-03 邱磊

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[Serusers] how can i get the ip?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip ?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 03 Sep '08

03 Sep '08

Thanks David for your answer, but can you explain how radius can do that since sending BYE messages is more at the signaling level. I dont quite understand how radius can even communicate at the SIP signaling level. From: David Villasmil Sent: Sunday, August 31, 2008 4:33 PM To: Matteo D'Amato Subject: Re: [Kamailio-Users] CDRTool and Prepaid You could use radius. On Sun, Aug 31, 2008 at 10:09 PM, Matteo D'Amato > wrote: Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Serusers] SER 0.9.7-pre3 and timers
by Steven C. Blair 02 Sep '08

02 Sep '08

I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. Thanks, Steve

1 0

[Serusers] Error opening ser's fifo /tmp/ser_fifo
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 6.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

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[Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu 8.04LTs
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 8.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744 -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

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[Serusers] Local number portability
by inge 02 Sep '08

02 Sep '08

Hi all, Is someone know if it's possible to use SER for prefix calls for LNP ? Then the call will be sent to AS5350 and the prefix must define the carrier for out ported number. I search information on this topic but I find nothing about SIP and LNP. Thanks for our informations. Regards

1 0

[Kamailio-Users] route number
by BERGANZ François 02 Sep '08

02 Sep '08

Hello, I need to have a route number like route(1000) But after some tests, I cant have more than route(39) How can I have more? Thank you

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[Kamailio-Users] How to config OpenSER as IMS server
by Steven Wu 01 Sep '08

01 Sep '08

Hi everybody, Currently we are working on an IMS project. This project need a IMS server. Finally we choiced OpenSER as the IMS server. But have no experience how to use OpenSER as a IMS server. Could someone tell me how to config it? Best regards, Steven Wu

3 2

[Kamailio-Users] Mediaproxy 2.0 - Dispatcher vs Relay
by Nuno Marques 01 Sep '08

01 Sep '08

Hi, I'm trying to understand better how mediaproxy works. Just read the mediaproxy-ng.org page... i understood the necessity of using mediaproxy for NAT traversal, but i have one doubt... - What's the difference between the dispatcher and the relay? Can someone explain me the concept or point me to some literature so i can clear my mind :) Thanks in advance, Nuno

3 2

[Kamailio-Users] kamailio 1.5 and beyond
by Henning Westerholt 01 Sep '08

01 Sep '08

Hi all, i want to share some thoughts where i think the priorities for the next major release(s) of Kamailio should be. This is the continuation of the work i (and many others) have done for the previous releases, but i thought it makes sense to summarize it a bit on a higher (abstraction) level. 1. Refactoring and cleanup We still have quite a lot of old code in our repository, which is no longer working or obsolete. Some implementations uses way to much gotos, or macros that are several pages long. This makes important parts of the core and modules harder to understand and maintain then necessary. Especially in the area of utility functions there is still to much duplicated code. proposed actions: - identification and removal of no longer working code in the core and modules - identification and unification of redundant (duplicated) code in core and modules to one implementation - identification and removal of obselete code parts in core and modules. - refactoring of code to use less gotos and macros 2. User and developer documentation With regards to user documentation quite a lot of work has been done in the past releases. But with the developer documentation, especially in certain modules, there is still plenty of content missing. As the involvement of individuals and companies can change, documentation is necessary to ensure a continued maintenance of the code. proposed actions: - conversion of already existing documentation to doxygen format - add a short doxygen module definition to every module, explaining the purpose - add a short doxygen file definition to every file, explaining the purpose - extend doxygen docs in important areas like the core and modules like TM - autogeneration of database related module documentation from the XML scheme definitions 3. Consolidation in core and modules At the moment we've 85 modules in our repository. Some modules implement only one or two functions, which makes no real sense with regards to the maintainance overhead that comes with it. 24 of the modules are using database access and 7 implement database interfaces. There is no clear boundary in some areas of the core and modules, this hinders the further development of the server. proposed actions: - integrate several modules that share responsibilities and not implement that much functionality on their own into one - autogeneration of module interface and database infrastructure code from the XML database scheme definitions - identify obselete module functions that can be done easily with pseudo-variables now. Remove these functions. - move certain core parts that do not need to be there into modules, making the core smaller and more flexible - identify core APIs which are used from modules, evaluate eventual generalisations to make the modules less dependent from the actual core implementation 4. Quality assurance The complexity of several modules, and the special dependencies that some of them carry increase the work to test the server and reproduce bugs. Without proper test coverage maintenance work is slowed down and flexibility is lost. All "easy" bugs that get caught automatically will not reach the user, increasing the quality. proposed actions: - extend available test suite to cover more of core and modules - continue maintenance of available infrastructure for build tests (build bot, debian package autobuilder), evaluate extensions to increase coverage All this work will be done as in the past in an open and accountable way on the developer list. For bigger changes a request for comment will be posted in advance on this list. If existing functionality is deprecated, then this RFC will be also posted to the user list. If some functionality is moved or changed than this will be documented in the wiki (porting docs) as usual. I understand some changes will be probably add work on your side, in porting, configuration adjustments and the like. But i think this is necessary to ensure the further development of the Kamailio project, and to be able to deal with new challenges that will come up in the future. Any input and help with this regards to this individual tasks is of course much appreciated. In the end the success of the project depends in the future (like it did in the past) on your support. Without your involvement in contributions, discussions and bug reports i would be only to a much lesser extend involved in this project. Thank you, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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[Kamailio-Users] About Changing 401 Message
by Ali Jawad 31 Aug '08

31 Aug '08

Hi All When I do logon with a wrong username/pwd I get the following error 401 UNAUTHORIZED. If I do want to change that error "for whatever stupid reason :-)" I know I could change that message using the header files and recompile. However I only know where to change the UNAUTHORIZED keyword, what about the 401 number...apart from that could I change that message using ss_reply ? Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 31 Aug '08

31 Aug '08

Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo

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Re: [Kamailio-Users] [Kamailio-Business] project foundation
by Daniel-Constantin Mierla 31 Aug '08

31 Aug '08

Hello, On 08/13/08 08:37, Max van Biezen wrote: > Hi, > > >> a quick operational expenses estimation for the first year brought up >> the figure of ~3000Euro, that includes the expenses for creating the >> foundation, two hosted servers (one main, one backup), domain names and >> trademark registrations. The costs regarding TM registration expenses >> (US&EU) are not yet very clear to me, maybe someone can shed some light >> if he/she did it recently. >> > > I can only give an indication on EU trademark. That is 850,- one time > fee. This fee is due regardless the fact some company will complain or > not. As happened to me. > > The process can be done by yourself no lawyers needed. However there are > lawyers outhere that simply complain to everything in the hope some > company which they claim to represent willl pay for it. Anyway worst > case scenario would then be to drop the registration whihc means the 850 > is gone > > Bottom line just try it yourself. It is a long process before > registration is final (more than 6 months) > thank you for the details. Cheers, Daniel > Max van Biezen > > > -- Daniel-Constantin Mierla http://www.asipto.com

1 1

[Kamailio-Users] issue when adding subscriber to mysql
by Alex R.S.M 30 Aug '08

30 Aug '08

Hi, Ijust added mysql module to my openser 1.2 and tried to test it by adding a couple of SIP users to the database. I can verified there is only one default sub in the database: mysql> select count(*) from subscriber; +----------+ | count(*) | +----------+ | 1 | +----------+ 1 row in set (0.00 sec) mysql> show tables; I can add the first subscriber: # openserctl add 100 100 100@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 -e mysql_query -e insert -e 0 -e new user '100' added Howver, When I try to add the second subscriber I am getting the following error: # openserctl add 101 101 101@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 -e mysql_query -e insert -e 0 -e \E[37;31mERROR: introducing the new user '101' to the database failed root@bling1:/etc/openser# What might be the problem? Thanks, Alex _________________________________________________________________ See what people are saying about Windows Live. Check out featured posts. http://www.windowslive.com/connect?ocid=TXT_TAGLM_WL_connect2_082008

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[Serusers] SER Version
by Arun Kumar 30 Aug '08

30 Aug '08

Hi All, I'm planning to roll out a new production SER for my company needs suggestion on SER version. At present I run 0.9.6 and having some issue like waiting transaction, extra connection to mysql, so which version should I roll out. thanks arun

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[Kamailio-Users] openser children parameter
by toly 29 Aug '08

29 Aug '08

Greetings, What would be value of children parameter for few thousands user? Thanks, Toly -- View this message in context: http://www.nabble.com/openser-children-parameter-tp19192868p19192868.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 4

[Serusers] AVP and MySQL question
by Steven C. Blair 29 Aug '08

29 Aug '08

I have a SER 0.9.7-pre3 system. I use avopops to load flags from mysql to implement call andling features. I'd like to read a list of values for a single attribute from the database (and I can today) however the order in which the results are return is not consistent. I read that MySQL4 has an issue where results of a query are returned in a nondeterministic fashion. I think I can get my desired result by issuing my own query but I do not see an avpops function which allows for a user specified query. Does such a function exist? If so what is it's name? Thanks

2 1

[Kamailio-Users] Question regarding outbound configuration.
by Siddhardha Garige 29 Aug '08

29 Aug '08

Hello all, I am working on an outbound proxy configuration and here is my setup. UA ------> Outbound ------->firewall(NAT) -------->Proxy. Both UA and outbound are behind the NAT. I did some RFC study on outbound configuration for routing calls both inbound and outbound through outbound proxy. According to RFC 3327 Extension Header field for Registering Non-Adjacent contacts, PATH variable has to be introduced into REGISTRATIONS. Here are few questions and any help with answers is greatly appreciated. 1. Default openser.cfg will route request to destination if (URI != myself) and appends p-hint:outbound. What exactly is the significance of P-hint:outbound? http://tools.ietf.org/html/draft-ietf-sip-outbound-15 says ob parameter has to be added to PATH filed or Contact Field to retain path. P-hint got to do any thing regarding this? 2. I enabled path.so module. But openser does not at PATH filed to outgoing requests. Can someone direct me to a config example? Thanks in advance, Sid

1 1

[Kamailio-Users] event notifications and user aliases
by Alexander 'Leo' Bergolth 28 Aug '08

28 Aug '08

Hi! I'd like to implement P2P presence and extension monitoring for my SIP network. Our users register by using a numeric extension (e.g. sip:226050@wu-wien.ac.at), but most of the time, they are contacted using a user alias that looks like their email-address (e.g. sip:bergolth@wu-wien.ac.at). So the addresses stored in their buddy-lists won't be the "real"-SIP-URI of the users. Will there be a problem if I translate those aliases on subscription, so that a user may also subscribe for the presence of a user alias? If I rewrite SUBSCRIBE requests to use the "real" SIP-URI on subscription, the following NOTIFY messages will contain the wrong URI in the From-header and in the pidf body. But maybe this won't be a problem, since the watcher can assign it to the correct username because the NOTIFY belongs to the same dialog as the SUBSCRIBE? Will the user agents be able to correctly assign those notifications to their contacts? Any experiences with this? Cheers, --leo -- e-mail ::: Leo.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

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[Serusers] Call forwarding: Diversion header
by caio 28 Aug '08

28 Aug '08

Hi, Does anyone know how to implement draft-levy diversion header on for example bilnd call forwarding? I have this scenario: xlite ------ SER-0.9.7 ------ PSTN-GW ------ PSTN Case: Call from xlite-579950 to 579917 (both registered at SER), 579917 has fwdblind enabled to pstn-501866. where SER modifies INVITE when detect that user 579917 has fwdblind feature enabled. The call must be forwarded to the pstn number. but the PSTN-GW returns to SER a "302: Moved Temporarily" with a confused diversion header.. Here a few sip messages: = SER -----> PSTN-GW U +0.000824 200.2.127.187:5060 -> 200.2.127.153:5060 INVITE sip:501866@200.2.127.153:5060 SIP/2.0. Record-Route: <sip:200.2.127.187;ftag=2d0ad32d;lr=on>. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. Max-Forwards: 16. Contact: <sip:579950@192.168.132.22:35318>. To: "579917"<sip:579917@wsa.lab>. From: "579950"<sip:579950@wsa.lab>;tag=2d0ad32d. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 385. = PSTN-GW ----> SER U +0.051198 200.2.127.153:5060 -> 200.2.127.187:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. To: "579917" <sip:579917@wsa.lab>;tag=121881197764d164f2-9431-47bb-be95-8af6684ebc04. From: "579950" <sip:579950@wsa.lab>;tag=2d0ad32d. CSeq: 2 INVITE. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. Contact: <sip:579917@200.2.127.187:5060>. Diversion: <sip:501866@200.2.127.153:5060>;reason=unknown. Server: Epygi Quadro SIP User Agent/v4.1.23 (SOHOGW). Content-Length: 0. What is wrong? How the flow must be to satisfies diversion of levy draft? Any comments are welcome...i'm confused if SER must modify INVITE or "To" header or if a diversion must be set (add_diversion of diversion.so module) when SER send the first invite to PSTN-GW... Regards, -- caio

1 0

[Kamailio-Users] Why dialplan and drouting modules were removed from openser-1.3.3?
by Peter Lemenkov 28 Aug '08

28 Aug '08

Hello All! Subj. -- With best regards!

3 4

[Kamailio-Users] How to test RLS and trace messages?
by mahesh.peddi＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi, I am working on RLS Application server and integrated with SER IMS, i made changes in the configuration file with the help of Readme file. Its working fine. But how to confirm and test RLS is working. And how to trace the messages? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] Using pua_bla for BLF
by catalina oancea 28 Aug '08

28 Aug '08

Hello I am trying to make some phones work with BLF using Kamailio and I found an older messge ( http://lists.kamailio.net/pipermail/users/2007-October/013700.html) saying that I could do this using BLA. I have a Thomson ST2030 as the message says, but it doesn't work with presence and pua_bla. I get the error: ERROR:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_subscribe: event= dialog I undestood from that message that the event is supposed to be 'dialog', so what's the problem then? Thanks a lot Catalina

3 6

[Kamailio-Users] Fw: regarding 3rd party register request.
by srinivasa.reddy＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi all, This is Srinivas, i am working on OpenIMScore and Openser( as Application Server) . from few days i stuck at third party regaester request(that is interface between SCSCF and Presence ). You said that, the registration-event package is not enabled at the IMS Core side that mean in the scscf.cfg is it? If it is could you please provide where and what i have to change and i am attaching the scscf.cfg . And regarding 2nd point Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package That means openser does not support reg package or ? WITH BEST REGARDS, SRINIVAS. -------------------------------------------------------------------------------------------------------------------------------- Hi, >From your mail below, I could infer only that, the registration-event package is not enabled at he IMS Core side. Make sure, that you have done the necessary settings for that. Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package. Thanks, Noble Antony T. On Wed, Aug 27, 2008 at 3:54 PM, <mahesh.peddi(a)info-spectrum.com> wrote: Hi, I have one issue regarding openser could you kindly reply if you know . I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am connected it with IMS core .And for testing i was used UCTIMS client during the register with uctims client actually i need to get 4 messages. 1.Third party register request is going to presence server from scscf. 2.Notify messages coming from presence server to uctclient. 3.subscribe message must take route through the presence server. 4.publish message must take route through the presence server. But at present i am getting 3 and 4 messges corectly i am not getting 1 and 2. So please tell me what are the modifications or changes i need to do in any xml files or cfg files. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764 Regards, Srinivasa Reddy Kadiri Software Developer Infospectrum India Pvt. Ltd. Cell: +91 9921710062 Email: srinivasa.reddy(a)info-spectrum.com IP-Phone Ext. - 759

1 0

Re: [Kamailio-Users] How to test and trace the RLS messages?
by Daniel-Constantin Mierla 28 Aug '08

28 Aug '08

Hello, please keep cc-ing to mailing lists, there are people that can answer you quicker and better. As a matter of fact, although I have studied a bit IMS docs, that happened long time ago and don't know how those messages should be routed in IMS and where the mistake can be in your routing logic. Cheers, Daniel On 08/27/08 13:24, mahesh.peddi(a)info-spectrum.com wrote: > Hi, > > I have one issue regarding openser could you kindly reply if you know . > > I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am > connected it with IMS core .And for testing i was used UCTIMS client > during the register with uctims client > actually i need to get 4 messages. > 1.Third party register request is going to presence server from scscf. > 2.Notify messages coming from presence server to uctclient. > 3.subscribe message must take route through the presence server. > 4.publish message must take route through the presence server. > But at present i am getting 3 and 4 messges corectly i am not getting > 1 and 2. > So please tell me what are the modifications or changes i need to do > in any > xml files or cfg files. > > Regards, > Mahesh Peddi > Infospectrum India Pvt. Ltd. > Cell: +91 9765775176 > IP-Phone Ext. - 764 > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: <mahesh.peddi(a)info-spectrum.com> > Cc: "userlist" <users(a)lists.kamailio.org> > Sent: Wednesday, August 20, 2008 10:44 PM > Subject: Re: [Kamailio-Users] How to test and trace the RLS messages? > > >> Hello, >> >> On 08/20/08 09:44, mahesh.peddi(a)info-spectrum.com wrote: >>> Hi All, >>> I am working on openser 1.3.2 version with RLS as >>> "integrated_xcap_server". Its running without erros. >>> But my query is how to test, And please send me call flow between >>> IMS components and RLS server. >> perhaps you can find some hints in IMS specifications. You need in >> first place a sip phone with rls support, afaik, there are not that >> many. >> >> Cheers, >> Daniel >> >>> Regards, >>> Mahesh Peddi >>> Infospectrum India Pvt. Ltd. >>> Cell: +91 9765775176 >>> IP-Phone Ext. - 764 >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] openser v1.3.3. released
by Daniel-Constantin Mierla 27 Aug '08

27 Aug '08

Hi everybody, OPENSER v1.3.3 is out - it is a minor release of the branch 1.3, that includes the fixes since v1.3.2 committed so far- configuration file and database compatibility is preserved. OPENSER 1.3.3 is based on the latest version of branch 1.3, therefore those running 1.3.2 or earlier version in this branch are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/openser/1.3.3/src/ Detailed changelog: http://www.kamailio.org/pub/openser/1.3.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.3 openser Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.3.3/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] Radius Auth : 407 proxy auth and Get-User-Attr - crash
by Samuel Muller 27 Aug '08

27 Aug '08

Hello all, i'm using SER 2.0 with Radius accounting and authentication - based on x64 ubuntu machine. i've a problem with the Radius auth, and only for the proxy auth mechanism (method INVITE) : depending of the topology of the UAC, it crashes the SER or a there's a 407 proxy auh required. when the UAC is behind a simple NAT, i get the 407, with "preauth: credentials with realm 'sip.720.fr' not found" - i tried many configs, no way it doesn't work. but the REGISTER is ok - it's the same Radius request ? when the UAC is behind NAT but in a VPN (our customers), proxy auth is successfull even if the preauth don't find the credentials with the realm (why? it does not differ from another uac without VPN ???), but when SER requests the Radius to get the user attributes, it crashes ! in each case, there's no pbm with REGISTER requests. Radius user sample : 1010(a)sip.720.fr User-Password = "pwd" User-Name = "1010", SER-UID = "1010(a)sip.720.fr", SER-DID = "1", SER-Digest-Username = "1010", SER-Digest-Realm = "sip.720.fr", SER-Uri-User = "1010(a)sip.720.fr" Does anyone have any idea, or met already this problem ? What I have to verify ? Thanks for help, Samuel MULLER

1 0

[Kamailio-Users] multiple extensions/domains
by voipmo 27 Aug '08

27 Aug '08

I have been running asterisk servers for a few years now to host multiple organizations. I have been working on setting up OpenSER so that we can have failover and load balancing. My question is how do I keep my different organizations separate? I currently have the following registrations: 1001(a)test.example.com 1002(a)test.example.com 1001(a)another.example.com The problem is that if 1002(a)test.example.com dials 1001 it rings both domains 1001 phones. Am I going about this the right way? Is there a better method? How might I prevent calls between domains? -- View this message in context: http://www.nabble.com/multiple-extensions-domains-tp19173400p19173400.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Serusers] Help with Ser and LCR
by Jai Rangi 26 Aug '08

26 Aug '08

All, I need little help with LCR module in ser, Say these are the entries on my LCR tables, Prefix, | From_URI | Grp_id | Priority "13612499080" ;"%"; "46"; "1" "1" ;"%"; "1"; "1" "1" ;"%" "2" "2" Now any Say if someone calls 19495556666, Then ser sends the call to Server in Grp_id 1, if the grp_id=1 server does not respond, then SER sends the call to Server with grp_Id 2. Perfect, that exactly what I need. Now if some calls 13612499080", then SER will send the call to server with grp_id 46. If the call failes it should not send the call to server grp_id 1 or two. Right? In my case it is sending the call to server grp_id 1 after the call fails on server grp_id 46. How can I stop that. I want the calls for 13612499080 forwarded to only server grp_id 46 and is that fails then just terminate the call. I will appreciate if someone can provide any help or hint. Thank you, -JP

1 1

[Kamailio-Users] AVP to store user contact information
by Asim Riaz 26 Aug '08

26 Aug '08

Hi All, I am wondering if there is a way to store the user contact info into avp, relay it to existing Regitrar and store the original contact info into location through a function(which i dont know) once i received successful reply from my sip registrar. Thanks in advance Regards Asim Riaz

5 17

[Kamailio-Users] FW: | SMS FALAS | <--- did anyone else on this list get this message?
by Vadim Berezniker 26 Aug '08

26 Aug '08

Sorry this is offtopic but.. Did anyone else get this email? I'm just wondering if they scraped emails from this mailing list. -----Original Message----- From: KLIENT I NDERUAR [mailto:info@hi5.com] Sent: Tuesday, August 26, 2008 11:38 AM To: Vadim Berezniker Subject: | SMS FALAS | Klient i/e Nderuar Nese Deshironi te Ndegjoni Muzika ma te rejat ather vizitone Kosova-SMS dhe do te bindeni me muziken qe ofrojm ne. Gjithashtu per Vizitoret e Kosova-SMS ka Startu me ofert te RE. SMS FALAS MUNDSUAR NGA Kosova-SMS 60 SMS NE DIT NGA NR I YT KLIKO PER ME SHUM. --> http://www.kosova-sms.tk

1 0

[Serusers] Notice call ending
by inge 26 Aug '08

26 Aug '08

Hi all, I look for a workarround to notice the caller that the call will be hang up, by generating a beep. Is someone can help me because I don't find anything about this subject ? Thanks. Regards, Adrien.

1 0

[Kamailio-Users] Having Problem routing an ACK
by Stagg Shelton 26 Aug '08

26 Aug '08

Hello, I am having an issue routing an ACK back from a new upstream carrier. The basic situation is my openser 1.3.2 receives the INVITE. I do an alias_db_lookup, set some accounting flags, and send the call to t_relay The call makes it to the PBX, as expected, and the PBX sends a 100 trying, 180 ringing. The 180 ringing is sent to the carrier. When the incoming call is answered the PBX sends a 200OK to openser, and that 200OK is sent to the carrier. When the carrier sends the ACK back it makes it to OpenSER, but is not sent to the PBX. The PBX will send several more 200 OK attempts, but each ACK from the carrier stops at openser. From what I can tell from the wireshark packet captures, the Request- Line header in the ACK from the carrier is sip:TELNUMBER@MY_OPENSER_IPADDR I have compared these packets to similar traces from other carriers and the Request-Line header from the other carriers is sip:TELNUMBER@MY_PBX_IPADDR This new carrier is telling me that they are not doing anything wrong with the ACK. RFC3261 seems specific about the construction of the ACK, but at the same time i am getting lost going to all of the other sections trying to identify details. Basically I just need to get the ACK to the PBX. What other information can I provide with this problem. Thanks Stagg Shelton

4 20

[Serusers] ser 0.10 doucmentation
by Steven C. Blair 26 Aug '08

26 Aug '08

I was wondering if anyone has documentation on the new syntax used in SER 0.10.x. I see the following in the ser-advanced.cfg example file but I do not see these parameters documented anywhere. Thanks,Steve if ($t.fr_inv_timer) { if ($t.fr_timer) { t_set_fr("$t.fr_inv_timer", "$t.fr_timer"); } else { t_set_fr("$t.fr_inv_timer"); } } Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

6 32

[Kamailio-Users] CDRTool not applying rates
by Brian Del Shasta 25 Aug '08

25 Aug '08

I believe my rating tables are configured correctly now thanks to David Villasmil who sent me examples from his working setup. However, I am still unable to see any rates. The rating engine seems to be started and I have no problem identifying and matching calls based on specific criteria. But I still see no rates. I'm wondering if anyone knows the best place to troubleshoot the rating engine would be or if anyone has any other ideas? Thank you in advance, and thanks again David for your help!

3 7

Re: [Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

in this guide, he doesn't have the module presence_xml, but he has the parameter modparam (" presence ", " force_active ", 1) http://www.kamailio.net/dokuwiki/doku.php/presence:configuration-file it is strange, I believe to be seen in the wiki a same example for openser 1.3.2 greetingss rickygm

2 1

Re: [Serusers] call forward no answer (solved)
by caio 25 Aug '08

25 Aug '08

append_branch() is missed at ser examples... On Fri, Aug 15, 2008 at 4:22 PM, caio <elcaio(a)gmail.com> wrote: > Hi all, > I can't make call forwarding on no answer works on my ser-0.9.7 installation. > Here you can see a snippet of log, where the call falls on > failure_route and then to route_1 where t_relay is called. > > Log at: http://rafb.net/p/xwTmn195.html > > The call is from sip:579951 to sip:579915(who has enabled the feature > fwdnoanswer to sip:579916. > But I do not know why the INVITE after the append_branch() is still to > R-URI: sip:579915@.... > > After the relay then the call terminates with: > > Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ > Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ > Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ > Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ > > I follow the example of ser guide.., but do not work.. > I would appreciate any comment.. > > Regards, > > -- > caiogf > -- caio

1 0

[Serusers] ser + media server
by caio 25 Aug '08

25 Aug '08

Hi all, What do you recommend for media server application to integrate wih ser-0.9.x? SEM or Asterisk? Specifically i'm talking about running services such as IVR, conference and voicemail.. Thanks your suggestions.. -- caio

1 0

[Kamailio-Users] crashing problem
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

We're using Carrier route with about 53,000 routes stored in a mysql databse. We're getting this crash in production that we never had in the lab. The bt is below. Does anyone have any suggestions? nline static void free_to_params(struct to_body* tb) { struct to_param *tp=tb->param_lst; struct to_param *foo; while (tp){ foo = tp->next; <-- **** Line 75 ***** pkg_free(tp); tp=foo; } } -Jonathan #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 (gdb) backtrace full #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 No locals. #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 __FUNCTION__ = "clean_hdr_field" #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 No locals. #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 No locals. #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 msg = (struct sip_msg *) 0x8193030 __FUNCTION__ = "receive_msg" #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 len = 1271 buf = "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"... tmp = <value optimized out> from = (union sockaddr_union *) 0x8a91318 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {2850040017, 135426432, 3213031528, 3085483479}, addr32 = {2850040017, 135426432, 3213031528, 3085483479}, addr16 = {10449, 43488, 29056, 2066, 63592, 49026, 48599, 47080}, addr = "Ñ(à©\200q\022\bhø\202¿×½è·"}}, dst_ip = {af = 2, len = 4, u = {addrl = {509300942, 0, 0, 0}, addr32 = {509300942, 0, 0, 0}, addr16 = {20686, 7771, 0, 0, 0, 0, 0, 0}, addr = "ÎP[\036", '\0' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto = 1, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\023ÄÑ(à©\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 2850040017}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 2850040017, sin6_addr = {in6_u = {u6_addr8 = '\0' <repeats 15 times>, u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x818dd90} __FUNCTION__ = "udp_rcv_loop" #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 cfg_log_stderr = 0 cfg_stream = (FILE *) 0xa179008 c = <value optimized out> r = <value optimized out> tmp = 0x8148b9c "ÿÿÿÿ" tmp_len = <value optimized out> port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 639236262 rfd = 4 __FUNCTION__ = "main"

4 9

Re: [Kamailio-Users] Google reference to openser users and devel Archives
by H Quintana 25 Aug '08

25 Aug '08

Hi, My temporary solution was to edit the etc/hosts file in my windows xp machine. The path is: C:\WINDOWS\system32\drivers\etc\hosts And I added the lines: 193.22.119.66 www.openser.org 193.22.119.66 openser.org to point to the Kamailio servers. Hope this helps, Humberto > Subject: Re: [Kamailio-Users] Google reference to openser > users and > devel Archives > To: users(a)lists.kamailio.org > Message-ID: > <200808251326.44077.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="utf-8" > > On Friday 15 August 2008, Rafael Risco Gonzales-Vigil > wrote: > > This is for openser/kamailio and opensips list > administrators: > > google still has references to > http://www.openser.org/pipermail/users/ and > > http://www.openser.org/pipermail/devel/ for openser > related searchings and > > those urls does not contains old files anymore so its > dificult search > > information for learning and troubleshooting about > openser/kamailio or > > opensips... I hope someone can solve this issue for > the benefit of all > > current and future community users. > > Hi Rafael, > > we don't have access to the openser.org domain, so > there is nothing what we > can do here, sorry. All access of web and mail services to > openser.org will > go to opensips infrastructure. > > If you encounter such old links with openser.org, you can > simply > change "openser.org" to "kamailio.org", > then you'll be able to access the > content, apart from using the google archives, as Alex > pointed out. > > Cheers, > > Henning >

2 1

[Kamailio-Users] failure route configuration
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

I have this block in my configuration: <excerpt from failure_route[1]> if(!cr_rewrite_uri("1", "call_id")){ t_reply("403", "$fU@$fd Not allowed"); } else { # In case of failure, re-route the request prefix("+"); t_on_failure("2"); if (is_method("INVITE")) t_on_reply("1"); # Relay the request to the gateway t_relay(); } The output in my log file: Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5622]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_one Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: Failure Route Started Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_two Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:t_forward_nonack: no branch for forwarding Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:w_t_relay: t_forward_nonack failed What am I missing that I need to do prior to calling t_relay()? -Jonathan

6 11

[Kamailio-Users] Google reference to openser users and devel Archives
by Rafael Risco Gonzales-Vigil 25 Aug '08

25 Aug '08

This is for openser/kamailio and opensips list administrators: google still has references to http://www.openser.org/pipermail/users/ and http://www.openser.org/pipermail/devel/ for openser related searchings and those urls does not contains old files anymore so its dificult search information for learning and troubleshooting about openser/kamailio or opensips... I hope someone can solve this issue for the benefit of all current and future community users. rafael Lima - Peru

3 2

[Kamailio-Users] need help on starting opensips v1.4.1
by Do Nguyen Ha 25 Aug '08

25 Aug '08

hi opensips debian:/etc/opensips# /etc/init.d/opensips start Starting opensips: opensipsListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.254 [192.168.1.254]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.254 [192.168.1.254]:5060 Aliases: tcp: debian:5060 tcp: localhost:5060 udp: debian:5060 udp: localhost:5060 . debian:/etc/opensips# /etc/init.d/opensips status Status of opensips: opensips is not running. could anyone tell me the tips for this Thanks Ha`

2 1

[Kamailio-Users] Why does Kamailio reply 408 when 'fr_inv_timer' expires?
by Iñaki Baz Castillo 25 Aug '08

25 Aug '08

Hi, when 'fr_inv_timer' expires Kamailio CANCELs the pending branches and reply "408 Request Timeout" to the UAC. Well, I wonder if 408 is appropiate. I think "480 User Unavailable" or "487 Cancelled" would be better: I received those replies in other maillist: "408 indicates transaction timeout. You have sent request to next hop and it has not responded back. If the gateway has not responded at all (Including 100 trying), 408 is appropriate. 480 is more of application usage meaning that the user is not available and would be appropriate if the gateway has at least send one provisional response i.e. 100, 180, 183..." "408 is used consistently to mean IP-level timeout in sending a request from one element to another. It is also defined to be returned by a UA when the time specified in "Expires" has been reached. (In theory, two different error codes should be defined for these two situations, but that has not been done.) However, in a ring-no-answer situation, it is more common for an upstream proxy to send a CANCEL to the UA/gateway, in which case the UA/gateway provides a 487 response to the INVITE." This last point suggest that Kamailio could do the following: - If the proxy has received provisional responses from callee's then not send a 408 upstream since when the proxy sends CANCEL to UAS's it will receive some 487 and would return a 487 upstream. - If the proxy hasn't received provisional responses then this is a clear "408 Request timeout". So a possibility is that Kamailio just reply a 408 in real timeouts and reply 487 when 'fr_inv_timer' expires (or just reply nothing since UAS's will generate the 487). Opinions? -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 2

[Kamailio-Users] Where have the OLD Module Docs gone?
by Douglas Garstang 25 Aug '08

25 Aug '08

The module documentation for versions 1.2, 1.3 etc used to be available on the web site. Where has it gone? When you clicked on the documentation link, you used to be able to choose the version of OpenSER(Kam...OpenSIPS whatever) that you wanted. Now, your taken immediately to what appears to be the documentation for the LATEST version. Doug.

7 7

[Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

Hi, lists I have some problems with the simple presence in kamailio 1.4, I add to my kamailio.cfg the following line and when restarting the it shows me errors in this it lines Aug 24 19:31:41 [21946] ERROR:core:set_mod_param_regex: parameter <force_active> not found in module <presence> Aug 24 19:31:41 [21946] CRITICAL:core:yyerror: parse error in config file, line 50, column 19-20: Can't set module parameter Aug 24 19:31:41 [21946] ERROR:core:main: bad config file (1 errors) Aug 24 19:31:41 [21946] NOTICE:presence:destroy: destroy module ... I remove of the Makefile the modul presence kamailio.cfg loadmodule "presence.so" modparam("presence|auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:192.168.10.1:5060") where the error can be? regards rickygm

2 1

[Kamailio-Users] preparing openser 1.3.3
by Daniel-Constantin Mierla 24 Aug '08

24 Aug '08

Hello, I propose to build a new minor release for openser 1.3 branch on the 27th of August (Wednesday). There are already a lot of fixes since 1.3.2. Probably couple of backports from 1.4 will get in until Wednesday, the rest, if it is the case, shall get in 1.3.4, probably in about another month, as we are going to maintain 2 stable branches. Perhaps 1.4.1 will be out by same time or so. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] FreeRadius patch for OpenSer1.2.2
by Nuno Marques 24 Aug '08

24 Aug '08

Hi, I'm following the examples on the book "Building Telephony Systems with OpenSer". I came to Accounting and Billing chapter using FreeRadius. Flavio (the author) says we need to apply a patch to the freeradius to "fix a problem with accounting type 15". My question is - is it still necessary to apply that patch or the new version of freeradius allready covers it? I'm having big problems applying that patch because of the key signature problems. Appreciate any help in this subject. Thanks in advance, Nuno

5 5

[Kamailio-Users] AUTO: Miodrag Radulovic is out of the office (returning 07-09-2008)
by Miodrag Radulovic 23 Aug '08

23 Aug '08

I am out of the office until 07-09-2008. Privremeno nedostupan - Out-of-Office Message. Poruka koju ste poslali je prispela i bice sacuvana, ali napominjem da sam na odmoru van zemlje do 07.09.2008. i moj pristup elektronskoj posti bice veoma ogranicen. Ukoliko se radi o potrebi za hitnom intervencijom, dostupan sam preko mobilnog telefona 065/3016576. The message that you have sent will be saved in my inbox, but please note that due to summer holliday I will be out of the country until 07.09.2008. In that period my access to my email will be very limited. In case of urgency, I am available on my mobile phone: +381 65 3016576. Miodrag Radulovic General Manager COMUTEL Note: This is an automated response to your message "Users Digest, Vol 39, Issue 83" sent on 23.8.2008 0:49:18. This is the only notification you will receive while this person is away.

1 0

Re: [Serusers] SER - Waiting Transaction
by Arun Kumar 22 Aug '08

22 Aug '08

Hi, thanks for your reply. I'll ignore this but I see so many open connection to MySQL from SER and they don't get close automatically also resulting in denial of other service using my MySQL. Thanks arun On Mon, Aug 18, 2008 at 10:12 PM, Jiri Kuthan <jiri(a)iptel.org> wrote: > ignore it > > Arun Kumar wrote: > >> Hi, >> >> >> I'm running SER 0.96 in my production for a long but there is strange >> problem , when I do serctl moni and looks for transaction it shows waiting >> transaction in () almost 400+ can some one tell me why there are so many >> transaction in waiting while I see processing only 10-20 transaction. >> >> Thanks >> arun >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >

4 5

[Kamailio-Users] Problem to run MediaProxy 2.0.3
by Leonardo Santana 22 Aug '08

22 Aug '08

Hi all, I´m using mediaproxy 2.0.3 on OpenSuSE 11.0. When I run the "media-dispatcher" I see this message bellow: PS.: The OpenSuSE 11.0 doesn´t have the module "libgnutls-extra.so.13" only the libgnutls-extra.so.26". ######## Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Log opened. Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Starting MediaProxy Dispatcher 2.0.3 Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Traceback (most recent call last): Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/media-dispatcher", line 46, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.dispatcher import Dispatcher Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/dispatcher.py", line 31, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.tls import X509Credentials, X509NameValidator Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/tls.py", line 12, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls import crypto Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/crypto.py", line 11, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.validators import method_args, none, ignore, list_of, one_of Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/validators.py", line 9, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.constants import * Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/constants.py", line 38, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import constants Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/__init__.py", line 7, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import errors Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/errors.py", line 14, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library.functions import gnutls_strerror, gnutls_alert_get Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/functions.py", line 6, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] _libraries['libgnutls-extra.so.13'] = CDLL('libgnutls-extra.so.13') Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/ctypes/__init__.py", line 348, in __init__ Aug 21 17:18:32 nana media-dispatcher[5502]: [-] self._handle = _dlopen(self._name, mode) Aug 21 17:18:32 nana media-dispatcher[5502]: [-] OSError: libgnutls-extra.so.13: cannot open shared object file: No such file or directory ##### Thanks a lot!!!

2 1

[Kamailio-Users] RFC: branches/location records control
by Daniel-Constantin Mierla 22 Aug '08

22 Aug '08

Hello, one of the most hidden parts of openser to the config is the branch processing. Behind a branch is following structure: struct branch { char uri[MAX_URI_SIZE]; unsigned int len; /* Real destination of the request */ char dst_uri[MAX_URI_SIZE]; unsigned int dst_uri_len; /* Path vector of the request */ char path[MAX_PATH_SIZE]; unsigned int path_len; int q; /* Preference of the contact among contact within the array */ struct socket_info* force_send_socket; unsigned int flags; }; I am thinking in offering a fine access to this structure in configuration file, so the admin can control much closer the branch processing. By using while and and index, checks of the attributes in branches could be done. By doing it in main route, one can reduce the overload of stateful processing, by dropping a branch before going to transaction, for example. I am looking at something similar for user location records. Such a record has many attributes, and from config is no way to access them without doing a lookup and changing r-uri/branches. There could be many benefits, for example: - being able to detect if a user calls from a registered phone or not - detecting how many contacts a user has at a moment - ability to check and, eventually, drop processing before doing changes to branches and so on... The access will be provided via pseudo-variables, perhaps the location records will be fetched in variables upon calling a function that gets an AoR as parameter. I have seen on mailing lists discussions about needs that will be solved by this new feature. Any comments? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 7

[Kamailio-Users] kamailio-1.4.0 and SerMyAdmin
by luzango mfupe 22 Aug '08

22 Aug '08

Hi, Im bit cnfsd, does SerMyAdmin support Kamailio-1.4.0?. I did my configuration using SerMyAdmin 0.7-Beta.war and then 0.7-beta.war files, but everytime i pointed my browser to the IP address of my installation my Tomcat6 server return the message that SerMyAdmin does not exist!!! Can someone pls'e clarify this?? .. wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 4

[Kamailio-Users] openser 1.2.1 errors in the log
by toly 22 Aug '08

22 Aug '08

Greetings, Proxy compiled with TLS support, tls is not turned on Routing script inclulded below. Underlying database is Sybase. The following is happening: There are 25 users having as a contact each other. Softphone is Bria. Without any calls, just subscribe/notify proxy is running out of the socket in one hour. The users are or outside coming via corp firewall, or if inside, the route is setup the way the going via corp firewall, so they are natted. Proxy is in DMS on the same network as firewall. There are no problems with nat traversal. Daniel, could you please take look at the script? Thanks a lot, Toly. this is from log: <receive.c: 206> receive_msg: cleaning up 08/15 20:12:58 29156 debug <ip_addr.c: 109> tcpconn_new: new tcp connection to: xxx.xxx.xxx.xx 08/15 20:12:58 29156 debug <tcp_main.c: 402> tcpconn_new: on port 58544, type 2 08/15 20:12:58 29156 debug <tcp_main.c: 497> tcpconn_add: hashes: 96, 45 08/15 20:12:58 29156 debug <tcp_main.c: 993> handle_new_connect: new connection: 0xb6485590 98 flags: 0002 08/15 20:12:58 29156 debug <tcp_main.c: 935> WARNING: send2child: no free tcp receiver, connection passed to the least busy one (1) this is routing script: log_facility=LOG_LOCAL2 debug=7 fork=yes log_stderror=no listen=PROXY_ENV # This is where OpenSER installed port=5070 children=20 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true disable_tls = 1 listen = tls:PROXY_ENV:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = SSLv23 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" ###########################################################33 ###########################################################33 ###########################################################33 mpath = "/usr/local/lib/openser/modules" loadmodule "uac.so" loadmodule "psybase.so" loadmodule "pike.so" modparam("pike", "remove_latency", 120) modparam("pike", "reqs_density_per_unit", 10000) modparam("pike", "sampling_time_unit", 60) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0666 ) modparam("mi_fifo", "reply_dir", "/tmp/" ) modparam("mi_fifo", "reply_indent", "\t" ) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "avpops.so" modparam("avpops", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "uri.so" loadmodule "uri_db.so" modparam("uri_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") loadmodule "alias_db.so" modparam("alias_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list") #modparam("dispatcher", "dst_avp", "$avp(i:271)") #modparam("dispatcher", "grp_avp", "$avp(i:272)") #modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "flags", 2 ) # failover mode set flag=2 loadmodule "usrloc.so" modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "acc.so" # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1 ) #modparam("acc", "log_flag", 0 ) modparam("acc", "db_flag", 1 ) modparam("acc", "log_missed_flag", 1 ) modparam("acc", "db_missed_flag", 1 ) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("acc", "log_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "report_cancels", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "domain.so" modparam("domain", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "mediaproxy.so" modparam("mediaproxy","natping_interval", 10) modparam("mediaproxy","mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "received_avp", "$avp(i:801)") modparam("nathelper", "sipping_from", "sip:pinger@parusivoip.net") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", 7) loadmodule "registrar.so" modparam("registrar", "received_avp", "$avp(i:801)") modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 50) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "permissions.so" modparam("permissions", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("permissions", "db_mode", 1) ######################################################################## ######################################################################## ######################################################################## route { if ( is_method("REGISTER") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON REGISTER on IP=$si !!"); exit; } }; if ( is_method("INVITE") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON INVITE on IP=$si !!"); exit; } }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; xlog("L_NOTICE", "TRACKING [0] NEW REQUEST - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); force_rport(); # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && nat_uac_test("19")) { record_route_preset("PROXY_ENV:5070;nat=yes"); # This is where OpenSER installed xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE PRESET- M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else if (method!="REGISTER") { xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); record_route(); }; append_hf("P-hint: rr-enforced\r\n"); if ( search("User-Agent: Parus.*") && is_method("BYE|REFER")) { # check the transport of callee $avp(i:44) = $(ru{uri.user}); if ( $avp(i:44) != $tU ) { $avp(i:45) = $avp(i:44); } else { $avp(i:45) = $tU; } avp_db_query("exec oserIsTcpTransport @to_user='$avp(i:45)', @td='$td'", "$avp(i:11)"); xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si AVP11=$avp(i:11) AVP44=$avp(i:44) TD=$td"); if ( $avp(i:11) == "tcp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tcp"); } else if ( $avp(i:11) == "tls" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TLS - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tls"); } else if ( $avp(i:11) == "udp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS UDP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=udp"); } else { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS NOT TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); } } # ----------------------------------------------------------------- # ACC Section # ----------------------------------------------------------------- if ( is_method("CANCEL|BYE|INVITE|MESSAGE") ) { setflag(1); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if ( is_method("BYE|CANCEL") ) { xlog("L_NOTICE", "TRACKING [0] CALL TEAR DOWN: END_MEDIA_SESSION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); if (!has_totag()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if ( is_method("INVITE") ) { if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorized - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); }; }; if ( (nat_uac_test("19") || search("^Route:.*;nat=yes")) && is_method("INVITE|REFER|ACK") ) { setbflag(6); # NONONO fix_nated_contact() xlog("L_NOTICE", "TRACKING LOOSE_ROUTE using media proxy - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); use_media_proxy(); }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if ( !is_uri_host_local() ) { if ( is_from_local() || allow_trusted() ) { xlog("L_NOTICE", "TRACKING[0] IS_FROM_LOCAL OR TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else { xlog("L_NOTICE", "TRACKING[0] NOT IS_FROM_LOCAL OR NOT TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Forbidden"); exit; } }; if ( is_method("CANCEL|BYE") ) { route(4); exit; } else if ( is_method("ACK") ) { route(1); exit; } else if ( is_method("INVITE") ) { route(3); exit; } else if ( is_method("REGISTER") ) { route(2); exit; } else if ( is_method("OPTIONS") ) { route(11); exit; } else if ( is_method("MESSAGE") ) { xlog("L_NOTICE", "TRACKING [14] ROUTE14 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); route(14); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; xlog("L_NOTICE", "TRACKING[0] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { xlog("L_NOTICE", "TRACKING [1] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); if(isbflagset(6)) { xlog("L_NOTICE", "TRACKING [1] USING NATed reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("1"); } else { xlog("L_NOTICE", "TRACKING [1] USING Standard reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("2"); } if (!t_relay()) { xlog("L_NOTICE", "TRACKING [1] t_relay() Failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; exit; } # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- route[2] { sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); setbflag(7); xlog("L_NOTICE", "TRACKING [2] Fixed nated register - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); } if (!save("location")) { xlog("L_NOTICE", "TRACKING [2] Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); sl_reply_error(); }; xlog("L_NOTICE", "TRACKING [2] Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { xlog("L_NOTICE", "TRACKING [3] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING [3] proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING [3] spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); }; if (client_nat_test("3")) { fix_nated_contact(); setbflag(6); xlog("L_NOTICE", "TRACKING [3] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; if ( alias_db_lookup("dbaliases") ) { xlog("L_NOTICE", "TRACKING [3] callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } if ( !is_domain_local("$rd") ) { xlog("L_NOTICE", "TRACKING [3] call for foreign domain M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(5); route(1); exit; } if ( does_uri_exist() ) { xlog("L_NOTICE", "TRACKING [3] callee was local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (lookup("location")) { xlog("L_NOTICE", "TRACKING [3] LOOKUP LOCATION SUCC - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); append_hf("P-hint: usrloc applied\r\n"); remove_hf("Alert-Info"); if ( proto==tls ) { append_hf("Alert-Info: 1\r\n"); } route(5); t_on_branch("1"); route(1); exit; } }; xlog("L_NOTICE", "TRACKING [3] SOFTPHONE LOOKUP LOCATION NOT SUCC M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); # # call pstn handler # if ( route(7) ) { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler successful M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # will never hit it } else { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler failed M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # exit for now } # # never hit this for now, possibly in the future if # forward to different domain or so # xlog("L_NOTICE", "TRACKING[3] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); exit; } # ----------------------------------------------------------------- # PSTN call section # ----------------------------------------------------------------- route[7] { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (uri =~ "^sip:011.*@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION ATTEMPT TO MAKE INTERNATIONAL CALL M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); }; if ($(tU{s.len}) > 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION MORE THEN 11 DIGITS M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); } if ($(tU{s.len}) == 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION STRIP ONE DIGIT M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); } if ( ($(rU{s.len}) != 10) && ($(rU{s.len}) != 2) && ($(rU{s.len}) != 3) && ($(rU{s.len}) != 4) && ($(rU{s.len}) != 1) ) { xlog("L_NOTICE", "TRACKING [37] S.LEN != 10 or 1 or 2 or 3 or 4 M=$rm RURI=$ru F=$fu T=$tu rU=$rU IP=$si ID=$ci"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [7] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; if (uri =~ "^sip:[2-9]{1}[0-9]{2}[2-9]{1}[0-9]{2}[0-9]{4}@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN OUTBOUND $fU -> $ruri "); avp_db_query("exec oserGetCallIdForBvox @fU='$fU', @from_domain='$fd'", "$avp(i:12)"); uac_replace_from("$avp(i:12)"); xlog("L_NOTICE", "TRACKING [37] REPLACED FROM PSTN avp = $avp(i:12[1]) $fU -> $ruri "); route(5); # select next gateway simulates NEXTONE # 1 - DEV, 2 - PROD asterisk, 3 - john's nextone, 4 - prod nextone, 5 - qa ds_select_domain("GWGROUPID", "4"); t_on_failure("2"); xlog("L_NOTICE", "TRACKING [37] DISPATCHER FORWARD PSTN $fu -> $ruri "); route(1); return (1); # will never hit this } else { xlog("L_NOTICE", "TRACKING [37] Attempt to dial invalid number $fu -> $tu"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [37] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; } # ----------------------------------------------------------------- # DEV / TEST section # ----------------------------------------------------------------- route[6] { xlog("L_NOTICE", "TRACKING [6] DEV / TEST SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # CANCEL or BYE handler # ----------------------------------------------------------------- route[4] { xlog("L_NOTICE", "TRACKING [4] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (client_nat_test("3")) { fix_nated_contact(); xlog("L_NOTICE", "TRACKING [4] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; route(1); } # ----------------------------------------------------------------- # NAT traversal section # ----------------------------------------------------------------- route[5] { xlog("L_NOTICE", "TRACKING [5] NAT TRAVERSAL SECTION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (isbflagset(6)) { if (!isflagset(8)) { setflag(8); xlog("L_NOTICE", "TRACKING [5] BFLAG 6 IS SET, USING MEDIA PROXY - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); avp_printf("$avp(i:450)", "$dd"); if (avp_check("$avp(i:450)", "eq/$src_ip/g")) { xlog("L_NOTICE", "TRACKING [5] LOOKUP SUCC, FLAGS 6 AND 7 BEHIND SAME NAT $fu -> $tu $dd $src_ip"); } use_media_proxy(); }; }; } # ------------------------------------------------------------------------ # OPTIONS Message Handling # ------------------------------------------------------------------------ route[11] { xlog("L_NOTICE", "TRACKING [11] OPTIONS - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } xlog("L_NOTICE", "TRACKING [11] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("200", "Got it"); exit; } # ----------------------------------------------------------------- # NAT reply route # ----------------------------------------------------------------- onreply_route[1] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] NAT REPLY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); if (nat_uac_test("1")) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] CLIENT_NAT_TEST FIXED NATED CONTACT M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); fix_nated_contact(); }; # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- $rs $rr\n" ); } if ( isbflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { #if (!search("^Content-Length:\ +0")) { if ( $cl > 0 ) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] USE MEDIA PROXY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); use_media_proxy(); }; }; exit; } # ----------------------------------------------------------------- # standard reply route # ----------------------------------------------------------------- onreply_route[2] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[2] STANDARD REPLY- M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- $rs $rr\n" ); } exit; } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[1] { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(t_check_status("408|500|503")) { if ( t_check_status("408") ) { if ( ds_next_domain() ) { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] TRYING NEXT GW - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_on_failure("1"); route(1); return; } } else { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] NOT 408 STATUS - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_reply("503", "Service not available"); return; } } } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[2] { # If fr_timer expires t_check_status("408") is true, although $rs is <null> if( t_check_status("408") ){ xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- TIMEOUT for Gateway $rd\n" ); } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- $rs reason $rr\n" ); } # basically what we do is # 1. not worry about reasonable failures (ie busy, wrong number etc...) # 2. go to next destination gateway # 3. if no destination gateway then 503 (service unavailable) # 403 -- typically ISDN network says 'not a valid number' etc.. if( t_check_status("403") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Forbidden -> ISDN Cause Code 1\n" ); end_media_session(); return; } # 408 -- timeout -- typically the end party has not answered # Since we cancel t_on_failure() on a provisional response we should not be # getting a 408 timeout from a gateway at this stage.. it will just "fall through" #if( t_check_status("408") ){ # xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Timeout\n" ); # return; #} # 486 -- User Busy if( t_check_status("486") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Destination Busy\n" ); end_media_session(); return; } # 487 -- Request Cancelled (usually in response to a CANCEL transaction) if( t_check_status("487") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Request Cancelled\n" ); end_media_session(); return; } # ok... so at this stage we try the next gateway (unless we don't have one) # if no next gateway we bail... if( ds_next_domain() ){ #t_on_reply("1"); t_on_failure("2"); xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci Next gateway $fU -> $tU via $rd\n" ); route(1); return; } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci No more gateways -> 503.\n" ); end_media_session(); t_reply("503", "Service unavailable -- no more gateways" ); return; } } # ----------------------------------------------------------------- # BRANCH route # ----------------------------------------------------------------- branch_route[1] { if (isbflagset(6)) { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } else { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as not NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } } # ------------------------------------------------------------------------ # MESSAGE Message Handling # ------------------------------------------------------------------------ route[14] { if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp19007938p19007938.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 4

[Kamailio-Users] openser 1.2.1 errors in the log - tcp blocking
by toly 22 Aug '08

22 Aug '08

Greetings, This is openser 1.2.1 compiled with TLS, TLS is not turned on. This is part of the config file: listen=xxx.xxx.xxx.xxx # This is where OpenSER installed port=5070 children=25 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true here's the problem: I have 25 users having each other as a contact. Softphone is Bria, transport is tcp here comes subscribe: 08/21 15:07:46 8029 debug <loose.c: 115> find_first_route: No Route headers found 08/21 15:07:46 8029 debug <loose.c: 883> loose_route: There is no Route HF 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [18] - M=SUBSCRIBE RURI=sip:dmarkiew@tamtam.com F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100 .15 ID=YTBmOTJhNGI5M2RmZjYyNmM2NGFhMmI1NmRjMmY4ZWE. 08/21 15:07:46 8029 debug <dset.c: 445> rewrite_uri: Rewriting Request-URI with 'sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26' 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 0 flags C0 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 1 flags C0 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-URI=sip:76.222.53.11 2:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] USING NATed reply - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-UR I=sip:76.222.53.112:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 debug <t_lookup.c: 1037> DEBUG: t_newtran: T on entrance=0xffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=ffffffffffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=78 08/21 15:07:46 8029 debug <t_lookup.c: 407> t_lookup_request: start searching: hash=41567, isACK=0 and later I get this: 08/21 15:07:48 8029 err <tcp_main.c: 291> ERROR: tcp_blocking_connect: timeout 2 s elapsed from 2 s 08/21 15:07:48 8029 err <tcp_main.c: 459> ERROR: tcpconn_connect: tcp_blocking_connect failed 08/21 15:07:48 8029 err <tcp_main.c: 721> ERROR: tcp_send: connect failed 08/21 15:07:48 8029 err <../../forward.h: 105> msg_send: ERROR: tcp_send failed 08/21 15:07:48 8029 err <t_fwd.c: 767> ERROR:tm:t_forward_nonack: sending request failed 08/21 15:07:48 8029 debug <tcp_main.c: 718> tcp_send: no open tcp connection found, opening new one And I get a lot of it. Eventually I may get SO_ERROR(111) and then I have to restart proxy. I can post entire config script if anybody had/resolved/willing to take a look at the problem. Thanks a lot, Toly -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log---tcp-blocking-tp1909… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 1

[Kamailio-Users] Res: Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Edson, Please check the tomcat logs. They can be found at /usr/local/tomcat6/logs. When tomcat6 don't start, usually is a typo in the configuration files. Best regards, Flavio E. Goncalves ----- Mensagem original ---- De: Edson - Listas <4lists(a)gmail.com> Para: Flavio Goncalves <flavio(a)asteriskguide.com> Cc: Asim Riaz <ariaz78(a)gmail.com>; users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 14:05:27 Assunto: Re: [Kamailio-Users] Res: Help with SerMyAdmin Yesterday, we faced a similar/related problem, but because Tomcat process was killed. When we try to run it again, SerMyAdmin refuse to start, altought all configurations and DB definitions where there... Any ideas on how to restore it? Edson. Flavio Goncalves escreveu: > Hi Assim, > > I'm glad it is working. SerMyAdmin creates the database fields during > the first run, that's why it worked after restarting the tomcat. The sql > file creates the required users. I'm working to have a debian package > for SerMyAdmin soon. > > Regards, > > Flavio E. Goncalves > > ----- Mensagem original ---- > De: Asim Riaz <ariaz78(a)gmail.com> > Para: users <users(a)lists.kamailio.org> > Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 > Assunto: Re: [Kamailio-Users] Help with SerMyAdmin > > Its working thanks for the help. > I removed my existing serMyAdmin directory from tomcat, restarted the > tomcat and run the script mysql -u root -p openser <openser.sql > > regards > Asim Riaz > > On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com > <mailto:yangsengopenser@gmail.com>> wrote: > > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql > script - admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com <mailto:ariaz78@gmail.com>> > > When I am adding a user i am getting > java.lang.NullPointerException; > > also I cant login using admin username. > > On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques > <yangsengopenser(a)gmail.com <mailto:yangsengopenser@gmail.com>> > wrote: > > Hi again, > > I was looking at the database schema of openser and i > noticed that the subscriber table only has 12 columns till rpid. > Looking at the openser.sql script, it has more than 12 > columns (including the 'version' column...). > What is wrong? Should SerMyAdmin add this columns to the > subscriber table? > Is it lacking me something in the database subscribet table? > Any help with this? > > Thanks in advance, > > Nuno > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com > <mailto:muhammad.akl@gmail.com>> > > > > Hi Nuno > > i faced some problems your already facing and succeeded > to solve them so i'd like to as you > > did you install SERweb tables ? > > if though the problems come from there especially the > phplib_id field a duplication happened between it and > the "id" filed so you have to remove SERweb tables and > all thing will go smoothly without any errors > > p.s i recommend you to remove the whole database e.g > openser and install it again using openser_mysql.sh script > > e.g /path/to/openser_mysql.sh create > > press yes for presence and that stuff but for SERweb no > > try and tell me what you get > > BR > > Muhammad > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Assim, I'm glad it is working. SerMyAdmin creates the database fields during the first run, that's why it worked after restarting the tomcat. The sql file creates the required users. I'm working to have a debian package for SerMyAdmin soon. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Asim Riaz <ariaz78(a)gmail.com> Para: users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 Assunto: Re: [Kamailio-Users] Help with SerMyAdmin Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: How did you add the user? With openserctl or with sermyadmin? Note that you have to use the same login/pass that are on the sql script - admin@setup for login and secret for pass. 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> When I am adding a user i am getting java.lang.NullPointerException; also I cant login using admin username. On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: Hi again, I was looking at the database schema of openser and i noticed that the subscriber table only has 12 columns till rpid. Looking at the openser.sql script, it has more than 12 columns (including the 'version' column...). What is wrong? Should SerMyAdmin add this columns to the subscriber table? Is it lacking me something in the database subscribet table? Any help with this? Thanks in advance, Nuno 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> Hi Nuno i faced some problems your already facing and succeeded to solve them so i'd like to as you did you install SERweb tables ? if though the problems come from there especially the phplib_id field a duplication happened between it and the "id" filed so you have to remove SERweb tables and all thing will go smoothly without any errors p.s i recommend you to remove the whole database e.g openser and install it again using openser_mysql.sh script e.g /path/to/openser_mysql.sh create press yes for presence and that stuff but for SERweb no try and tell me what you get BR Muhammad _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

2 1

Re: [Kamailio-Users] Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql script - > admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> > >> When I am adding a user i am getting java.lang.NullPointerException; >> >> also I cant login using admin username. >> >> On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: >> >>> Hi again, >>> >>> I was looking at the database schema of openser and i noticed that the >>> subscriber table only has 12 columns till rpid. >>> Looking at the openser.sql script, it has more than 12 columns >>> (including the 'version' column...). >>> What is wrong? Should SerMyAdmin add this columns to the subscriber >>> table? >>> Is it lacking me something in the database subscribet table? >>> Any help with this? >>> >>> Thanks in advance, >>> >>> Nuno >>> >>> 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> >>> >>> >>>> >>>> Hi Nuno >>>> >>>> i faced some problems your already facing and succeeded to solve them so >>>> i'd like to as you >>>> >>>> did you install SERweb tables ? >>>> >>>> if though the problems come from there especially the phplib_id field a >>>> duplication happened between it and the "id" filed so you have to remove >>>> SERweb tables and all thing will go smoothly without any errors >>>> >>>> p.s i recommend you to remove the whole database e.g openser and install >>>> it again using openser_mysql.sh script >>>> >>>> e.g /path/to/openser_mysql.sh create >>>> >>>> press yes for presence and that stuff but for SERweb no >>>> >>>> try and tell me what you get >>>> >>>> BR >>>> >>>> Muhammad >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[Kamailio-Users] Help with SerMyAdmin
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I´m trying to install SerMyAdmin in my Openser 1.2.2 machine. I follow the steps that were on www.sermyadmin.org. I accomplished the installation task, but now when i try to do something on the DB (for example: create a new user) it gives me a lot of errors. My experience with Grails is null so i don't know were to start. Below i'll show the error messages. If anyone could give me an hand i'll appreciate. Thanks in advance, Nuno *Message:* could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Caused by:* org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Class:* UserController *At Line:* [147] *Code Snippet:* Stack org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] at UserController$_closure8.doCall(UserController.groovy:147) at UserController$_closure8.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [User] at $Proxy3.saveOrUpdate(Unknown Source) ... 31 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry '4000-192.168.137.1' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:2019) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 32 more

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[Kamailio-Users] Fwd: Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

---------- Forwarded message ---------- From: Asim Riaz <ariaz78(a)gmail.com> Date: Thu, Aug 21, 2008 at 2:49 PM Subject: Re: [Kamailio-Users] Help with SerMyAdmin To: Nuno Marques <yangsengopenser(a)gmail.com> my issue is same as Nano tried this command mysql -u root -p openser <openser.sql but still got ERROR 1054 (42S22) at line 1: Unknown column 'version' in 'field list' Regards Asim On Thu, Aug 21, 2008 at 12:26 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > Hi Muhammad, > > I droped the database and then created a new database as you told me... > Now i have to create all users again and an admin for sermyadmin... > But when i tried to run the mysql commands that are on > www.sermyadmin.org/openser/openser.sql they gave me an error: unknown > column 'version' in 'field list'. > Have any idea of what is going on? > By the way... do you know what happened to www.sermyadmin.org - it's seem > like it has been hacked or something... > > Many tahnks, > > Nuno > > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> > > >> >> Hi Nuno >> >> i faced some problems your already facing and succeeded to solve them so >> i'd like to as you >> >> did you install SERweb tables ? >> >> if though the problems come from there especially the phplib_id field a >> duplication happened between it and the "id" filed so you have to remove >> SERweb tables and all thing will go smoothly without any errors >> >> p.s i recommend you to remove the whole database e.g openser and install >> it again using openser_mysql.sh script >> >> e.g /path/to/openser_mysql.sh create >> >> press yes for presence and that stuff but for SERweb no >> >> try and tell me what you get >> >> BR >> >> Muhammad >> > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >

2 1

[Kamailio-Users] Change project name in Surceforge?
by Iñaki Baz Castillo 21 Aug '08

21 Aug '08

Hi, is not time to change the project name in SourceForge? XD It appears still as "OpenSER SIP Server". -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[Kamailio-Users] SIP Session timeouts
by Jason Penton 21 Aug '08

21 Aug '08

Hi All, I have a question about the best solution to ensure that SIP calls are terminated when the 2 parties drop off the network (wireless). Is there a std way of doing this (maybe using sst module) or should this be implemented using some specialised code? If not supported in a std manner in Kamailio, i thought about creating a child process within the DLG module that continuously checks all dialogs and the connected parties and pings them for connectivity. Should connectivity of any participant drop for more than a specific unit of time then the call is focefully torn down by openser. Any ideas? Cheers Jason

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[Kamailio-Users] RATELIMIT link on modules docs page?
by Alex Balashov 21 Aug '08

21 Aug '08

Why does the RATELIMIT link on http://www.kamailio.org/docs/modules/1.4.x/ go to the documentation for the Perl module? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Permissions module error !
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I tried to install on my Openser 1.2.2 the permissions module so i can use an asterisk gateway. I populated the trusted table with information about the gateway. When i tried to run OpenSer it gave me some errors. Can anyone tell me what could be the problem? Below are the erros and the data on trusted table. Thanks in advance, Nuno 0(3824) init_tcp: using epoll_lt as the io watch method (auto detected) 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing labvoipua:/etc/openser# 0(0) permissions - initializing 0(0) Default allow file (//etc/openser/permissions.allow) not found => empty rule set 0(0) Default deny file (//etc/openser/permissions.deny) not found => empty rule set 0(0) hash_table_insert(): Unknown protocol 0(0) ERROR: permissions: trusted_reload(): Hash table problem 0(0) init_trusted(): Reload of trusted table failed 0(0) Error while initializing allow_trusted function 0(0) init_mod(): Error while initializing module permissions ERROR: error while initializing modules 0(0) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy TRUSTED TABLE +----+----------------+-------+--------------+------+---------+ | id | src_ip | proto | from_pattern | tag | version | +----+----------------+-------+--------------+------+---------+ | 1 | 192.168.137.20 | Any | ^sip:.*$ | NULL | 3 | +----+----------------+-------+--------------+------+---------+

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[Kamailio-Users] Speed dial
by salman farci 21 Aug '08

21 Aug '08

Hi, I have openser 1.2.2 installed with mysql as backend. I was able to run a simple call service after registering users in location table. Now that I'm trying implement a speed_dial, it returns "404 call not found" error at the Xlite panel after dialing a short dial number as per speed_dial table entry. But if I short dial my own short dial number it dials back to me(working) ! What might b the problem ? script file : ........... if (uri=~"sip:[0-9]{2}@.*") sd_lookup("speed_dial"); ............ speed_dial table entry : +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | id | username | domain | sd_username | sd_domain | new_uri | fname | lname | description | +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | 5 | 103 | | 33 | 10.1.2.214 | 103(a)10.1.2.214 | | | 4 | 102 | | 22 | 10.1.2.214 | 102(a)10.1.2.214 | | | 6 | 101 | | 11 | 10.1.2.214 | 101(a)10.1.2.214 | | | 9 | 105 | | 55 | 10.1.2.214 | 105(a)10.1.2.214 | | +----+------------+----------+----------------+--------------+----------------+----------+----------+-----------------+ can anyone clarify ? Thanx regards Farci

1 0

[Kamailio-Users] Second Logo
by Darren Sessions 20 Aug '08

20 Aug '08

Good work. I tinkered a bit and came up with something very similar. - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Aug 20, 2008, at 4:52 PM, Ricky Gutierrez wrote: > logo with basic colors > > Regards > > rickygm > > ----- Original Message ---- > From: Johansson Olle E <oej(a)edvina.net> > To: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Cc: devel(a)lists.kamailio.org > Sent: Wednesday, August 20, 2008 4:55:21 AM > Subject: Re: [Kamailio-Devel] logo > > Ricky, > I really like your logo. But I also see issue with using it in pure > black/white situations - no greyshades. > > Can you try to create such a version of it as well? > > This is often needed when producing t-shirts and other stuff, where an > outline is used. > > Thanks. > /O > > <kamailio-new.png>_______________________________________________ > Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel

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[Kamailio-Users] Commercial support?
by Robert McGilvray 20 Aug '08

20 Aug '08

With the departure of voice system from the project are there any commercial entities associated with the development team that offer support for Kamailio? Regards, Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

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[Kamailio-Users] How to test and trace the RLS messages?
by mahesh.peddi＠info-spectrum.com 20 Aug '08

20 Aug '08

Hi All, I am working on openser 1.3.2 version with RLS as "integrated_xcap_server". Its running without erros. But my query is how to test, And please send me call flow between IMS components and RLS server. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

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Re: [Serusers] Content-Type: application/isup read encoded message
by Jiri Kuthan 19 Aug '08

19 Aug '08

I think the best learning experience is OOB http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… it is not really simple, on the contrary -- it is quite complete, but for that it gives you all kinds of examples that can be probably useful to you. -jiri michael(a)teldelnort.com wrote: > Hello jiri, > > Ok, > > Do yo have an idea of a simple example that will help me familiarize the > usage in ser.cfg? > > -michael > > -----Original Message----- > *From:* Jiri Kuthan [mailto:jiri@iptel.org] > *Sent:* Tuesday, August 19, 2008 03:44 PM > *To:* michael(a)teldelnort.com > *Subject:* Re: [Serusers] Content-Type: application/isup read > encoded message > > I don't think so for this particular one -- just try to use it as > other select operators. -jirimichael(a)teldelnort.com wrote:> Hello > Jiri,> > Thanks for the response.> > Is there any documentation on > how to use select operator with @context?> > > I am trying to filter > a search on the context for "II Digits:"> > -michael> > > > -----Original Message-----> *From:* Jiri Kuthan > [mailto:jiri@iptel.org]> *Sent:* Tuesday, August 19, 2008 02:09 AM> > *To:* michael(a)teldelnort.com> *Cc:* serusers(a)iptel.org> *Subject:* > Re: [Serusers] Content-Type: application/isup read> encoded message> > > I think the select operator, @content, should get you access to > the> body. There is however noparsing of SDP to my knowledge > (unless> someone corrects me, I remember it has been> > frequentlymentioned.)-jirimichael(a)teldelnort.com wrote:> Hello,> >> > How can I evaluate the Content-Type: multipart/mixed;boundary= with> > SER?> > Content-Type: application/isup; base=ansi92; version=ansi> > >> Is there a module that allows SER read the contents of the > encoded> message?> I am try to get the > > "Originating Line > Information> ----> II Digits:"> > from the message.> > -michael> > > >> > ------------------------------------------------------------------------>> > > _______________________________________________> Serusers > mailing> list> Serusers(a)lists.iptel.org>> > http://lists.iptel.org/mailman/listinfo/serusers

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[Kamailio-Users] Ratelimit outgoing SIP INVITE messages after dispatcher
by sergejf 19 Aug '08

19 Aug '08

Hello, We are connected to several VoIP carriers that impose a calls per second (CPS) rate limit on their SIP trunks. For example, on TRUNK1 we can only make call attempts at 10 CPS. Our calls are originally made by a dialer that does not support rate limiting, hence it always blasts calls as fast as it can. Upon receiving these SIP INVITE messages we determine what carrier to use and perform a mapping to a dispatcher group. Call attempts are then load-balanced between all SIP trunks from the chosen carrier. My question is whether it is possible to use the ratelimit module in conjunction with the dispatcher (or carrierroute) to ensure we don't overload the carrier SIP trunks? If this is possible, then I wonder how to configure ratelimit to support a 10 CPS call attempts rate. Thanks in advance for any advice on the topic! Serge -- View this message in context: http://www.nabble.com/Ratelimit-outgoing-SIP-INVITE-messages-after-dispatch… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] Content-Type: application/isup read encoded message
by michael＠teldelnort.com 19 Aug '08

19 Aug '08

Hello, How can I evaluate the Content-Type: multipart/mixed;boundary= with SER? Content-Type: application/isup; base=ansi92; version=ansi Is there a module that allows SER read the contents of the encoded message? I am try to get the "Originating Line Information ---- II Digits:" from the message. -michael

3 2

[Kamailio-Users] About Asterisk2Billing
by Ali Jawad 19 Aug '08

19 Aug '08

Hi All Was anyone able to get the accounting part of asterisk2billing to work with Openser ? This looks like a pretty mature solution to me. Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] Accounting for OPenser
by Ali Jawad 19 Aug '08

19 Aug '08

Hi I am looking for a accounting solution, that I can use with Openser without involving Asterisk. Has anyone tested something similar ? I need the solution to work for my prepaid customers, for example he has got 10$ once the credit is used. Deny any more call.s With Regards Ali Jawad System Administrator

3 2

[Kamailio-Users] SIPp dispatcher load testing failure with too many hops - why?
by sergejf 18 Aug '08

18 Aug '08

Hello, I'm attempting to load test a simple dispatcher script in OpenSER 1.3.x using SIPp (built-in UAC and UAS scenarios). My dispatcher.list only has one address 8.XX.XX.12 (a SIPp instance, UAS). SIPp on 8.XX.XX.10 sends the INVITE messages to 63.XXX.XXX.110. Everything goes well until the UAC sends ACK and BYE in quick sequence. OpenSER tries to t_relay() it to itself, which results in a "too many hops" error: 2008-08-18 11:16:56:708 1219072616.708649: Aborting call on unexpected message for Call-Id '1-3548(a)8.XX.XX.10': while expecting '200' (index 8), received 'SIP/2.0 483 To Many Hops Via: SIP/2.0/UDP 8.XX.XX.10:5061;branch=z9hG4bK-3548-1-7 From: sipp <sip:sipp@8.XX.XX.10:5061>;tag=3548SIPpTag001 To: sut <sip:service@63.XXX.XXX.110:5060>;tag=2148SIPpTag011 Call-ID: 1-3548(a)8.XX.XX.10 CSeq: 2 BYE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 I have tried modifying the SIPp UAC script to re-use the Contact: header from the SIP 200 OK message in the R-URI of the SIP ACK and SIP BYE, to no avail. I have loaded the TM module but I suppose that since the SIP 200 OK is the final message of the transaction, SIP ACK and SIP BYE won't automatically be relayed to 8.XX.XX.12. Is this an OpenSER issue or a SIPp scripting issue? Any advice would be appreciated. Please find below some relevant lines from my openser.cfg: route{ xlog("TRACE:ROUTE: src($si:$sp) dst($Ri:$Rp) msg($mb)\n"); # initial checkings if ( !mf_process_maxfwd_header("10") ) { xlog("SCRIPT:ERROR: $rm (from $si:$sp) too many hops\n"); sl_send_reply("483","To Many Hops"); exit; }; if (method==CANCEL) { if (t_check_trans()) t_relay(); exit; } # routing if (has_totag()) { xlog("SCRIPT0:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); loose_route(); if (method=="INVITE") record_route(); route(1); exit; } The SIP ACK and SIP BYE follow the route defined in the above if {} statement. The SIP INVITE follows the dispatcher route shown below: record_route(); # perform load balancing # set algorithm (4 = round robin) $avp(alg) = 4; # set the group $avp(grp) = 2; if (method=="INVITE") xlog("CALL_START: RURI=[$ru] group=[$avp(grp)] callid=$ci\n"); # do balancing if(!ds_select_dst("$avp(grp)", "$avp(alg)")) { xlog("CALL_DROP:INTERNAL: no destinations for [$ru] group=[$avp(grp)], callid=$ci\n"); sl_send_reply("404", "no dst"); exit; } if(avp_check("$avp(i:273)", "eq/i:1")) { if(!avp_pushto("$ruri","$avp(i:271)")) { xlog("CALL_DROP:INTERNAL: cannot push to ruri [$avp(i:271)], group=[$avp(grp)], callid=$ci\n"); sl_send_reply("500", "cannot get dst"); exit; } } else { # redundancy t_on_failure("1"); } xlog("SCRIPT:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); # do forward route(1); exit; -- View this message in context: http://www.nabble.com/SIPp-dispatcher-load-testing-failure-with-too-many-ho… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] poll: project name
by Daniel-Constantin Mierla 18 Aug '08

18 Aug '08

Hello everybody, some people expressed inconveniences regarding the new name, so we would like to get your opinion about, therefore this is a poll that will get us to a decision. Here is the question: Do you like the name "Kamailio"? 1) Yes, I love it. Please keep it. 2) It is OK. 3) I do not mind the name at all 4) I do not like it 5) I hate the name and we should change it There is no need of explanations, as it is up to personal taste and decision. Just vote by option. Cheers, Daniel PS. for those interested... As short history about renaming story. On the 28th of March the board was informed about issues with SER part of project name. You can easily find who has the trademarks. On the 7th of April the board was informed about an agreement for the deadline as of September 30, 2008 to use old name. From there on, all old releases, documents, etc. can be referred by the old name, but nothing new. Other few discussions about what should/could be kept/changed followed and on 12th of May started the discussion about the new name with two main direction, something similar, to remind about old project or just something completely new, to be used as brand name. Couple of proposals and discussions regarding possible names followed up to 16th of July when was the deadline and the start of vote. Majority was reached on 17th but it was required to have opinions from all members of the board, last came from Bogdan, on the 28th of July. All discussions about migration were on the board mailing list. No change were done to openser.org configuration, just the new domain with kamailio in it was pointed to same server. Every step done was reported to board, following a schedule discussed there. Another discussion was about making the renaming and 1.4.0 release in same day. Being practically impossible to do all the work in one day, it was decided to do first the announcement about renaming. That happened on 28th when all agreed with the new name. As you can see, there was no 5 months of renaming duration. There was no change in openser.org DNS configuration undertaken with/without the knowledge of all board members. The new name domains were reserved when majority was achieved, in any case; dns entries for new name were pointed to old server shortly after booking the domains. -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] SIP and IP multicast
by Steven C. Blair 18 Aug '08

18 Aug '08

Has anyone had any experience using SIP via SER to setup an IP multicast session with a group of phones?

2 1

[Kamailio-Users] Looking for a VOIP Engineer
by Jonathan K. Creasy 18 Aug '08

18 Aug '08

Voxitas Inc. is a national VOIP provider based in St. Louis, MO. and we are currently looking for qualified applicants to fill Tier Three support positions and a VOIP Engineer position. The reason I am posting to this list is because we are looking to fill the VOIP Engineer position with someone who has OpenSER experience and is comfortable configuring and maintaining these systems. For details please contact our VP of Operations Zack Sargent. Zack Sargent VP Operations and Engineering Voxitas, Inc. 210 N. Tucker, St. Louis, MO. 63101 314-266-4000 x108 zsargent(a)voxitas.com Thank you! -Jonathan

1 0

[Kamailio-Users] PortaBilling
by Ali Jawad 18 Aug '08

18 Aug '08

Hi All Has anyone tried portabilling with Openser ? If so can you please provide a sample of your config file ? THanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

1 0

[Serusers] SER - Waiting Transaction
by Arun Kumar 18 Aug '08

18 Aug '08

Hi, I'm running SER 0.96 in my production for a long but there is strange problem , when I do serctl moni and looks for transaction it shows waiting transaction in () almost 400+ can some one tell me why there are so many transaction in waiting while I see processing only 10-20 transaction. Thanks arun

1 0

[Kamailio-Users] pick_branch failed (lowest==-1) for code 302--HELP!!
by Krunal Patel 18 Aug '08

18 Aug '08

Hi, Here is the scenario: -----------------------Invite--------------> <----------100 Trying---------------------- <----------302 Moved temporarily-------- -----------------------ACK--------------> Now I would like to route call to contact returned in 302 responce. But I am getting problem: Aug 17 18:10:41 [16667] DBG:core:parse_headers: flags=80 Aug 17 18:10:41 [16667] DBG:tm:t_should_relay_response: T_code=100, new_code=302 Aug 17 18:10:41 [16667] DBG:tm:t_pick_branch: picked branch -1, code 999 Aug 17 18:10:41 [16667] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 302 Aug 17 18:10:41 [16667] DBG:tm:relay_reply: branch=0, save=0, relay=-1 Please help me to fix this. Thanking you Krunal Patel

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