- sr-users - kamailio.org

[OpenSER-Users] Privacy Question
by Helmut Kuper 08 May '08

08 May '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I found an incomming Privacy-Header Field from my PSTN-Gateway-Provider like this: Privacy: header;user;id Openser 1.4 says: May 8 11:22:25 [29986] ERROR:core:parse_privacy: invalid privacy value RFC 3323 says: Privacy-hdr = "Privacy" HCOLON priv-value *(";" priv-value) priv-value = "header" / "session" / "user" / "none" / "critical" ~ / token The values of priv-value today are restricted to the above options, ~ although further options can be defined as appropriate (see Section ~ 7). Each legitimate priv-value can appear zero or one times in a ~ Privacy header. So that means, the incomming Privacy-header is OK, doesn't it? a Bug? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgiyI0ACgkQ4tZeNddg3dzbfACgjyLMZsVFW30bdTkiEJu+4pH2 XPoAnAhBEQ0IvecHApVqdOP7nSLh563x =4nWD -----END PGP SIGNATURE-----

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[OpenSER-Users] regarding presence module facing problem
by Amit Vijayvargiya 08 May '08

08 May '08

Hi All, i m add presence module in openser 1.3. but it not working properly. openser given this error --> DBG:tm:run_reqin_callbacks: trans=0xb60fa590, callback type 1, id 0 entered May 8 21:09:14 [20993] DBG:core:parse_headers: flags=ffffffffffffffff May 8 21:09:14 [20993] DBG:presence:search_event: start event= [message-summary] May 8 21:09:14 [20993] ERROR:presence:handle_subscribe: Missing or unsupported event header field value May 8 21:09:14 [20993] ERROR:presence:handle_subscribe: event= message-summary May 8 21:09:14 [20993] DBG:core:parse_headers: flags=ffffffffffffffff May 8 21:09:14 [20993] DBG:core:check_via_address: params 172.22.24.192, 172.22.24.192, 0 May 8 21:09:14 [20993] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Thanks & Regards Amit

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[OpenSER-Users] OPenSER DISC Problem
by Dilip 08 May '08

08 May '08

Hello EveryOne, In the openser auth-diameter module,the first phase of digest AUTHETICATION of SIP is completly happening.But In the second phase when OPENSER sends the request to diameter client with authorization header,DISC server cant authorize the USER.Because the response of OPENSER and DISC SERVER are different. As per the code, In the DISC Server code, sipauth module is there which contains db.c file. In this file upto calc_response function call comes ,which calculate the response .This response and OPENSER response are different. Is that any configuration File Problem ? I have attached config file here. What was the reason for it? How can I solve this response problem? Thanks To Evrybody Regards, Dilip # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) #/* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes #*/ #check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=192.168.1.1 #fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib64/openser/modules/mysql.so" #loadmodule "/usr/local/lib64/openser/modules/acc.so" loadmodule "/usr/local/lib64/openser/modules/sl.so" loadmodule "/usr/local/lib64/openser/modules/tm.so" loadmodule "/usr/local/lib64/openser/modules/rr.so" loadmodule "/usr/local/lib64/openser/modules/maxfwd.so" loadmodule "/usr/local/lib64/openser/modules/usrloc.so" loadmodule "/usr/local/lib64/openser/modules/registrar.so" loadmodule "/usr/local/lib64/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib64/openser/modules/auth.so" loadmodule "/usr/local/lib64/openser/modules/auth_db.so" loadmodule "/usr/local/lib64/openser/modules/auth_diameter.so" ##Dilip loadmodule "/usr/local/lib64/openser/modules/domain.so" ##Dilip ############################################################ #modparam("dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list") #modparam("acc", "log_flag", 1) #modparam("acc", "log_missed_flag", 1) #modparam("acc", "service_type", 15) ############################################################# # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc|auth_db","db_url","mysql://openser:openserrw@localhost/openser") #Dilip #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "passwd_h") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ############################################################################################################ # Add By Dilip modparam("auth_diameter", "use_domain", 1) modparam("auth_diameter", "diameter_client_host", "linux173") modparam("auth_diameter", "diameter_client_port", 3000) ############################################################################################################ # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; #Add By Dilip if (method=="INVITE") { if (!diameter_www_authorize("linux173")) { www_challenge("", "0"); exit; }; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("192.168.1.173", "subscriber")) { # www_challenge("192.168.1.173", "0"); # exit; # }; # Add by Dilip if(!diameter_www_authorize("linux173")) #if(!diameter_www_authorize("")) { #/* user is not authorized */ www_challenge("linux173", "1"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

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[OpenSER-Users] openseradmin-03 modification
by Zen Kato 08 May '08

08 May '08

Hi, I installed openseradmin-0.3 on openser-1.2.3-notls. There are some errors,but works fine. Is it possible for the following modifications using Ruby-on-Rail even for a Ruby beginner? user login (only 2 screens) ============ http://localhost:3000/user/login Login Required Login : 0103 Password: 0103 [login>>] --------------------------------- [logout] Name: Domain: Username: 0103 Password: 0103 Email Address: 0103@localhost Caller ID: Voicemail: No Voicemail Password: Created At: Last Updated At: [Edit] ---------------------------------- [logout] Username: 0103 Password: 0103 Domain: First Name: Last Name: Email Address: 0103@localhost Caller ID: Voicemail[ ] Voicemail Password: [Edit] show ---------------------------------- admin login =========== http://localhost:3000/admin/login Login Required Login : openser Password: openserrw [login>>] no modification for admin screens. -- Zen Kato

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[OpenSER-Users] Increase MAX_BRANCHES value?
by Zappasodi Daniele 07 May '08

07 May '08

Hello, I'm interested in this question too. I have tried to modify the tm module and I have found a solution that extends the maximum number of branches and it seems to work, but I'm still testing it. Can anyone, more expert than me on the tm module, give a look to the attached file with my patch? Bye Daniele Hi, thanks for your answer... I increased the SHM_MEM_SIZE to 64 and left KG_MEM_POOL_SIZE 4*1024*1024 . Anyway i found that is not possible to raise MAX_BRANCHES to values higher than 30 due to tm module internal limitation. Bye, Marcello ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

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Re: [OpenSER-Users] srv load-balancing
by Stanisław Pitucha 07 May '08

07 May '08

----- "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> wrote: > OpenSER it is not ignoring the weights - the received records are > ordered by weight. The tricky question is about records with the same > > weight - which to be used? openser is right now using the first > returned > by the DNS server. Ok - maybe I should expand that a bit: It's not disallowed behaviour, but rfc2782 specifies standard way for handling weights and sip protocol doesn't really need other handling afaik: --->8--- from http://www.faqs.org/rfcs/rfc2782.html --- The following algorithm SHOULD be used to order the SRV RRs of the same priority: To select a target to be contacted next, arrange all SRV RRs (that have not been ordered yet) in any order, except that all those with weight 0 are placed at the beginning of the list. Compute the sum of the weights of those RRs, and with each RR associate the running sum in the selected order. Then choose a uniform random number between 0 and the sum computed (inclusive), and select the RR whose running sum value is the first in the selected order which is greater than or equal to the random number selected. --->8--- So even if they have same weight: server1=50 and server2=50, openser should get random number X between 0 and 100 and select server1 for X<=50, or server2 for X>50. Currently openser is not using SRV records correctly imho - it's client responsibility to get all those RRs and select one of them randomly from all servers with selected priority (whether they have same weight or not). Please correct me if I'm wrong. Cheers, Stan

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[OpenSER-Users] Prepaid Service with SER + B2BUA + Radius
by Ilker YILMAZ 07 May '08

07 May '08

Dear Group, I would like to have your comments on setting up successful prepaid service using SER + Sippy B2BUA and a Radius Server. Did anybody accomplished to drop a call when a user's balance is empty with B2BUA? I will really appreciate if you can share your experinces on this issue.. Thank You Ilker

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[OpenSER-Users] 200 OK retransmissions on missing ACK can cause subsequent calls to fail
by Sean O'Donnell 07 May '08

07 May '08

Hi all, Im using openser as a call distributor/proxy between a soft-switch/SBC and voicemail platform. Im seeing a problem with openser in that it is sometimes cancels an in-progress call (fr_inv_timer firing) because it didnt match the 200/OK with the call. After some investigation, I noticed that this was happening after a missing ACK on a previous call caused the voicemail platform to retransmit 200/OK responses beyond the TM wt_timer expiration, which in turn left several openser child processes (those that received a 200 after wt_timer expiration) in a state such that they might not properly match transactions on subsequent calls. My setup: I have openser 1.2.0 operating on a linux box with two network interfaces, with one interface (call it the outside interface) taking incoming calls from the soft-switch, and the other (inside) connected to the VM platform. I have openser configured to use both interfaces (see config below) and the TM wt_timer set to 5 seconds (default). As this is a voicemail system, all of the call traffic is inbound from the soft-switch. Given the traffic flow, for the most part the openser child processes servicing the inside interface are handling responses (180,183,200) from the VM platform. Call scenario: When an INVITE arrives from the soft-switch, openser forwards it to the VM platform. The VM platform responds with a 180 and then a 200. I've noticed several instances where the soft-switch did not respond with an ACK. This caused the VM platform to retransmit the 200 several times over a 10 second period. These were absorbed correctly by openser for the duration of wt_timer. After the timer expired, however, each openser child process that received a retransmitted 200 logged something like this: 4(2715) DEBUG: t_reply_matching: hash 45870 label 727647196 branch 0 4(2715) DEBUG: t_reply_matching: no matching transaction exists 4(2715) DEBUG: t_reply_matching: failure to match a transaction 4(2715) DEBUG: t_check: end=(nil) When I look at the TM code, the static variable T in t_lookup.c is now NULL for this child process. On a subsequent inbound call, the INVITE is passed to the VM correctly, and the 180 transaction matches (causing the fr_inv_timer to be armed). If the 200 is read by child proc 2715, I see: 4(2715) DEBUG: t_check: start=(nil) 4(2715) DEBUG: t_check: T previously sought and not found The 200 is forwarded back to the soft-switch, which responds with an ACK. Both end-points think the call is up, but since openser never matched the 200 with the call, the fr_inv_timer fires and cancels the call. Basically, child proc 2715 wont match any transaction after this unless it happens to process a request. I think this problem is made worse by the fact that Im using two network interfaces, and that the openser children on the inside interface handle (for the most part) only responses. This problem was touched on here: http://lists.openser.org/pipermail/users/2007-November/014188.html but I didnt see any follow up. Also, Ive checked openser 1.2.3 and 1.3.1 for fixes, but I dont think this has been addressed. I have a work around, I think, by upping the wt_timer to something like 15 seconds, but I was wondering if there is any scenario in which leaving T=NULL is desirable. Thanks in advance Sean ==== CONFIG ===== # # VMPROXY-PROXY - route between softswitch and media servers # # ----------- global configuration parameters ------------------------ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=8 # multi-homed (i.e. listen on two or more interfaces) mhomed=1 alias=sipproxy1 alias=sipproxy1.vmproxy.com port=5060 #inside address listen=192.168.1.100 #outside address listen=10.55.2.100 disable_dns_blacklist=yes # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/openser/lib/openser/modules/mysql.so" mpath="/usr/local/openser/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "exec.so" loadmodule "xlog.so" loadmodule "textops.so" loadmodule "options.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/openser/lib/openser/modules/auth.so" #loadmodule "/usr/local/openser/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # local memory #modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # uncomment this for multiple record routes #modparam("rr", "enable_double_rr", 1) # -- tm params -- modparam("tm", "fr_timer", 5) modparam("tm", "fr_inv_timer", 60) # -- mi_fifo parms modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo_proxy") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); return; }; xlog("L_INFO","$Tf VMPROXY: <$rm> message From:<$fu> To:<$tu> srcip:<$si>\n"); # # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # record_route(); # # loose-route processing # messages that already have routes on them are proxied here # if (loose_route()) { t_relay(); return; }; # # message not for us, route it through # if ( !(uri==myself) ) { if (!t_relay()) { sl_reply_error(); }; return; }; xlog("L_DBG","$Tf VMPROXY: <$rm> for me\n"); # handle options meant for me if( method == "OPTIONS" ) { options_reply(); return; }; if( method == "REGISTER" || method == "PUBLISH" || method == "SUBSCRIBE" || method == "MESSAGE" ) { sl_send_reply("405","Method Not Allowed"); return; }; # route INVITE based on source if( method == "INVITE" ) { exec_dset("/usr/local/vmproxy/bin/routeINVITE.sh"); xlog("L_DBG","$Tf VMPROXY: <$rm> after routeINVITE: new destination <$ru>\n"); t_on_failure("1"); }; # route NOTIFY to softswitch if( method == "NOTIFY" ) { exec_dset("/usr/local/vmproxy/bin/routeNOTIFY.sh"); xlog("L_DBG","$Tf VMPROXY: <$rm> after routeNOTIFY: new destination <$ru>\n"); t_on_failure("2"); }; # # NOTE: this logic is only executed if the request didn't have # a route on it (tsk,tsk). If it did, it would have been handed via # loose route function above. if( method == "ACK" || method == "BYE" || method == "CANCEL" || method == "PRACK" ) { exec_dset("/usr/local/vmproxy/bin/routeACK.sh"); xlog("L_DBG","$Tf VMPROXY: <$rm> after routeACK: new destination <$ru>\n"); }; # routing script error? if( uri=="sip:err@127.0.0.1" ) { sl_reply_error(); return; }; # forward to current uri if (!t_relay()) { sl_reply_error(); }; } # # Initial INVITE route failed, retry # failure_route[1] { # endpoint canceled call if( t_was_cancelled()) { xlog("L_DBG", "$Tf VMPROXY: call <id:$ci> cancelled\n"); return; } # don't retry on 40[0-7|5xx failures if( !t_check_status("40[0-7]|420|487|488|5[0-9][0-9]")) { xlog("L_ERR","$Tf VMPROXY: ERROR routing call <$rm> from <$fu> to <$ru>\n"); exec_dset("/usr/local/vmproxy/bin/routeINVITE2.sh"); xlog("L_ERR","$Tf VMPROXY: Rerouting call <$rm> to <$ru>\n"); append_branch(); t_relay(); } } # # Initial NOTIFY route failed, retry # failure_route[2] { # endpoint canceled call if( t_was_cancelled()) { xlog("L_DBG", "$Tf VMPROXY: call <id:$ci> cancelled\n"); return; } # don't retry on 40[0-7|5xx failures if( !t_check_status("40[0-7]|5[0-9][0-9]")) { xlog("L_ERR","$Tf VMPROXY: ERROR routing <$rm> from <$fu> to <$ru>\n"); exec_dset("/usr/local/vmproxy/bin/routeNOTIFY2.sh"); xlog("L_ERR","$Tf VMPROXY: Rerouting <$rm> to <$ru>\n"); append_branch(); t_relay(); } }

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[Serusers] SER Nokia CANCEL Problem
by Abdul Qadir 07 May '08

07 May '08

Hi , I tried to call from one nokia sip (E61 and other models )phone to another nokia sip phone. The call works fine. The problem comes only when I call from Phone A to Phone B and then immediately cancel the call(from Phone A). The Phone A will hangup the call as it sent CANCEL but the SER will ignore this CANCEL and still send INVITE to Phone B resulting in a ghost call situation. I tried to capture a log of message and found that Phone A "CANCEL" message is received on SER even before any provisional response from Phone B. Therefor SER doesnot relay this CANCEL request to Phone B. I even checked RFC which clearly says that UAC should not send CANCEL untill it receives any provisional response. I talked to Nokia expert and they said the 100 Trying message from your server is considered as provisional response, therefor behaviour of client is absolutely correct. Is there any way I can stop 100 Trying message and still run statefull SER, so that I can verify what nokia said. Any ideas suggestions are welcome. Thanking you all in advance. Best Regards, Abdul Qadir --------------------------------- Don't be flakey. Get Yahoo! Mail for Mobile and always stay connected to friends.

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[OpenSER-Users] Pseudovariable for whole SIP-message
by Peter Lemenkov 07 May '08

07 May '08

Hello All! Is there a pseudovariable denoting the whole SIP-message? If not - are there plans for implementing such functionality? -- With best regards!

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Re: [OpenSER-Users] Version table
by Henning Westerholt 07 May '08

07 May '08

On Wednesday 07 May 2008, Bård Aase wrote: >> [...] > Thanks, I will check this out now, I'm using OpenSER 1.3.1 and the > sip_buddies table looks like this: > [..] You'll find the subscriber DB docs (for trunk) at: http://www.openser.org/docs/db-tables/openser-db-devel.html#GEN-DB-SUBSCRIB… AFAIK there were some fiels removed, but except this it should be the same format as in the 1.3 branch. > The openser.cfg configuration for auth_db: > [..] > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "user_column", "name") > modparam("auth_db", "domain_column", "domain") > modparam("auth_db", "password_column", "secret") > modparam("auth_db", "password_column_2", "ha1b") > modparam("auth_db", "use_domain", 0) > modparam("auth_db", "db_url", > "mysql://UNDISCLOSED:UNDISCLOSED@UNDISCLOSED/asterisk") > > All the fields referred to here exists in the asterisk.sip_buddies > table, so my guess now is that this is just a matter of disabling the > version check, or bumping the version number. I think the same. If the semantics of the colums in question are also the same, then it should work just fine. Regards, Henning

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[OpenSER-Users] Fwd: Version table
by Bård Aase 07 May '08

07 May '08

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Re: [OpenSER-Users] [OpenSER-Devel] Memory dump for OpenSER 1.3.1 which slowly leaks memory
by Henning Westerholt 07 May '08

07 May '08

On Tuesday 06 May 2008, Julian Yap wrote: > Private memory is default. Should I increase this? Is there a > recommended rule of thumb? > > Shared memory is set for 128Mb. There were memory errors at 32Mb but > none since this was increased to 128Mb. > > Using the setting 'children = 8'. (moved to user list) Hi Julien, you only need to increase this if you use some modules that needs to load a big dataset on startup (e.g. carrierroute), and if you get allocation errors for the private memory. If you run with 128 MB shared memory, then the server should not use more then this memory, i.e. the depletion of memory you observed is not caused from OpenSER. The ammount of memory that is used from the server is: ((actual number of children) x private memory) + shared memory Perhaps you can adapt your "monitoring" to log the actual biggest memory consumer on the system? Cheers, Henning

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[OpenSER-Users] Version table
by Bård Aase 07 May '08

07 May '08

I am trying to switch from SER to OpenSER, and I'm using mysql for data storage. Asterisk's sip_buddies table is supposed to contain authentication data. The configuration is set up, the openser database is appearantly set up and working, but when I try to authenticate against asterisk.sip_buddies i get the following errors: May 6 12:36:58 [25672] ERROR:auth_db:auth_fixup: invalid table version (use openser_mysql.sh reinstall) May 6 12:36:58 [25672] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 399 May 6 12:36:58 [25672] CRITICAL:core:fix_expr: fix_actions error May 6 12:36:58 [25672] ERROR:core:main: failed to fix configuration with err code -1 The version tables in asterisk and openser is as follows: SELECT * FROM asterisk.`version` 5 SELECT * FROM openser.`version` 6 What is the differences between database version 5 and 6? Since the asterisk database is shared between multiple asterisks and SER instances, its not trivial to upgrade the tables if that would have implications on asterisk and/or ser Thanks in advance. -- Bård Aase gTalk/Jabber: elzapp(a)elzapp.com MSN: msn(a)elzapp.com http://blog.elzapp.com

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[OpenSER-Users] Log errors when processing a contact list with RLS module
by Pablo Guijarro Enríquez 07 May '08

07 May '08

Hi, I am running openser v1.3.1 and I cannot get the RLS module to work properly when handling a contact list already retrieved from database. Some errors related with the pua module appear in the log when the server seems to be trying to generate the subscribe requests for each of the contacts in the list (see attached, lines 60 to 62). Apr 10 13:02:23 comm ./openser[20184]: ERROR:pua:subs_build_hdr: getting event from list Apr 10 13:02:23 comm ./openser[20184]: ERROR:pua:send_subscribe: while building extra headers Apr 10 13:02:23 comm ./openser[20184]: ERROR:rls:process_list_and_exec: infunction given as a parameter Those requests are never sent. What could this be due to? Thanks! Pablo

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[OpenSER-Users] Problems with SIMPLE-XMPP presence
by Pablo Guijarro Enríquez 06 May '08

06 May '08

Hi everybody, I have some problems to get presence information exchanged between SIP users and xmpp ones. SIP clients (X-Lite) depend on an openser server v1.3.1, with all necessary modules working within it, and xmpp clients (Psi) rely on an xmpp server (ejabberd) which is in the same machine. The link between both sip and xmpp servers is established when openser starts, and the exchange of instant messages between sip and xmpp users works fine. So does presence too, as long as there are only sip users or only xmpp users involved, but it does not work between the two worlds in any direction. Moreover, I do not see any packet being exchanged between the sip and the xmpp servers when a user from one domain subscribe to one from the other, or when they change their status. I dont know what the problem can be. No errors appear in the log and I thought adding xmpp presence to openser would be straightforward once the IM was already working. Openser config file and part of the log file (the subscription to an xmpp user) are attached. Hope someone can give me some clue. Thanks in advance! Paul

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[Serusers] Issue for subst()
by inge 06 May '08

06 May '08

Hi all, I find a solution to replace in my From the string "anonymous" by "92". So, I do the following : subst('/^(From|f): (.*) <sip:(.*)@(.*)>;/\1: \"Anonymous\" <sip:92@ \4>;/g'); The result is : From: "Anonymous" <sip:anonymous@From: "Anonymous" <sip:92@0172292965@domain.tld>;tag=21910a7967574581o0. I try a lot of pattern, but it doesn't run. Thanks for your support. Regards, Adrien

1 0

[Serusers] SER MySQL Memory Problem
by Arun Kumar 06 May '08

06 May '08

Hi All, I'm facing a strange issue: I've installed SER and MySQL on two machines, SER(200K users in subscriber table) is running with 6GB and MySQL with 5GB of memory but in just 2 days MySQL is using almost 5GB of memory and my SER response is getting very slow some time number of transaction waiting will be 1300+ :( any solution to this will be really helpful. thanks in advance. arun

1 0

[OpenSER-Users] null socket list
by Michael Barinek 05 May '08

05 May '08

I have openser-1.3.1-notls installed although am unable to start openser. Any suggestions/ideas regarding the below error? Any existing docs/links would be great. Thanks /// Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.86.56 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.87.100 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.37.129.2 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.211.55.2 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.86.56 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.87.100 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.37.129.2 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.211.55.2 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.138.86.56 [10.138.86.56]:5060 udp: 10.138.87.100 [10.138.87.100]:5060 udp: 10.37.129.2 [10.37.129.2]:5060 udp: 10.211.55.2 [10.211.55.2]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.138.86.56 [10.138.86.56]:5060 tcp: 10.138.87.100 [10.138.87.100]:5060 tcp: 10.37.129.2 [10.37.129.2]:5060 tcp: 10.211.55.2 [10.211.55.2]:5060 Aliases: tcp: my.machine.name:5060 udp: my.machine.name:5060 Mar 20 08:59:14 [18124] WARNING:core:main: no fork mode and more than one listen address found(will use only the first one) Mar 20 08:59:14 [18124] INFO:core:init_tcp: using kqueue as the TCP io watch method (auto detected) Mar 20 08:59:14 [18124] NOTICE:core:main: version: openser 1.3.1-notls (i386/darwin) Mar 20 08:59:14 [18124] INFO:core:main: using 32 Mb shared memory Mar 20 08:59:14 [18124] INFO:core:main: using 1 Mb private memory per process Mar 20 08:59:14 [18124] INFO:sl:mod_init: Initializing StateLess engine Mar 20 08:59:14 [18124] INFO:tm:mod_init: TM - initializing... Mar 20 08:59:14 [18124] CRITICAL:tm:uac_init: null socket list Mar 20 08:59:14 [18124] ERROR:tm:mod_init: uac_init failed Mar 20 08:59:14 [18124] ERROR:core:init_mod: failed to initialize module tm Mar 20 08:59:14 [18124] ERROR:core:main: error while initializing modules

3 4

[Serusers] Can't get the value of 'Digest-User-name', 'Digest-Realm', 'Digest-Method', 'Digest-Uri', 'Digest-Nonce', 'Digest-Response'
by johnson elangbam 05 May '08

05 May '08

hi, I am using SER 0.9.6 along with free radius 2.0.3 and radius client 0.5.6, I need to authenticate a user by using the digest attributes, the problem is that the radius client did not any send the following attributes to the radius server: Digest-User-name, Digest-Realm, Digest-Method, Digest-Uri, Digest-Nonce and Digest-Response. Please tell me how do I send these attributes so that i can get the values to the radius server, here is the output of the SER when running in debug mode: 0(5069) WARNING: fix_socket_list: could not rev. resolve 192.168.1.227 0(5069) WARNING: fix_socket_list: could not rev. resolve 192.168.1.227 Listening on udp: 192.168.1.227 [192.168.1.227]:5060 tcp: 192.168.1.227 [192.168.1.227]:5060 Aliases: 0(5069) fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(5069) fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16548 0(5069) fm_malloc_init(0xb5f3e000, 33554432), start=0xb5f3e000 0(5069) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: sl_module stateless - initializing 0(0) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(0) unixsock_register_cmd: New command (sl_stats) registered 0(0) DEBUG: MD5 calculated: 79977158fb94a9d13bb3c426277eb749 0(0) DEBUG: init_mod: tm [root@localhost ser]# 0(0) TM - (size of cell=2776, sip_msg=760) initializing... 0(0) Call-ID initialization: '6f7e7ee5' 0(0) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(0) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(0) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(0) DEBUG: register_fifo_cmd: new command (t_reply) registered 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(0) unixsock_register_cmd: New command (t_stats) registered 0(0) DEBUG: MD5 calculated: 5fee62c0d1ead6c68c47a98572cd20ab 0(0) DEBUG: MD5 calculated: 716032d4c0061ac55304d86f005be6a0 0(0) DEBUG: init_mod: rr 0(0) rr - initializing 0(0) DEBUG: init_mod: maxfwd 0(0) Maxfwd module- initializing 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: found <db_use_table> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_init> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_close> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_raw_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_free_result> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) DEBUG: init_mod: registrar 0(0) registrar - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(0) find_export: found <ul_bind_usrloc> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_register_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_get_all_ucontacts> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_insert_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_delete_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_get_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_lock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_unlock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_release_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_insert_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_delete_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_get_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_update_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_register_watcher> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_unregister_watcher> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_register_ulcb> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(0) DEBUG: init_mod: textops textops - initializing 0(0) DEBUG: init_mod: xlog 0(0) XLOG: initializing ... 0(0) DEBUG: init_mod: auth 0(0) auth module - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(0) DEBUG: init_mod: auth_db 0(0) auth_db module - initializing 0(0) find_mod_export: found <db_use_table> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_init> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_close> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_raw_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_free_result> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_export: found <pre_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(0) find_export: found <post_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(0) DEBUG: init_mod: auth_radius 0(0) auth_radius - Initializing 0(0) auth_radius: No `Cisco' vendor in Radius dictionary 0(0) find_export: found <pre_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(0) find_export: found <post_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute 0(0) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5073) shm_mem_destroy 0(5073) destroying the shared memory lock Thanks and Regards, Elangbam Johnson

1 0

[OpenSER-Users] [OT] La biblia del SIP y NAT
by Iñaki Baz Castillo 05 May '08

05 May '08

Hola, sólo era para comentar la existencia de un draft que desconocía y que reúne toda la información posible sobre SIP y NAT: http://tools.ietf.org/html/draft-ietf-sipping-nat-scenarios Ala, me lo guardo y cuando alguien pregunte algo sobre "oigo pero no me oyen" se lo enchufo XD -- Iñaki Baz Castillo

2 1

[OpenSER-Users] srv load-balancing
by stanis＠zimbra-1.gradwell.net 04 May '08

04 May '08

Hi, I've got a problem with openser servers and load balancing. Current setup contains 2 asterisks and 2 opensers. I've got a srv record xxx for both asterisk-1 and asterisk-2 - same priorities and weights. Openser's config runs basically: rewritehostport("xxx"); t_relay(); But all packets go to first host on the list anyways. For example now I get this from dns (same order): _sip._udp.xxx. 300 IN SRV 1 50 5060 asterisk-2 _sip._udp.xxx. 300 IN SRV 1 50 5060 asterisk-1 and got 6 calls on asterisk-2 and none on asterisk-1. If I change the order, all calls go to the first host on the list. Shouldn't openser do load-balancing in this scenario? I don't use any options in t_relay(). Thanks for ideas.

4 3

[OpenSER-Users] Mediaproxy problem - Nated->PSTN calls
by Krunal Patel 03 May '08

03 May '08

Hi, I am facing mediaproxy problem during Nated-pstn calls. Here is the problems: 1. Mediaproxy session has been broken 2. No called address on mediaproxy server 3. Mediaproxy session status is inactive 4. No bytes has been passed through mediaproxy So please help me to solve all this problems. Thanking you in advance.. - Krunal Patel

2 1

[Serusers] [Fwd: Please Help : 2]
by Akintayo Olusegun 03 May '08

03 May '08

I figured out the cause of the errror..... Its funny.....on linux you execute a command and parameters like this sayIt helloworld right.. but on suse you execute a command like this ./sayit helloworld The only exception is if you specify a full path to sayIt like this /usr/bin/sayIt helloworld so in the ser_mysql.sh file gen_ha1 should be changed to ./gen_sha1 and bingo.....no errors.... This same thing goes for all other commands that generate command not found error...it actually happened with serctl too and i fixed it Thanks for listening Akintayo Olusegun Access Solutions

1 0

[Serusers] Please Help : 2
by Akintayo Olusegun 03 May '08

03 May '08

1 0

[Serusers] Please Help
by Akintayo Olusegun 03 May '08

03 May '08

1 0

[OpenSER-Users] running openser on port 5070
by toly 03 May '08

03 May '08

Greetings and regards to Everyone. I'm running Openser/Mediaproxy with NAT traversal. Opeser initially was running on port 5060. Our client is behind Cisco firewall and at some point they turned on Sip fixup (which is beyond our control ) because they need sip fixup for some other application and as a result nat travesal can't work. Per my conversation with their IT guy they have no problem with our sip proxy running on port 5070. I've reconfigured proxy on port 5070 - everything is working with no problems. Here are my questions: 1. Anything wrong running sip proxy on port 5070 or some other port (if it does not interfere with other services) in general. 2. Would such solution considered as permanent solution? TIA Toly -- View this message in context: http://www.nabble.com/running-openser-on-port-5070-tp17036191p17036191.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[OpenSER-Users] LCR Module / Function: load_gws() error
by Darren Sessions 03 May '08

03 May '08

Question . . I'm trying to use the load_gws with a particular group id per the docs for 1.3, and am getting the following error message when I try and start OpenSER: May 1 20:40:05 [16853] DBG:core:fix_actions: fixing load_gws, line 487 May 1 20:40:05 [16853] ERROR:lcr:fixstringloadgws: Wrong value <2> for param <1>! May 1 20:40:05 [16853] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 487 May 1 20:40:05 [16853] CRITICAL:core:fix_expr: fix_actions error May 1 20:40:05 [16853] ERROR:core:main: failed to fix configuration with err code -1 May 1 20:40:05 [16853] INFO:snmpstats:mod_destroy: The SNMPStats module got the kill signal May 1 20:40:05 [16853] INFO:snmpstats:mod_destroy: Shutting down the AgentX Sub-Agent! May 1 20:40:05 [16853] DBG:xlog:destroy: destroy module... May 1 20:40:05 [16853] DBG:tm:tm_shutdown: tm_shutdown : start May 1 20:40:05 [16853] DBG:tm:unlink_timer_lists: emptying DELETE list May 1 20:40:05 [16853] DBG:tm:tm_shutdown: emptying hash table May 1 20:40:05 [16853] DBG:tm:tm_shutdown: releasing timers May 1 20:40:05 [16853] DBG:tm:tm_shutdown: removing semaphores May 1 20:40:05 [16853] DBG:tm:tm_shutdown: destroying callback lists May 1 20:40:05 [16853] DBG:tm:tm_shutdown: tm_shutdown : done May 1 20:40:05 [16853] DBG:core:shm_mem_destroy: May 1 20:40:05 [16853] DBG:core:shm_mem_destroy: destroying the shared memory lock Am I missing something? Per the docs at http://www.openser.org/docs/modules/1.3.x/lcr.html#AEN303 , I should be able to use a particular gateway group id in that function. My config code excerpt: if (from_gw("1000")) { xlog("L_ERR", "Route 14: Call originating from incoming vendor\n"); if (!load_gws("2")) { xlog("L_ERR", "Route 14: Error loading gateways (GW Group 2) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "Termination Currently Unavailable"); exit; } if (!next_gw()) { xlog("L_ERR", "Route 14: No gateways available (GW Group 2) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(10); } Basically, it's checking to see if the call is originating from one of the incoming gateways in group id 1000, and if it is, I'd like to pull a gateway from group 2 and send the call to it. Any help would be appreciated. Thanks, - Darren

2 2

[OpenSER-Users] NATED-PSTN mediaproxy problem
by Krunal Patel 02 May '08

02 May '08

Hi One more thing, During Nated to PSTN calls , on reply , reply route is not executed. Here is the trace from ngrep ####################################################################################### # U xxx.xxx.16.194:5061 -> xxx.xxx.190.51:5060 INVITE sip:800900800@XXXXXX.XXXXX.XXX SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport;branch=z9hG4bK1038F227 01BD0EF5F76A88C87628C608..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:8009 00800(a)XXXXXX.XXXXX.XXX>..Contact: <sip:08281962103@xxx.xxx.16.194:5061>..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F 113099@192.168.1.10..CSeq: 20325 INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 11 05d..Content-Length: 293....v=0..o=08281962103 2844250539 2844250642 IN IP4 xxx.xxx.16.194..s=X-Lite..c=IN IP4 122.169. 16.194..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rt pmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U xxx.xxx.190.51:5060 -> xxx.xxx.16.194:5061 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport=5061;branch=z9hG4bK1038F22701BD0E F5F76A88C87628C608..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@ XXXXXX.XXXXX.XXX>;tag=b802e73ab92fec121a4f880263d541a3.ef67..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099(a)192.1 68.1.10..CSeq: 20325 INVITE..Proxy-Authenticate: Digest realm="XXXXXX.XXXXX.XXX", nonce="481b0547bb2c135b83a7938 0026cdaf6a2007b4a"..Server: OpenSER (1.2.1-notls (i386/linux))..Content-Length: 0.... # U xxx.xxx.16.194:5061 -> xxx.xxx.190.51:5060 ACK sip:800900800@XXXXXX.XXXXX.XXX SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport;branch=z9hG4bK1038F22701B D0EF5F76A88C87628C608..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:8009008 00(a)XXXXXX.XXXXX.XXX>;tag=b802e73ab92fec121a4f880263d541a3.ef67..Contact: <sip:08281962103@xxx.xxx.16.194:5061>.. Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20325 ACK..Max-Forwards: 70..Content-Length: 0.... # U xxx.xxx.16.194:5061 -> xxx.xxx.190.51:5060 INVITE sip:800900800@XXXXXX.XXXXX.XXX SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport;branch=z9hG4bK531E08BE 030E99FE369F25E3CAEC7443..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:8009 00800(a)XXXXXX.XXXXX.XXX>..Contact: <sip:08281962103@xxx.xxx.16.194:5061>..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F 113099@192.168.1.10..CSeq: 20326 INVITE..Proxy-Authorization: Digest username="08281962103",realm="voiptest.convergenze .it",nonce="481b0547bb2c135b83a79380026cdaf6a2007b4a",response="a66a7fff4dbfba9dc37a5e4c3b44652a",uri="sip:800900800@vo iptest.convergenze.it"..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1105d..Content-Leng th: 293....v=0..o=08281962103 2844250539 2844250642 IN IP4 xxx.xxx.16.194..s=X-Lite..c=IN IP4 xxx.xxx.16.194..t=0 0..m= audio 8000 RTP/AVP 0 8 3 98 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000 ..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U xxx.xxx.190.51:5060 -> xxx.xxx.179.28:5060 INVITE sip:800900800@xxx.xxx.179.28:5060;transport=udp SIP/2.0..Record-Route: <sip:XXXXXX.XXXXX.XXX:5060;nat=yes ;ftag=660878483;lr=on>..Via: SIP/2.0/UDP xxx.xxx.190.51;branch=z9hG4bKa5cc.576d5334.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5 061;rport=5061;branch=z9hG4bK531E08BE030E99FE369F25E3CAEC7443..From: 08281962103 <sip:08281962103@voiptest.convergenze. it:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>..Contact: <sip:08281962103@xxx.xxx.16.194:5061>..Ca ll-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..Max-Forwards: 69..Content-Type: applicati on/sdp..User-Agent: X-Lite release 1105d..Content-Length: 314....v=0..o=08281962103 2844250539 2844250642 IN IP4 122.16 9.16.194..s=X-Lite..c=IN IP4 213.136.179.21..t=0 0..m=audio 42214 RTP/AVP 0 8 3 98 101..a=rtpmap:0 pcmu/8000..a=rtpmap: 8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv ..a=direction:active.. # U xxx.xxx.179.28:5060 -> xxx.xxx.190.51:5060 SIP/2.0 100 Trying..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..From: 08281962103 < sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>;tag=cfcb0b247c 4126c..Via: SIP/2.0/UDP xxx.xxx.190.51;branch=z9hG4bKa5cc.576d5334.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;branch=z9hG4b K531E08BE030E99FE369F25E3CAEC7443;rport=5061..Content-Length: 0..User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05.... # U xxx.xxx.179.28:5060 -> xxx.xxx.190.51:5060 SIP/2.0 183 Session Progress..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..From: 082 81962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>;tag= cfcb0b247c4126c..Via: SIP/2.0/UDP xxx.xxx.190.51;branch=z9hG4bKa5cc.576d5334.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;bra nch=z9hG4bK531E08BE030E99FE369F25E3CAEC7443;rport=5061..Record-Route: <sip:XXXXXX.XXXXX.XXX:5060;nat=yes;ftag=66 0878483;lr=on>..Content-Length: 211..Content-Type: application/sdp..Contact: sip:800900800@xxx.xxx.179.28:5060..User-Ag ent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05....v=0..o=MxSIP 0 370160 IN IP4 213.136.179 .28..s=SIP Call..c=IN IP4 xxx.xxx.179.28..t=0 0..m=audio 15536 RTP/AVP 0 8 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8 000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U xxx.xxx.190.51:5060 -> xxx.xxx.16.194:5061 SIP/2.0 183 Session Progress..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..From: 082 81962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>;tag= cfcb0b247c4126c..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;branch=z9hG4bK531E08BE030E99FE369F25E3CAEC7443;rport=5061..Record -Route: <sip:XXXXXX.XXXXX.XXX:5060;nat=yes;ftag=660878483;lr=on>..Content-Length: 211..Content-Type: application /sdp..Contact: sip:800900800@xxx.xxx.179.28:5060..User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05....v=0..o=MxSIP 0 370160 IN IP4 xxx.xxx.179.28..s=SIP Call..c=IN IP4 xxx.xxx.179.28..t=0 0..m=audio 15536 RTP/AVP 0 8 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U xxx.xxx.179.28:5060 -> xxx.xxx.190.51:5060 SIP/2.0 200 OK..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..From: 08281962103 <sip: 08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>;tag=cfcb0b247c4126 c..Via: SIP/2.0/UDP xxx.xxx.190.51;branch=z9hG4bKa5cc.576d5334.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;branch=z9hG4bK531 E08BE030E99FE369F25E3CAEC7443;rport=5061..Record-Route: <sip:XXXXXX.XXXXX.XXX:5060;nat=yes;ftag=660878483;lr=on> ..Content-Length: 211..Content-Type: application/sdp..Supported: replaces..Contact: sip:800900800@xxx.xxx.179.28:5060.. User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05....v=0..o=MxSIP 0 370160 IN IP4 213. 136.179.28..s=SIP Call..c=IN IP4 xxx.xxx.179.28..t=0 0..m=audio 15536 RTP/AVP 0 8 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U xxx.xxx.190.51:5060 -> xxx.xxx.16.194:5061 SIP/2.0 200 OK..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..From: 08281962103 <sip: 08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>;tag=cfcb0b247c4126 c..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;branch=z9hG4bK531E08BE030E99FE369F25E3CAEC7443;rport=5061..Record-Route: <sip:v oiptest.convergenze.it:5060;nat=yes;ftag=660878483;lr=on>..Content-Length: 211..Content-Type: application/sdp..Supporte d: replaces..Contact: sip:800900800@xxx.xxx.179.28:5060..User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3 .2.8.45 00a0ba009b05....v=0..o=MxSIP 0 370160 IN IP4 xxx.xxx.179.28..s=SIP Call..c=IN IP4 xxx.xxx.179.28..t=0 0..m=audi o 15536 RTP/AVP 0 8 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U xxx.xxx.179.28:5060 -> xxx.xxx.190.51:5060 SIP/2.0 200 OK..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..From: 08281962103 <sip: 08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>;tag=cfcb0b247c4126 c..Via: SIP/2.0/UDP xxx.xxx.190.51;branch=z9hG4bKa5cc.576d5334.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;branch=z9hG4bK531 E08BE030E99FE369F25E3CAEC7443;rport=5061..Record-Route: <sip:XXXXXX.XXXXX.XXX:5060;nat=yes;ftag=660878483;lr=on> ..Content-Length: 211..Content-Type: application/sdp..Supported: replaces..Contact: sip:800900800@xxx.xxx.179.28:5060.. User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05....v=0..o=MxSIP 0 370160 IN IP4 213. 136.179.28..s=SIP Call..c=IN IP4 xxx.xxx.179.28..t=0 0..m=audio 15536 RTP/AVP 0 8 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U xxx.xxx.190.51:5060 -> xxx.xxx.16.194:5061 SIP/2.0 200 OK..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 INVITE..From: 08281962103 <sip: 08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@XXXXXX.XXXXX.XXX>;tag=cfcb0b247c4126 c..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;branch=z9hG4bK531E08BE030E99FE369F25E3CAEC7443;rport=5061..Record-Route: <sip:v oiptest.convergenze.it:5060;nat=yes;ftag=660878483;lr=on>..Content-Length: 211..Content-Type: application/sdp..Supporte d: replaces..Contact: sip:800900800@xxx.xxx.179.28:5060..User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3 .2.8.45 00a0ba009b05....v=0..o=MxSIP 0 370160 IN IP4 xxx.xxx.179.28..s=SIP Call..c=IN IP4 xxx.xxx.179.28..t=0 0..m=audi o 15536 RTP/AVP 0 8 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U xxx.xxx.16.194:5061 -> xxx.xxx.190.51:5060 ACK sip:800900800@xxx.xxx.179.28:5060 SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport;branch=z9hG4bK5F9BAF3F87BEB79 D755EC35828AFCD6C..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@v oiptest.convergenze.it>;tag=cfcb0b247c4126c..Contact: <sip:08281962103@xxx.xxx.16.194:5061>..Route: <sip:voiptest.conve rgenze.it:5060;nat=yes;ftag=660878483;lr=on>..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 A CK..Max-Forwards: 70..Content-Length: 0.... # U xxx.xxx.190.51:5060 -> xxx.xxx.179.28:5060 ACK sip:800900800@xxx.xxx.179.28:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.190.51;lr=on;ftag=660878483>..Via: SIP/2.0/UDP xxx.xxx.190.51;branch=z9hG4bKa5cc.576d5334.2..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport=5061;branch=z9hG4bK5F9BAF3F87B EB79D755EC35828AFCD6C..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:8009008 00(a)XXXXXX.XXXXX.XXX>;tag=cfcb0b247c4126c..Contact: <sip:08281962103@xxx.xxx.16.194:5061>..Call-ID: 4CAE1C97-67A7 -8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 ACK..Max-Forwards: 69..Content-Length: 0.... # U xxx.xxx.16.194:5061 -> xxx.xxx.190.51:5060 ACK sip:800900800@xxx.xxx.179.28:5060 SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport;branch=z9hG4bK5F9BAF3F87BEB79 D755EC35828AFCD6C..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:800900800@v oiptest.convergenze.it>;tag=cfcb0b247c4126c..Contact: <sip:08281962103@xxx.xxx.16.194:5061>..Route: <sip:voiptest.conve rgenze.it:5060;nat=yes;ftag=660878483;lr=on>..Call-ID: 4CAE1C97-67A7-8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 A CK..Max-Forwards: 70..Content-Length: 0.... # U xxx.xxx.190.51:5060 -> xxx.xxx.179.28:5060 ACK sip:800900800@xxx.xxx.179.28:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.190.51;lr=on;ftag=660878483>..Via: SIP/2.0/UDP xxx.xxx.190.51;branch=z9hG4bKa5cc.576d5334.2..Via: SIP/2.0/UDP xxx.xxx.16.194:5061;rport=5061;branch=z9hG4bK5F9BAF3F87B EB79D755EC35828AFCD6C..From: 08281962103 <sip:08281962103@XXXXXX.XXXXX.XXX:5061>;tag=660878483..To: <sip:8009008 00(a)XXXXXX.XXXXX.XXX>;tag=cfcb0b247c4126c..Contact: <sip:08281962103@xxx.xxx.16.194:5061>..Call-ID: 4CAE1C97-67A7 -8A07-B3FF-4E122F113099@192.168.1.10..CSeq: 20326 ACK..Max-Forwards: 69..Content-Length: 0.... -- Thanks & regards, Krunal Patel Software Engineer Ecosmob Technologies Pvt. Ltd.

2 1

[OpenSER-Users] Mediaproxy problem - Nated->PSTN calls
by Krunal Patel 02 May '08

02 May '08

Hi, I am facing mediaproxy problem during Nated-pstn calls. Here is the problems: 1. Mediaproxy session has been broken 2. No called address on mediaproxy server 3. Mediaproxy session status is inactive 4. No bytes has been passed through mediaproxy So please help me to solve all this problems. Thanking you in advance.. - Krunal Patel

2 1

[Serusers] Ringback tone on SER
by Thorsten 01 May '08

01 May '08

Hi guys, I'm trying to set up a SER server between 2 asterisk machines. I run into 2 issues. Whenever I call someone I don't get any ringback tone even so the call initiating asterisk machine gets the 180 message after 100. <--- SIP read from 10.4.1.80:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134>;tag=as4c964973 To: <sip:017683035400@10.4.1.80> Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)82.98.89.134 CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1" <-------------> --- (9 headers 0 lines) --- mg03*CLI> <--- SIP read from 10.4.1.80:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134>;tag=as4c964973 To: <sip:017683035400@10.4.1.80>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2 Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)82.98.89.134 CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1 On SER I've configured to send this message: if (method=="INVITE") { if (uri =~ "sip:0[0-9]@*") { route(3); sl_send_reply("180", "Ringing"); break; } }; The other issue is that I don't see the caller id on the receiver side. I don't know if it is a asterisk or a SER issue. Only if I set the caller id on asterisk manual in extensions.conf with exten => _X.,1,Set(CALLERID(num)=06965006100) I'll see the caller id on the receiver side. I would really appreciate any help Thanks Thorsten

3 4

[Serusers] Asterisk SER NAT issue
by Thorsten 01 May '08

01 May '08

Hi guys, I'm trying to get this constellation running Asterisk (priavte IP)---->SER (public IP)---->Asterisk (public IP) I've already Asterisk set up to put the external IP to the SIP messages with this parameter in sip.conf externip = 82.98.89.129 but I'm not sure what I have to configure on SER. Do I need the nathelper? My config of SER looks like this: # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) #debug=0 fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #listen=10.4.1.80 port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/ser/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/lib/ser/modules/avpops.so" loadmodule "/usr/local/ser/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/lib/ser/modules/auth.so" loadmodule "/usr/local/ser/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(1); break; } if (method=="REGISTER") { record_route(); save("location"); break; }; if (method=="INVITE") { if (uri =~ "sip:0[0-9]@*") { route(3); break; } }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; #log(1, "----------------------------Ringing--------------------\n"); #sl_send_reply("180", "Ringing"); } route[3] { if (uri =~ "sip:0[0-9]@*") { log(1, "Forwarding to MG03 \n"); rewritehostport("82.98.89.176:5060"); route(1); break; } } I appreciate any help Thanks Thorsten

2 1

Re: [OpenSER-Users] [Openser] Problem with OpenSER anx openxcap integration
by piff62 piff62 01 May '08

01 May '08

Yes, I'm sure that my openser server works in the same time with openxcap. I don't understand. Thomas 2008/4/30 KARUMANCHI KARUNENDRA <karun6(a)yahoo.com>: > Hi *thomas,* > * Did you get openser working.* > *-Karun* > > ----- Original Message ---- > From: piff62 piff62 <piff62(a)gmail.com> > To: users(a)lists.openser.org > Sent: Wednesday, April 30, 2008 1:11:47 AM > Subject: [OpenSER-Users] [Openser] Problem with OpenSER anx openxcap > integration > > Hi everybody, > > I'm working about a IMS prototype architecture. > > So, I have install my server SIP : OpenSER and i have install a XDMS > (openxcap) for storage of privacy rules and presence. > > I have configured the 2 servers and they works but I already have one > error. > > This error occured when I change my privacy rules (in eyebeam). The > changes arrived to the openxcap and the openxcap want to prevent the openser > with a xmlrpc request. > > When openxcap call the xmlrpc function, i have this message error : > > *Error while notifying OpenSER management interface for 'user' > thomas@domain : <Fault -500: 'Command (refreshWatchers) processing > failed.\n'> > > *I understand that openxcap try call the openser refreshWatchers function > and failed but i don't know why ! > > someone would have already had this problem ? > > Thanks ! > > Thomas > > > ------------------------------ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now.<http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8…> >

1 0

[Serusers] msilo error
by Valentin Rosic 30 Apr '08

30 Apr '08

hello, i have problem with msilo module, actually with retrieving offline messages. messages for offline users are stored in mysql, but when they register I get -------------------------------------------------------- 0(18729) ERROR: parse_uri: uri too short: <> (0) 0(18729) ERROR: uri2dst: bad_uri: 0(18729) t_uac: no socket found -------------------------------------------------------- I found that's the same error for storing offline messages but not retrieving them -> http://lists.iptel.org/pipermail/serusers/2007-May/032978.html will applying patch do any good? p.s. my config is in attachment -- srdacan pozdrav/regards, Valentin ------------------------------------------------ Valentin Rosic SRCE - Sveucilisni racunski centar J. Marohnica 5, 10000 Zagreb, Croatia email: valentin.rosic @ srce.hr web: http://www.srce.hr tel: +385/1/6165 303 fax: +385/1/6165 559 ------------------------------------------------ # # $Id: ser.cfg,v 1.36.2.3 2007/03/08 10:51:45 calrissian Exp $ # # First start SER sample config script with: # database, accounting, authentication, multi-domain support # PSTN GW section, named flags, named routes, global-, # domain- and user-preferences with AVPs # Several of these features are only here for demonstration purpose # what can be achieved with the SER config script language. # # If you look for a simpler version with a lot less dependencies # please refer to the ser-basic.cfg file in your SER distribution. # To get this config running you need to execute the following commands # with the new serctl (the capital word are just place holders) # - ser_ctl domain add DOMAINNAME # - ser_ctl user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have PID header for your user # - ser_attr add uid=UID asserted_id="PID" # If you want to have gateway support # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 description="The gateway IP for the default ser.cfg" default_flags=33 # - ser_attr add global gw_ip=GATEWAY-IP # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) #memdbg=10 # memory debug log level #memlog=10 # memory statistics log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) # Uncomment these lines to enter debugging mode fork=no log_stderror=yes alias="161.53.2.236" alias="srce.hr" alias="xyz.hr" alias="aurora.srce.hr" alias="161.53.0.162" listen=161.53.2.236 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 #user=ser #group=ser #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # # ------------------ module loading ---------------------------------- # load a SQL database for authentication, domains, user AVPs etc. 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/xlog.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/ctl.so" loadmodule "/usr/local/lib/ser/modules/fifo.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/gflags.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avp_db.so" loadmodule "/usr/local/lib/ser/modules/acc_db.so" loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" # ----------------- setting script FLAGS ----------------------------- flags FLAG_ACC : 1 # include message in accouting avpflags dialog_cookie; # handled by rr module # ----------------- setting module-specific parameters --------------- # specify the path to you database here modparam("acc_db|auth_db|avp_db|domain|gflags|usrloc|uri_db|msilo", "db_url", "mysql://ser:heslo@127.0.0.1/ser") # -- usrloc params -- # as we use the database anyway we will use it for usrloc as well modparam("usrloc", "db_mode", 1) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # # limit the length of the AVP cookie to only necessary ones modparam("rr", "cookie_filter", "(account)") # # you probably do not want that someone can simply read and change # the AVP cookie in your Routes, thus should really change this # secret value below modparam("rr", "cookie_secret", "MyRRAVPcookiesecret") # -- gflags params -- # load the global AVPs modparam("gflags", "load_global_attrs", 1) # -- domain params -- # load the domain AVPs modparam("domain", "load_domain_attrs", 1) # -- ctl params -- # by default ctl listens on unixs:/tmp/ser_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") # listen on tcp, localhost #modparam("ctl", "binrpc", "tcp:localhost:2046") # -- acc_db params -- # failed transactions (=negative responses) should be logged to modparam("acc_db", "failed_transactions", 1) # comment the next line if you dont want to have accouting to DB modparam("acc_db", "log_flag", "FLAG_ACC") # -- tm params -- # uncomment the following line if you want to avoid that each new reply # restarts the resend timer (see INBOUND route below) #modparam("tm", "restart_fr_on_each_reply", "0") # -- xmlrpc params -- # using a sub-route from the module is a lot safer then relying on the # request method to distinguish HTTP from SIP modparam("xmlrpc", "route", "RPC"); #modparam("tm", "fr_timer", 10 ) #modparam("tm", "fr_inv_timer", 15 ) #modparam("tm", "wt_timer", 10 ) #modparam("msilo", "expire_time", 86400) # 1 day modparam("msilo", "expire_time", 120) # 120 sec modparam("msilo", "registrar", "sip:registrar@srce.hr") modparam("msilo", "check_time", 10) modparam("msilo", "clean_period", 3) modparam("msilo", "use_contact", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # if you have a PSTN gateway just un-comment the follwoing line and # specify the IP address of it to route calls to it #$gw_ip = "1.2.3.4" # first do some initial sanity checks route(INIT); # check if the request is routed via Route header or # needs a Record-Route header route(RR); # check if the request belongs to our proxy route(DOMAIN); # handle REGISTER requests route(REGISTRAR); # from here on we want to know you is calling route(AUTHENTICATION); # check if we should be outbound proxy for a local user route(OUTBOUND); # check if the request is for a local user route(INBOUND); # here you could for example try to do an ENUM lookup before # the call gets routed to the PSTN #route(ENUM); # lets see if someone wants to call a PSTN number route(PSTN); # nothing matched, reject it finally sl_reply("404", "No route matched"); } route[FORWARD] { # if the downstream UA does not support MESSAGE requests # go to failure_route[1] if (method=="MESSAGE") { t_on_failure("MESSAGE"); } # here you could decide wether this call needs a RTP relay or not log(1, "LOG: usao u FORWARD rutu \n"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } drop; } route[INIT] { log(1, "LOG: usao INIT \n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_reply("483","Too Many Hops"); drop; } if (msg:len >= max_len ) { sl_reply("513", "Message too big"); drop; } # you could add some NAT detection here for example # or you cuold call here some of the check from the sanity module # lets account all initial INVITEs # further in-dialog requests are accounted by a RR cookie (see below) if (method=="INVITE" || method=="BYE") { log(1, "LOG: usao u INVITE u INIT ruti ACC \n"); setflag(FLAG_ACC); } } route[RPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && src_ip==127.0.0.1) { if (msg:len >= 8192) { sl_reply("513", "Request to big"); drop; } # lets see if a module wants to answer this dispatch_rpc(); drop; } } route[RR] { # subsequent messages withing a dialog should take the # path determined by record-routing log(1, "LOG: usao u Record rutu (RR) \n"); if (loose_route()) { log(1, "LOG: usao u RR - Loose root funkcija \n"); # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); # if the Route contained the accounting AVP cookie we # set the accounting flag for the acc_db module. # this is more for demonstration purpose as this could # also be solved without RR cookies. # Note: this means all in-dialog request will show up in the # accouting tables, so prepare your accounting software for this ;-) if ($account == "yes") { setflag(FLAG_ACC); } # for broken devices which overwrite their Route's with each # (not present) RR from within dialog requests it is better # to repeat the RRing # and if we call rr after loose_route the AVP cookies are restored # automatically :) record_route(); route(FORWARD); } else if (!method=="REGISTER") { log("1, LOG: usao u RR- REGISTER funkcija \n"); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if the inital INVITE got the ACC flag store this in # an RR AVP cookie. this is more for demonstration purpose if (isflagset(FLAG_ACC)) { $account = "yes"; setavpflag($account, "dialog_cookie"); } record_route(); } } route[DOMAIN] { log("LOG: usao u DOMAIN rutu \n"); # check if the caller is from a local domain lookup_domain("$fd", "@from.uri.host"); # check if the callee is at a local domain lookup_domain("$td", "@ruri.host"); # we dont know the domain of the caller and also not # the domain of the callee -> somone uses our proxy as # a relay if (!$t.did && !$f.did) { sl_reply("403", "Relaying Forbidden"); drop; } } route[REGISTRAR] { xlog("L_ERR","usao u REGISTRAR rutu\n"); # if the request is a REGISTER lets take care of it if (method=="REGISTER") { xlog("L_ERR", "usao u REGISTER metodu u REGISTRAR routi \n"); # check if the REGISTER if for one of our local domains if (!$t.did) { sl_reply("403", "Register forwarding forbidden"); drop; } # we want only authenticated users to be registered if (!www_authenticate("$fd.digest_realm", "credentials")) { xlog("L_ERR", "usao u www authenticate funkciju u REGISTRAR routi \n"); if ($? == -2) { sl_reply("500", "Internal Server Error"); } else if ($? == -3) { sl_reply("400", "Bad Request"); } else { if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("401", "Unauthorized"); } drop; } # check if the authenticated user is the same as the target user if (!lookup_user("$tu.uid", "@to.uri")) { xlog("L_ERR", "usao u !lookup_user funkciju u REGISTRAR routi \n"); sl_reply("404", "Unknown user in To"); drop; } if ($f.uid != $t.uid) { sl_reply("403", "Authentication and To-Header mismatch"); drop; } # check if the authenticated user is the same as the request originator # you may uncomment it if you care, what uri is in From header #if (!lookup_user("$fu.uid", "@from.uri")) { # sl_reply("404", "Unknown user in From"); # drop; #} #if ($fu.uid != $tu.uid) { # sl_reply("403", "Authentication and From-Header mismatch"); # drop; #} # everyhting is fine so lets store the binding if (!save_contacts("location")) { xlog("L_ERR", "usao u !save_contacts funkciju u REGISTRAR routi \n"); sl_reply("400", "Invalid REGISTER Request"); drop; } # MSILO - dumping user's offline messages if (m_dump("")) { xlog("L_DBG", "MSILO: offline messages dumped - if they were\n"); } else { xlog("L_DBG", "MSILO: no offline messages dumped\n"); }; drop; } } route[AUTHENTICATION] { if (method=="CANCEL" || method=="ACK") { # you are not allowed to challenge these methods break; } # requests from non-local to local domains should be permitted # remove this if you want a walled garden if (! $f.did) { break; } # as gateways are usually not able to authenticate for their # requests you will have trust them base on some other information # like the source IP address. WARNING: if at all this is only safe # in a local network!!! #if (src_ip==a.b.c.d) { # break; #} if (!proxy_authenticate("$fd.digest_realm", "credentials")) { if ($? == -2) { sl_reply("500", "Internal Server Error"); } else if ($? == -3) { sl_reply("400", "Bad Request"); } else { if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("407", "Proxy Authentication Required"); } drop; } # check if the UID from the authentication meets the From header $authuid = $uid; if (!lookup_user("$fu.uid", "@from.uri")) { del_attr("$uid"); } if ($fu.uid != $fr.authuid) { sl_reply("403", "Fake Identity"); drop; } # load the user AVPs (preferences) of the caller, e.g. for RPID header load_attrs("$fu", "$f.uid"); } route[OUTBOUND] { # if a local user calls to a foreign domain we play outbound proxy for him # comment this out if you want a walled garden if ($f.did && ! $t.did) { append_hf("P-hint: outbound\r\n"); route(FORWARD); } } route[INBOUND] { # lets see if know the callee if (lookup_user("$tu.uid", "@ruri")) { # load the preferences of the callee to have his timeout values loaded load_attrs("$tu", "$t.uid"); # if you want to know if the callee username was an alias # check it like this #if (! $tu.uri_canonical) { # if the alias URI has different AVPs/preferences # you can load them into the URI track like this #load_attrs("$tr", "@ruri"); #} # native SIP destinations are handled using our USRLOC DB if (lookup_contacts("location")) { append_hf("P-hint: usrloc applied\r\n"); # we set the TM module timers according to the prefences # of the callee (avoid too long ringing of his phones) # Note1: timer values have to be in ms now! # Note2: this makes even more sense if you switch to a voicemail # in a FAILURE route if ($t.fr_inv_timer) { if ($t.fr_timer) { t_set_fr("$t.fr_inv_timer", "$t.fr_timer"); } else { t_set_fr("$t.fr_inv_timer"); } } route(FORWARD); } else { # User is offline # we care about MESSAGEs if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; } # store only text messages NOT isComposing... ! if (search("^(Content-Type|c):.*application/im-iscomposing\+xml.*")) { t_reply("202", "Ignored"); break; } xlog("L_NOTICE", "MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0","")) { xlog("L_NOTICE", "MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; } else { xlog("L_NOTICE", "MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; drop; } xlog("L_NOTICE","route[INBOUND] 480 User Offline"); sl_reply("480", "User Offline"); drop; } } } route[PSTN] { # Only if the AVP 'gw_ip' is set and the request URI contains # only a number we consider sending this to the PSTN GW. # Only users from a local domain are permitted to make calls. # Additionally you might want to check the acl AVP to verify # that the user is allowed to make such expensives calls. if ($f.did && $gw_ip && uri=~"sips?:\+?[0-9]{3,18}@.*") { # probably you need to convert the number in the request # URI according to the requirements of your gateway here # if an AVP 'asserted_id' is set we insert an RPID header if ($asserted_id) { xlset_attr("$rpidheader", "<sip:%$asserted_id@%@ruri.host>;screen=yes"); replace_attr_hf("Remote-Party-ID", "$rpidheader"); } # just replace the domain part of the RURI with the # value from the AVP and send it out attr2uri("$gw_ip", "domain"); route(FORWARD); } } failure_route[MESSAGE] { xlog("L_ERR", "failure route MESSAGE\n"); # forwarding failed -- check if the request was a MESSAGE if (method == "MESSAGE") { xlog("L_NOTICE", "MSILO: the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("0","")) { xlog("L_NOTICE", "MSILO: offline message stored\n"); t_reply("202", "Accepted"); } else { xlog("L_NOTICE", "MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); } } }

1 1

[OpenSER-Users] CDRTOOL SOAP/XML
by Ser 30 Apr '08

30 Apr '08

Hello, Iam New to this List and OpenSER... Does anyone has set up the CDR Tool? We set this up in 6.4.1 and have the Problem that we are not able to create a billing user. We did not found anything regarding the soap/xml. Is it Possible to get a documentation or a example user? Thank you Wolfgang

1 0

[OpenSER-Users] CDRTOOL
by Wolfgang Uelpenich 30 Apr '08

30 Apr '08

Hello, does anyone has set up the CDR Tool? We set this up in 6.4.1 and have the Problem that we are not able to create a billing user. We did not found anything regarding the soap/xml Is it Possible to get a documentation or a example user? Thank you Wolfgang Wolfgang Uelpenich WU-Solutions Ltd. & CO Kommanditgesellschaft Schnitterweg 2 65375 Oestrich-Winkel Mail: <mailto:wu@wu-solutions.com> wu(a)wu-solutions.com Web: <file:///C:\Dokumente%20und%20Einstellungen\wolfgang\Anwendungsdaten\Microso ft\Signatures\www.wu-solutions.com> <http://www.wu-solutions.com/> www.wu-solutions.com Tel: +49-6723-6298000 Fax: +49-6723-9081709 Amtsgericht Wiesbaden * HRA8736 Geschäftsleitung: Wolfgang Uelpenich

1 0

[OpenSER-Users] example config 1.3.X
by Pezhman Lali 30 Apr '08

30 Apr '08

Dear I am searching to find a config file, for registering user via database and nat traversal(rtpproxy or mediaproxy) please help me , to find the best ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[OpenSER-Users] Portability in Openser
by Aadilkhan Maniyar 30 Apr '08

30 Apr '08

Hi All, I have a query regarding the portability of OpenSER. I would like to know about the different OS and Hardware that OpenSER can be run on. Any info regarding the above would be very much helpful. Regards, Aadil

2 4

[OpenSER-Users] [Openser] Problem with OpenSER anx openxcap integration
by piff62 piff62 30 Apr '08

30 Apr '08

Hi everybody, I'm working about a IMS prototype architecture. So, I have install my server SIP : OpenSER and i have install a XDMS (openxcap) for storage of privacy rules and presence. I have configured the 2 servers and they works but I already have one error. This error occured when I change my privacy rules (in eyebeam). The changes arrived to the openxcap and the openxcap want to prevent the openser with a xmlrpc request. When openxcap call the xmlrpc function, i have this message error : *Error while notifying OpenSER management interface for 'user' thomas@domain: <Fault -500: 'Command (refreshWatchers) processing failed.\n'> *I understand that openxcap try call the openser refreshWatchers function and failed but i don't know why ! someone would have already had this problem ? Thanks ! Thomas

1 0

[OpenSER-Users] Change Openser's default behaviour for destination 0.0.0.0
by Jayesh Nambiar 30 Apr '08

30 Apr '08

Hi All, Whenever openser is made to route a call to 0.0.0.0, I have observed that it replies as 403 Forbidden to the call initiator. Is there a method to change this default behaviour and change the reply message from 403 Forbidden to 404 Not Found or any other response code. Basically I query a set of 5 destination IP addresses for every call that comes in, in a particular order and route the call in a serial manner if call fails to particular IP using failure routes. Suppose if there are only 4 IPs available for a particular destination, I substitute the 5th IP address as 0.0.0.0. Now if a user has dialed an invalid number Openser keeps routing the call to next IP address until it comes to 0.0.0.0. As soon as it gets 0.0.0.0 it replies back as 403 Forbidden, in this case I want to reply back as 404 Not Found which is a better user reply in case of an invalid number. Any pointers/suggestions will be highly appreciated. Thanks, --- Jayesh Connect with friends all over the world. Get Yahoo! India Messenger at http://in.messenger.yahoo.com/?wm=n/

1 0

[OpenSER-Users] Problems with SerMyAdmin Installation
by Mark Koopmann 29 Apr '08

29 Apr '08

Hi everybody, does anybody has experience with installing the java tool SerMyAdmin? The Tool is running but every action ends with an runtime exception. Trying to register a new user ends with the following message: Grails Runtime Exception Error Details Message: Caused by: java.lang.NullPointerException Class: RegisterUserController At Line: [98] Code Snippet: (empty) Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: java.lang.NullPointerException 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: java.lang.NullPointerException at RegisterUser.toString(RegisterUser.groovy) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy:98) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy) ... 29 more Unfortunately, i dont have much experience with installing such java stuff. It would be great if one of you can give me a hint or something. I dont have any idea how to solve this problem. Thanks for any comment! Best regards Mark Koopmann

1 0

[OpenSER-Users] Cannot authenticate SUBSCRIBE
by Eric PTAK 29 Apr '08

29 Apr '08

Hi list ! I'm building an agent and currently working on Digest authentication using auth and auth_db modules. I'm trying to authenticate all requests for compliance test purpose. It works fine with REGISTER but I can't authenticate for others methods like SUBSCRIBE, the module is looping authentication. This is a part of my config file : if (uri==myself) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; // others things }; And this is the requests / responses : REGISTER sip:sip.awl.test SIP/2.0 Call-ID: 0f13c7c272f62943f2f6345c07249fcd(a)10.24.238.150 CSeq: 1 REGISTER From: <sip:0000@sip.awl.test>;tag=9c0477b2 To: <sip:0000@sip.awl.test> Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bKb71ee89041bd9f3631cf46a8fc76b46c Max-Forwards: 70 Contact: <sip:0000@10.24.238.150:23456 ;transport=tcp>;expires=3600,<sip:0000@10.24.238.150:23456 ;transport=udp>;expires=3600 Content-Length: 0 SIP/2.0 401 Unauthorized Call-ID: 0f13c7c272f62943f2f6345c07249fcd(a)10.24.238.150 CSeq: 1 REGISTER From: <sip:0000@sip.awl.test>;tag=9c0477b2 To: <sip:0000@sip.awl.test>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8696 Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bKb71ee89041bd9f3631cf46a8fc76b46c WWW-Authenticate: Digest realm="sip.awl.test", nonce="4811de4caf098eb6f25ea5db9a788578a2d4be35" Server: OpenSER (1.3.1-notls (i386/linux)) Content-Length: 0 REGISTER sip:sip.awl.test SIP/2.0 Call-ID: 0f13c7c272f62943f2f6345c07249fcd(a)10.24.238.150 CSeq: 2 REGISTER From: <sip:0000@sip.awl.test>;tag=9c0477b2 To: <sip:0000@sip.awl.test>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8696 Max-Forwards: 70 Contact: <sip:0000@10.24.238.150:23456 ;transport=tcp>;expires=3600,<sip:0000@10.24.238.150:23456 ;transport=udp>;expires=3600 Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bK4bff3e93bf1dc502db28c4d414e60b4f Authorization: Digest response="efb87138e6340d7a80a1f13ef26821da",username="0000",realm="sip.awl.test",nonce="4811de4caf098eb6f25ea5db9a788578a2d4be35",uri="sip:sip.awl.test",algorithm=MD5 Content-Length: 0 SIP/2.0 200 OK Call-ID: 0f13c7c272f62943f2f6345c07249fcd(a)10.24.238.150 CSeq: 2 REGISTER From: <sip:0000@sip.awl.test>;tag=9c0477b2 To: <sip:0000@sip.awl.test>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8696 Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bK4bff3e93bf1dc502db28c4d414e60b4f Contact: <sip:0000@10.24.238.150:23456;transport=udp>;expires=3600, <sip:0000@10.24.238.150:23456;transport=tcp>;expires=3600 Server: OpenSER (1.3.1-notls (i386/linux)) Content-Length: 0 SUBSCRIBE sip:0000@sip.awl.test SIP/2.0 Call-ID: d27f913809ccfa308b497ce92c3d01ef(a)10.24.238.150 CSeq: 1 SUBSCRIBE From: <sip:0000@sip.awl.test>;tag=3b68764f To: <sip:0000@sip.awl.test> Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bKae0ab5a2b4dd462eb139276b814b413c Max-Forwards: 70 Contact: <sip:0000@10.24.238.150:23456> Expires: 3600 Event: presence.winfo Content-Length: 0 SIP/2.0 401 Unauthorized Call-ID: d27f913809ccfa308b497ce92c3d01ef(a)10.24.238.150 CSeq: 1 SUBSCRIBE From: <sip:0000@sip.awl.test>;tag=3b68764f To: <sip:0000@sip.awl.test>;tag=329cfeaa6ded039da25ff8cbb8668bd2.78ff Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bKae0ab5a2b4dd462eb139276b814b413c WWW-Authenticate: Digest realm="sip.awl.test", nonce="4811de4caf098eb6f25ea5db9a788578a2d4be35" Server: OpenSER (1.3.1-notls (i386/linux)) Content-Length: 0 SUBSCRIBE sip:0000@sip.awl.test SIP/2.0 Call-ID: d27f913809ccfa308b497ce92c3d01ef(a)10.24.238.150 CSeq: 2 SUBSCRIBE From: <sip:0000@sip.awl.test>;tag=3b68764f To: <sip:0000@sip.awl.test>;tag=329cfeaa6ded039da25ff8cbb8668bd2.78ff Max-Forwards: 70 Contact: <sip:0000@10.24.238.150:23456> Expires: 3600 Event: presence.winfo Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bK0afcac1c88fce00c3bfca270ed379ee4 Authorization: Digest response="efb87138e6340d7a80a1f13ef26821da",username="0000",realm="sip.awl.test",nonce="4811de4caf098eb6f25ea5db9a788578a2d4be35",uri="sip:sip.awl.test",algorithm=MD5 Content-Length: 0 SIP/2.0 401 Unauthorized Call-ID: d27f913809ccfa308b497ce92c3d01ef(a)10.24.238.150 CSeq: 2 SUBSCRIBE From: <sip:0000@sip.awl.test>;tag=3b68764f To: <sip:0000@sip.awl.test>;tag=329cfeaa6ded039da25ff8cbb8668bd2.78ff Via: SIP/2.0/UDP 10.24.238.150:23456 ;branch=z9hG4bK0afcac1c88fce00c3bfca270ed379ee4 WWW-Authenticate: Digest realm="sip.awl.test", nonce="4811de4caf098eb6f25ea5db9a788578a2d4be35" Server: OpenSER (1.3.1-notls (i386/linux)) Content-Length: 0 As you can see, auth module always use nonce="4811de4caf098eb6f25ea5db9a788578a2d4be35" in its responses, for both register and subscribe. thanks for your help, Eric.

2 2

[OpenSER-Users] userblacklist
by Helmut Kuper 29 Apr '08

29 Apr '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is it possible to add a way to userblacklist so that we have two tables (one for incomming and one for outgoing (user|global)blacklists) ? I enhanced userblacklist module following the sledge hammer method (just duplicating the code) so I have now a 2nd userblacklist table which can be requested by my own check_user_blacklist2() function controlled by 2 additional modparams (use_domain2, db_table2) It would be nice to have one check_blacklist function with a tablename as an additional parameter. Or is it designed for using in two separate proxies (one incomming and one outgoing proxy)? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgE1mgACgkQ4tZeNddg3dwR/gCcDlq80Yj7sKzXejhcvIhlK5M+ cjcAoJluwmkS/4iuR8NXnIWhdbeB8bn5 =H1M5 -----END PGP SIGNATURE-----

2 10

[OpenSER-Users] Bigger OpenSER installation (2-5 Mio subs) and High Availability
by Adrian Georgescu 29 Apr '08

29 Apr '08

Hi Peter, If you care for a comercial solution that scales linearly with the amount of hardware without using any load balancers here is one: http://www.ag-projects.com/SIPThor.html Regards, Adrian >>>>>>>>>>>>>>> Hello, I have done a project for a SIP provider who is rapidly growing. Designed as a SBC for about 50.000 subscribers at first, I setup a system with a high availability cluster (HA/DRBD) on 2 servers which had one single database on the DRBD volume. Failover works like a charm with only very limited downtime. Now we have to consider to scale the system to about 2-5 Mio subs, so I have to challenge with some things I cannot oversee at the moment. My question is: How do you deal with big installations? - how do you load balance the openser? Is it advisable put put some dedicated SIP hardware in front for load balancing? Or is the dispatcher module on a seperate machine sufficiant (again HA-issue)? - we have a portion of calls who have to go through mediaproxy (How do you load balance this?) - Do you face database issues? We think about setting up a database on 2 seperate servers. Is MySQL ok for this or do you use other DBMS? - Form your exoperience: Are there other things I have to consider? Best regards and thanks for your support. Peter

3 4

[OpenSER-Users] OPENSER Dinner - May 6, Barcelona, Spain
by Daniel-Constantin Mierla 29 Apr '08

29 Apr '08

Hello everybody, as previously announced couple of weeks ago, there is going to be an OpenSER Social Networking Event in Barcelona, Spain, May 6, 9:00PM at "Braseria El Glop". http://www.elglop.com/ The primary goal of the meeting is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to any of these events is free, everybody pays for its expenses. Anyone interested in openser, sip or voip in general is welcome to join. So far, I got confirmation to participate from about 10 persons: - Olle E. Johansson, main Asterisk SIP developer, CEO at http://www.edvina.net - Elena-Ramona Modroiu, co-founder OpenSER, COO at http://www.asipto.com - Jesus Rodriguez, developer OpenSER -- FreeBSD packages maintainer, CTO at http://www.voztele.com - Inaki Baz Castillo - OpenSER mailing list activist, VoIP consultant - myself :-) - several other openser, wesip (http://www.wesip.com) and asterisk folks If you need more details about location, Jesus can assist in English and Spanish. If you didn't yet, please write me or Jesus before the end of this week if you plan to come and want to be sure a seat is reserved for you. Cheers, Daniel Previous event of such type was in Bucharest Romania, April 12 -- here are some photos: http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/data/Sm_IMG_1908.JPG.html http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/data/Sm_IMG_1900.JPG.html Blogged a bit about at: http://miconda.wordpress.com/2008/04/13/the-rosdev2008-weekend/ -- http://www.asipto.com

1 0

[Serusers] UAC MODULE Support on SER 2.0.0-rc1
by Alberto Cruz 28 Apr '08

28 Apr '08

Hi, I tried to find the patch "replace_from.patch" for the UAC module in the "/module/uac/doc" path but it isn't there. Is there any change about this information or is the UAC module not longer supported in SER 2.0.0-rc1? Regards Alberto Cruz

1 0

[OpenSER-Users] IPv6 issues in openser
by Klaus Darilion 28 Apr '08

28 Apr '08

Hi! Following some IPv6 issues: 1) I configured openser (1.3.0) to listen on an IPv6 address too: listen = udp:[2001:7b8:2ff:1da::2]:5060 listen = tcp:[2001:7b8:2ff:1da::2]:5060 I also configured bind9 and apache2 to listen on this IPv6 address. But somehow openser does it different then others: # netstat -anp|grep 2001 tcp6 0 0 2001:7b8:2ff:1da:::5060 :::* tcp6 0 0 2001:7b8:2ff:1da::2:80 :::* tcp6 0 0 2001:7b8:2ff:1da::2:53 :::* udp6 0 0 2001:7b8:2ff:1da::2:53 :::* udp6 0 0 2001:7b8:2ff:1da:::5060 :::* As you see, openser binds to [2001:7b8:2ff:1da::] whereas others binds to [2001:7b8:2ff:1da::2]. Actually this didn't caused any problems, but I still surprised about the difference. Can someone explain me please this behavior? 2) I found out that openser allows this syntax too (without []). listen = udp:2001:7b8:2ff:1da::2 I think it would be better to disallow that syntax and require [] as it is ambiguous when specifying the port too, e.g.: listen = udp:2001:7b8:2ff:1da::2:5060 3) Looks like the syntax in the socket column of the location table is not IPv6 compatible: ERROR:usrloc:parse_phostport: too many colons in udp:2001:7B8:2FF:1DA:0:0:0:2:5060 ERROR:usrloc:dbrow2info: bad socket <udp:2001:7B8:2FF:1DA:0:0:0:2:5060> ERROR:usrloc:preload_udomain: sipping record for klaus1 in table location I guess the socket should use [] for IPv6 addresses too. regards Klaus

2 3

[OpenSER-Users] Tie BYE to INVITE?
by Douglas Garstang 28 Apr '08

28 Apr '08

Is there a way I can tie a BYE to an INVITE? The setflag() and isflagset() commands only work on a transaction, right? And, a BYE is a new transaction, so how could I tell if a BYE is related to a previous INVITE? Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

3 2

[OpenSER-Users] [NEW] First OpenSER teaching book
by Bogdan-Andrei Iancu 28 Apr '08

28 Apr '08

**Hi everybody, I'm really happy to announce that the first one hundred percent book dedicated to the OpenSER software is finally out. *"Building Telephony Systems with OpenSER"* goes where no one has went before - writing a book to help people to understand OpenSER software and to teach them how to setup OpenSER based platforms - installation, configuration, integration. Such a book was waited by community for some time, and there are many thinks about it to be said - for more on this book, please see the announcement on the openser web page - http://www.openser.org/mos/view/News/NewsItem/OpenSER-First-Book/ Regards, Bogdan

2 2

[OpenSER-Users] SIGILL when starting openser
by Christian Koch 28 Apr '08

28 Apr '08

Hi, when starting openser 1.3.1 it doesn't really start, but stops after processing a SIGILL. When perfoming strace -f /usr/local/sbin/openser I get the following output (these are only the last few lines): time([1208951399]) = 1208951399 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 send(3, "<143>Apr 23 13:49:59 /usr/local/"..., 147, MSG_NOSIGNAL) = 147 --- SIGILL (Illegal instruction) @ 0 (0) --- Process 12746 detached I've compiled openser from source on the same machine. When starting openser with -V everything works just fine. Output: version: openser 1.3.1-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 3590 2008-01-28 17:46:56Z bogdan_iancu $ main.c compiled on 10:47:24 Apr 23 2008 with gcc 3.4.5 So I guessed the problem could occur inside a module. I removed auth.so and auth_db.so from my config file, but openser still doesn't start. But, it doesn't receive a SIGILL anymore, but SIGTERM. Output from strace: time([1208951661]) = 1208951661 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 send(3, "<139>Apr 23 13:54:21 /usr/local/"..., 103, MSG_NOSIGNAL) = 103 kill(0, SIGTERM) = 0 --- SIGTERM (Terminated) @ 0 (0) --- sigreturn() = ? (mask now []) time([1208951661]) = 1208951661 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 [...] stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=837, ...}) = 0 send(3, "<143>Apr 23 13:54:21 /usr/local/"..., 113, MSG_NOSIGNAL) = 113 munmap(0xb5f58000, 33554432) = 0 exit_group(-1) = ? Process 12752 detached Any ideas how to fix this problem? Thx, Christian

3 9

[OpenSER-Users] ERROR:presence:mod_init: Wrong version v-1 for table <presentity>, need v2
by Andrea Polidoro 28 Apr '08

28 Apr '08

Dear all, I'm trying to install in a debian sid machine the openser 1.3 in order to use it as presence server. I downloaded the openser and the presence module from the official debian repository. As database I'm using mysql. Unfortunately, when I try to start the openSer I receive this error: "ERROR:presence:mod_init: Wrong version v-1 for table <presentity>, need v2" I found the presentity table structure in this link: http://www.openser.org/docs/db-tables/openser-db-1.3.x.html#GEN-DB-PRESENTI… and I compared it with the presentity table in my system (this is the output): mysql> describe presentity; +---------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +---------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | username | varchar(64) | NO | MUL | NULL | | | domain | varchar(64) | NO | | NULL | | | event | varchar(64) | NO | | NULL | | | etag | varchar(64) | NO | | NULL | | | expires | int(11) | NO | | NULL | | | received_time | int(11) | NO | | NULL | | | body | blob | NO | | NULL | | +---------------+------------------+------+-----+---------+----------------+ 8 rows in set (0.03 sec) How you can see, they seam identical. I searched this error in your mailing list but I didn't find anything. Please, could you help me to understood witch is the error? the presence parameters in my openser.cfg are: modparam("presence", "db_url", "mysql://openser:openserrw@127.0.0.1/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "server_address", "sip:192.168.100.22:5060") Thank you in advance for your help, Best Regards Andrea Polidoro -- ******************************************** Andrea Polidoro PhD Student University of Rome "Tor Vergata" Department of Electronics Engineering Via del Politecnico, 1 - 00133 Rome - ITALY phone: +39 06 7259 7773 mobile: +39 320 4391892 email: andrea.polidoro(a)uniroma2.it ********************************************

2 1

[OpenSER-Users] Global variables in scripts.
by Peter Lemenkov 28 Apr '08

28 Apr '08

Hello All! Are any ways to define global variables in openser's scripts? I need somethng like ========================================== $var(router1) = "192.168.1.1" $var(router2) = "192.168.1.2" route { ... myfunc1("$ru") switch($retcode) case 1: $ruri = "sip:" + $rU + "@" + $var(route1); case 2: $ruri = "sip:" + $rU + "@" + $var(route2); } failure_route[1] { myfunc2("$ru") switch($retcode) case 1: $ruri = "sip:" + $rU + "@" + $var(route1); case 2: $ruri = "sip:" + $rU + "@" + $var(route2); } onreply_route[1] { myfunc3("$ru") switch($retcode) case 1: $ruri = "sip:" + $rU + "@" + $var(route1); case 2: $ruri = "sip:" + $rU + "@" + $var(route2); } ========================================== Right now (AFAIK) variables may be defined only within one route. Maybe I missing something? What's the scope of visibility of variables within configuration script? -- With best regards!

6 6

[OpenSER-Users] ERROR:presence:mod_init: Wrong version v-1 for table <presentity>, need v2
by Andrea Polidoro 28 Apr '08

28 Apr '08

Dear all, I'm trying to install in a debian sid machine the openser 1.3 in order to use it as presence server. I downloaded the openser and the presence module from the official debian repository. As database I'm using mysql. Unfortunately, when I try to start the openSer I receive this error: "ERROR:presence:mod_init: Wrong version v-1 for table <presentity>, need v2" I found the presentity table structure in this link: http://www.openser.org/docs/db-tables/openser-db-1.3.x.html#GEN-DB-PRESENTI… and I compared it with the presentity table in my system (this is the output): mysql> describe presentity; +---------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +---------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | username | varchar(64) | NO | MUL | NULL | | | domain | varchar(64) | NO | | NULL | | | event | varchar(64) | NO | | NULL | | | etag | varchar(64) | NO | | NULL | | | expires | int(11) | NO | | NULL | | | received_time | int(11) | NO | | NULL | | | body | blob | NO | | NULL | | +---------------+------------------+------+-----+---------+----------------+ 8 rows in set (0.03 sec) How you can see, they seam identical. I searched this error in your mailing list but I didn't find anything. Please, could you help me to understood witch is the error? the presence parameters in my openser.cfg are: modparam("presence", "db_url", "mysql://openser:openserrw@127.0.0.1/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "server_address", "sip:192.168.100.22:5060") Thank you in advance for your help, Best Regards Andrea Polidoro -- ******************************************** Andrea Polidoro PhD Student University of Rome "Tor Vergata" Department of Electronics Engineering Via del Politecnico, 1 - 00133 Rome - ITALY phone: +39 06 7259 7773 mobile: +39 320 4391892 email: andrea.polidoro(a)uniroma2.it ********************************************

1 0

[OpenSER-Users] Openser Transaction State Machine
by Aadilkhan Maniyar 28 Apr '08

28 Apr '08

Hi, We are in the process of evaluating the architecture of OpenSER for a local deployment. For this purpose I was looking at the State Machine implemented in OpenSER. More specifically I am looking at the state machine related to the INVITE transaction. In order to know the State machine, what files do I have to look at? Can anyone please guide me. Has the state machine been documented at some place, so that I can read about it? Any help would be appreciated. Thanks a lot, Aadil

2 1

[OpenSER-Users] mediaproxy server NEW FEATURE!!!
by Gonzalo J. Sambucaro 28 Apr '08

28 Apr '08

Folks, I developed a new feature in the Media Proxy project of AG-Proyect. With this new feature the Media Proxy Server is now able to manage the problem of the NAT IP change in an established call. It is currently in production and working without any problem. Somebody wants to test it? I would like to know what to do to make this feature available to everybody. Does anybody know? Regards, ------------------------------ Details: ------- Supose that there is an established rtp connection between two endpoints and the media proxy is in the middle doing the relay of the rtp streams, let´s say (MP = mediaproxy) EP A <--------->[NAT with IP1]<-----------> MP <--------------------> EP B EP A sends rtp to MP_IP:MP_PORT passing through the NAT box. EP B sends rtp to MP_IP:MP_PORT without passing through a NAT box. The MP know that the caller = NAT_IP1:NAT_PORT1, and the called = EP_B_IP:EP_B_PORT Now, supose that the NAT box change their PUBLIC IP from IP1 to IP2, so this escenary EP A <------->[NAT with IP1]<---------> MP <-------------> EP B will change to this EP A <------->[NAT with IP2]<---------> MP <--------------> EP B so the MP should detect that change of IPs and continue relaying the rtp streams but now to IP2:PORT2 instead of IP1:PORT1. Well, that was the situation y have experienced. To fix this, I developed this solution, changing rtphandler.py file: 1) When the first rtp packet of a source arrives, save the SSRC field in the MP. - Save the SSRC of the caller. - Save the SSRC of the called. 2) If arrives a rtp packet with unknown source IP but with the same SSRC field of some of the two streams, updates the binding (with the new IP detected) between the caller and the MP or between the called and the MP according to the field SSRC previously saved. Note: SSRC (RFC 3550 RTP), (from the rfc: "The SSRC identifier carried in the RTP header and in various fields of RTCP packets is a random 32-bit number that is required to be globally unique within an RTP session ") -------------------------------- -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo.sambucaro(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo.sambucaro(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

5 8

[OpenSER-Users] Speeddial
by CSB 28 Apr '08

28 Apr '08

Is it possible to use speeddial to associate one number with two destinations. E.g. user dials 10 which then calls both sip:bob@domain.com and sip:jane@domain.com. If not then could anyone suggest an alternative scheme whereby a user dials a short code which can be associated with multiple URIs for parallel or serial forking? Thanks Cameron

2 1

[OpenSER-Users] Bigger OpenSER installation (2-5 Mio subs) and High Availability
by Peter P GMX 28 Apr '08

28 Apr '08

Hello, I have done a project for a SIP provider who is rapidly growing. Designed as a SBC for about 50.000 subscribers at first, I setup a system with a high availability cluster (HA/DRBD) on 2 servers which had one single database on the DRBD volume. Failover works like a charm with only very limited downtime. Now we have to consider to scale the system to about 2-5 Mio subs, so I have to challenge with some things I cannot oversee at the moment. My question is: How do you deal with big installations? - how do you load balance the openser? Is it advisable put put some dedicated SIP hardware in front for load balancing? Or is the dispatcher module on a seperate machine sufficiant (again HA-issue)? - we have a portion of calls who have to go through mediaproxy (How do you load balance this?) - Do you face database issues? We think about setting up a database on 2 seperate servers. Is MySQL ok for this or do you use other DBMS? - Form your exoperience: Are there other things I have to consider? Best regards and thanks for your support. Peter

2 1

[OpenSER-Users] Proxy Authorization - Two Digests
by Adrian Georgescu 28 Apr '08

28 Apr '08

Ash, You can set up the SIP Proxies to use the same nonce. Adrian >>>>>>>>>> We are having a separate issue now with our ATA - the ATA (a cheap generic brand) we use as our CPE, won't issue two digest authentication response unlike X-Lite. Seems like we will be forced to use authentication at B2BUA level only after all. Anyone knows any ATA positively supports sending two separate authentication digests for two separate nonce in same INVITE? Ovidiu Sas wrote: > I don't think it is reported. > > On Fri, Apr 25, 2008 at 6:36 AM, Iñaki Baz Castillo <ibc at in.ilimit.es> wrote: > >> El Friday 25 April 2008 09:57:09 Ovidiu Sas escribió: >> >> >>> Yes, it should work but it doesn't. >>> >> > I remember trying this long time ago (using openser 1.0 and chaining >> > several openser proxies) and I hit the same issue: openser is not able >> > to deal properly with multiple Proxy-Authorization headers. >> >> So this is a bug (IMHO an important bug), isn't? is it reported? >> >> >> -- >> >>

1 0

[OpenSER-Users] interaction between msrp and openser
by Adrian Georgescu 28 Apr '08

28 Apr '08

Hi Gaoqiang, MSRPRelay is not dependent on OpenSER. You can set it up separately based on the instructions available at http://msrprelay.org. You only need a client with support for the MSRP relay extension, the SIP session can be established by any version of OpenSER. I believe you refer to the MSRP chat server when you say "session switch". MSRP chat server is work in progress, no software has been released yet, we plan to release a first version this summer. Regards, Adrian >>> Hi list, Could anyone here tell me is the openser 1.3.x version support MSRP(e.g. MSRPRelay)? Is there any document describe how to configure the openser work with MSRP(MSRPRelay) as a message session switch. Does anyone have successful experience configure openser to work with the MSRP Relay. Thanks a lot. -- Gaoqiang Qian Automation System and Technology Helsinki University Of Technology GSM: +358415058138

1 0

[OpenSER-Users] Force OpenSER to send all SIP messages to src_port
by Höskuldur 27 Apr '08

27 Apr '08

Hi, How can I force OpenSER to send all SIP messages to source port of the UA? That is, the same port as userloc lists in MySQL/openser/location/received column? For some reason it only sends the Invite message to that port, but then all Info,Ack,Ok messages go to the port found in VIA...which the UA doesn't receive because it's behind NAT. I've tried force_rport(); without any success, but I must admit that I'm not sure I used it in the right route. My config is previously posted here: http://www.voipuser.org/forum_topic_12946.html Any ideas?

1 0

[Serusers] RTPProxy
by Strassburger, Patrick 27 Apr '08

27 Apr '08

> Hi all, > > Is there an easy way to disable RTPProxy for certain calls eg. based > on called number? > Am using the ser.cfg from > http://www.iptel.org/ser/howtos/optimizing_the_use_of_rtp_proxy. > > For whatever reason all calls are passing the RTPProxy. > - I would like to disable the RTPProxy for calls where I know the > destination is able to handle NAT. > - For calls where I know both UAT are behind the same NAT. > As soon as I update the realm of those clients with whatever > value beside the ser IP adress, those useres cannot login anymore. > > Is this sufficient information - do you need the ser.cfg? > > Thanks > Patrick > > > ************************************************************************************* The message is intended for the named addressee only and may not be disclosed to or used by anyone else, nor may it be copied in any way. The contents of this message and its attachments are confidential and may also be subject to legal privilege. If you are not the named addressee and/or have received this message in error, please advise us by e-mailing security(a)colt.net and delete the message and any attachments without retaining any copies. Internet communications are not secure and COLT does not accept responsibility for this message, its contents nor responsibility for any viruses. No contracts can be created or varied on behalf of COLT Telecommunications, its subsidiaries or affiliates ("COLT") and any other party by email Communications unless expressly agreed in writing with such other party. Please note that incoming emails will be automatically scanned to eliminate potential viruses and unsolicited promotional emails. For more information refer to www.colt.net or contact us on +44(0)20 7390 3900.

5 4

[OpenSER-Users] SIP SIMPLE - XMPP interoperability issue
by Gaoqiang Qian 27 Apr '08

27 Apr '08

Hello everyone, I have setup the openser server with the very basic default configuration. Now I have to configure my openser server to connect with a xmpp envoironment on another host with a purpose of piloting the SIP SIMPLE - XMPP IOP. So does anyone here have successful experience of configure openser to connect with XMPP on different hosts? Or could someone provide me some instruction how to configure it. Thanks a lot in advance -- Gaoqiang Qian Automation System and Technology Helsinki University Of Technology GSM: +358415058138

3 4

[OpenSER-Users] OpenSER+mediaproxy changes SIP dst port so client never receives ACK, causing timeout
by Höskuldur 27 Apr '08

27 Apr '08

I hope I'm not waisting your time with a problem that has a obvious solution, I've actually tried to find a similar case without any success. I'm making a call between two NATed clients which are using STUN. The call goes through OpenSER which is configured to use mediaproxy - see my config at the bottom of my email Below is a dump from the location table: MySQL / openser / location _______________________________________________________________________ | contact | received | flag | ----------------------------------------------------------------------- |sip:1007@130.XXX.XXX.XXX:49162 |sip:130.XXX.XXX.XXX:49163 | 1 | |sip:1003@85.XXX.XXX.XXX:3676 | sip:85.XXX.XXX.XXX:3688 | 1 | ----------------------------------------------------------------------- The problem is that the call from 1003 to 1007 initiates OK, but disconnects after 20-30 seconds. From my logs it looks like OpenSER stops sending SIP messages to port 49163 after getting SDP from 1007, but starts sending to 49162 instead, so 1007 never gets the ACK message. So 1007 repeatedly sends SDP description but OpenSER replies to the wrong port. Please see the attached picture, graph from Wireshark.If the picture doesn't attach, it can be found here: http://hoski.public.is/openser.jpg So my question is, why does OpenSER stop sending SIP messages to port: 49163 and how can I fix it? Thanks in advance, Hoski listen = XX.XXX.XX.XXX mpath = "/usr/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility=LOG_LOCAL7 log_stderror = no tcp_children = 4 fifo = "/tmp/openser_fifo" fifo_db_url="mysql://openser:XXXX@localhost/openser" # ATH: PrÃ³eftir BOSTON fifo_group = "openser" fifo_mode = 0666 fifo_user = "openser" mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = no check_via = no dns = no rev_dns = no syn_branch = yes disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 30) loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) 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", "retr_timer1p1", 1) modparam("tm", "retr_timer1p2", 2) modparam("tm", "retr_timer1p3", 4) modparam("tm", "retr_timer2", 4) 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", "fr_timer_avp", "s:callee_fr_timer") modparam("tm", "fr_inv_timer_avp", "s:callee_fr_inv_timer") modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 1) loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:XXXXXXX@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 180) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "received_avp", 42) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") loadmodule "textops.so" loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "use_domain", 0) modparam("registrar", "case_sensitive", 0) modparam("registrar", "desc_time_order", 0) modparam("registrar", "received_avp", 42) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "use_branch_flags", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "nat_flag", 6) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "s:rpid") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:XXXXXX@localhost/openser") 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 "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:XXXXXXX@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[1] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); use_media_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("1"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); if(is_method("INVITE") && isflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } if(nat_uac_test("19")) { fix_nated_contact(); setflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$ruri")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); } } exit; } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[1] { route(10); route(1); }

1 2

[OpenSER-Users] Proxy Authorization - Two Digests
by Ash Rah 26 Apr '08

26 Apr '08

Hello, I am trying to make a design like below to work. X-Lite ----- OpenSER ----- Asterisk ----->(PSTN Calls) X-Lite registers with OpenSer and PSTN calls are routed through Asterisk from OpenSER. When a call is sent to Asterisk, Asterisk tries to authenticate the user on X-Lite. I maintain same username and password for both OpenSER and Asterisk. Now when an INVITE from X-Lite hits OpenSER, it goes through the following script and is asked for Proxy Authorization: if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } When I dial a PSTN number from X-Lite, X-Lite at some point, ends up sending two Digests (one for OpenSER and one for Atserisk) in same INVITE but gets stuck with Proxy Authorization failure (from OpenSER). If I take off the above proxy_authorize section from OpenSER script, everything works fine. Can anyone suggest a solution to this. Thanks in advance. U 2008/04/23 13:28:42.314669 110.110.110.110:26986 -> 120.120.120.120:5060 INVITE sip:6048484848484@sip.dummydomain.com SIP/2.0. Via: SIP/2.0/UDP 172.16.40.14:26986;branch=z9hG4bK-d87543-886860777744b40e-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1274229212@110.110.110.110:26986>. To: "6048484848484"<sip:6048484848484@sip.dummydomain.com>. From: "1274229212"<sip:1274229212@sip.dummydomain.com>;tag=7d74b26b. Call-ID: ZjIyNDQzOWIxZTM2MWJjMTgzNmE1YWE3ZDY1M2RjZWE.. CSeq: 3 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1274229212",realm="asterisk",nonce="01d3972c",uri="sip:6048484848484@sip.dummydomain.com",response="ff9058f8ea89c55d0b110d4eccf27e9c",algorithm=MD5. Proxy-Authorization: Digest username="1274229212",realm="sip.dummydomain.com",nonce="480ee655da312e1c8f977cae40a747d26f7e9c5f",uri="sip:6048484848484@sip.dummydomain.com",response="361700cce632c00ff70ede5e5126c6ac",algo rithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 333. . v=0. o=- 9 2 IN IP4 172.16.40.14. s=CounterPath X-Lite 3.0. c=IN IP4 172.16.40.14. t=0 0. m=audio 45136 RTP/AVP 0 101. a=alt:1 3 : gpvy8HMY JXNZYRF+ 172.16.40.14 45136. a=alt:2 2 : 8S3XPC3M 6q9Z76Pq 192.168.38.1 45136. a=alt:3 1 : rISpUdBc PRYZ7B/8 192.168.23.1 45136. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 2008/04/23 13:28:42.314910 120.120.120.120:5060 -> 110.110.110.110:26986 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 172.16.40.14:26986;branch=z9hG4bK-d87543-886860777744b40e-1--d87543-;rport=26986;received=110.110.110.110. To: "6048484848484"<sip:6048484848484@sip.dummydomain.com>;tag=058e81974577b8ca6a831d36c0f6fe25.d85d. From: "1274229212"<sip:1274229212@sip.dummydomain.com>;tag=7d74b26b. Call-ID: ZjIyNDQzOWIxZTM2MWJjMTgzNmE1YWE3ZDY1M2RjZWE.. CSeq: 3 INVITE. Proxy-Authenticate: Digest realm="sip.dummydomain.com", nonce="480ee6560e7141c28e990448575d0918ce86a82d". Server: OpenSER (1.3.1-notls (i386/linux)). Content-Length: 0.

8 15

[OpenSER-Users] interaction between msrp and openser
by Gaoqiang Qian 26 Apr '08

26 Apr '08

Hi list, Could anyone here tell me is the openser 1.3.x version support MSRP(e.g. MSRPRelay)? Is there any document describe how to configure the openser work with MSRP(MSRPRelay) as a message session switch. Does anyone have successful experience configure openser to work with the MSRP Relay. Thanks a lot. -- Gaoqiang Qian Automation System and Technology Helsinki University Of Technology GSM: +358415058138

1 0

[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 26 Apr '08

26 Apr '08

I will be out of the office starting 04/23/2008 and will not return until 04/29/2008.

1 0

[Serusers] Multiple INVITEs crash Polycom phones
by Steven C. Blair 25 Apr '08

25 Apr '08

Hello: I have a case where multiple INVITEs crash Polycom phones. I'm wondering if anyone else has seen this and if so how you addressed the problem. We are using SER0.9.7pre3. We have Polycom 550s and Cisco 7960s running SIP 2.2.2.0084 and 7.3 respectively. These phones are configured for simultaneous/parallel ringing which we call a ring group. In a ring group the same 5-digit extension appears on each phone. So in this case the AOR has three contact addresses in the location table. When a call arrives for the AOR all three phones ring and if you answer one of them at least one, possible all, Polycom phones will reboot. Ngrep traces on the proxy show multiple invites being sent to each phone each with a different branch tag in the Via header. I'm not sure what is happening so any insight anyone can share will be appreciated. Polycom admits the reboot is a bug but believes the multiple invites is a configuration problem in our proxy. My questions is where and why does this only cause a problem for Polycom phones? Thanks,steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

1 0

[OpenSER-Users] Fwd: Install Perl Module
by Marcio 25 Apr '08

25 Apr '08

---------- Forwarded message ---------- From: Marcio <mrgalhan(a)gmail.com> Date: 23/04/2008 21:06 Subject: Re: [OpenSER-Users] Install Perl Module To: Mik Cheez <michael_bulk(a)wildgate.com> Thanks Mik.... Actually....I had not tried....was following the step-by-step module Perl.... But you don't set environment vars? PERLLDOPTS:, PERLCCOPTS: and TYPEMAP:.... Marcio 2008/4/23, Mik Cheez <michael_bulk(a)wildgate.com>: > > All I've ever had to do is add the following in 'Makefile': > > include_modules?= perl > > I also take out the 'perl' entry from 'exclude_modules?= ...' > > Will that work, or is that part of what you've been doing? > > Mik > > Marcio wrote: > > > Hi... > > Anybody could me help, to install perl module as is explained in 1.2 > > item of the variables enviroments would have put the output this > > cmds....into " Makefile.deps -> CROSS_COMPILE ?=" <output cmd "perl > > -MExtUtils::Embed -e ldopts">, <"perl -MExtUtils::Embed -e ccopts"> and > > <"echo "`perl -MConfig -e 'print > > $Config{installprivlib}'`/ExtUtils/typemap"> ??? > > or just put the output of these cmds into environments variables : > > PERLLDOPTS:, PERLCCOPTS: and TYPEMAP: and run the Makefile.....??? > > Does anyone know how...? > > Any hints.....?? > > thx... > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > >

3 3

[OpenSER-Users] OpenIMSCore + OpenSER
by Habib, Yeasin 25 Apr '08

25 Apr '08

Hi, Does anybody have a new VMWare Image of OpenIMSCore server where OpenSER is also installed? At present I am using the new VMWare image provided by Dragos Vingarzan. I have tried to install OpenSER on top of that image, but unfortunately I failed. If anybody has a working VMWare Image with OpenSER server then it will be highly appreciable if I get the link for it. Best regards Yeasin Habib

1 0

[OpenSER-Users] OpenSER Docs?
by Douglas Garstang 25 Apr '08

25 Apr '08

So.. I've been using OpenSER for about 2 years now and it's stillmostly a mystery to me. Just about everything I have done with it hasbeen my observation and reverse engineering. Apart from the API specs on openser.org, which are NOT documentation, does any good documentation exist? Are there any books on SER/OpenSER? Where can I find good documentation with examples on the use of avpops? Why append_branch is required, and other mysteries? Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

3 2

[OpenSER-Users] parallel forking
by Allan Chao ( 趙廷睿 ) 25 Apr '08

25 Apr '08

Dear All: I have a question about parallel forking , In generally, UA send a INVITE message to Poxy server(openser), and build a "early dialog" when receives a 180 ring response from proxy, Now, if UA receives two 180 ringing from Proxy Server that uses parallel foring , that means UA has two early dialogs in itself , is it right ? thanks, allan DISCLAIMER: Sample Disclaimer added in a VBScript.

3 4

[OpenSER-Users] Only a single phone will register
by Dave Poirier 25 Apr '08

25 Apr '08

Hello All, I setup my Openser 1.3 and mysql. Authentication seems to work for only the first phone that registers. I did a tcpdump from the server and the rest of the phones get a 401 unauthorized. I can go into mysql and actually see the correct username password in the database. I also double checked it on the phones. Like I said... all three of the phones will register but only if they are the first. Is it something obvious that I'm missing? Anyone else run into this? If I do a "openserctl ul show" (with debug on) I get the following output.... database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: location table=512 records=1 max_slot=1 AOR:: 1024 Contact:: sip:1024@172.17.5.118:5060 Q= Expires:: 2657 Callid:: d35614ae-9db18357-7bd13fe4(a)172.17.5.118 Cseq:: 2 User-agent:: PolycomSoundPointIP-SPIP_550-UA/3.0.0.0258 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:198.236.66.129:5060 Methods:: 8063 FIFO command was: :ul_dump:openser_receiver_22488 Is the "max_sot=1" a concern? If so I am at a loss for where to change it. Thanks for any help, Dave -- David Poirier

3 2

[OpenSER-Users] problems with debian openser initd script
by Robson Souza 24 Apr '08

24 Apr '08

Hello Could you guys please give some help? I've just compiled and installed OpenSER 1.3.1 with tls in a Debian 4.0 R3. The modules mysql and snmpstats are included but not used yet. The openserctl commands is working smoothly in order to start, monitoring and stop. But I still can't start openser on boot. I have coppied the openser.init and openser.default files from source packaging directory to /etc/init.d/openser and /etc/default/openser respectively. Then, after configuring /etc/default/openser file, I configured /etc/init.d/openser as executable. So, when i call "/etc/init.d/openser start", got the output: grep: /etc/openser/openser.cfg: No such file or directory Starting openser: openser already running. But, after type "/etc/init.d/openser status", I got: Status of openser: openser is not running And really it is not working. So, how to get openser starting on boot? Is there another way? --------------------------------- Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!

3 3

[OpenSER-Users] new database module: db_oracle
by Henning Westerholt 24 Apr '08

24 Apr '08

Hi all, thanks to Iouri Kharon a new database access module was included in the trunk: db_oracle - Database access for Oracle database So if you use at the moment the db_unixodbc module via ODBC to access your Oracle database, now is a good time to try this new one if the code and SQL scheme work for you. You find the documentation of the module at the usual place: http://www.openser.org/docs/modules/devel/db_oracle.html Best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[OpenSER-Users] All media traffic gets passed through Mediaproxy?
by Ash Rah 24 Apr '08

24 Apr '08

Hello, Sorry for a novice question and equally sorry if it is already answered elsewhere. I have an OpenSER setup with Mediaproxy in same box. I am including the configuration at the bottom. My question is does ALL media traffic (even the media traffic between non NAT'ed clinets, and media traffic between NAT'ed clients where both-way media traffic works perfectly fine without a media proxy) passes through Mediaproxy dispatcher/server? Or only the media traffic between (troubled) NAT'ed clients are passed through MediaProxy server? My goal is to have the OpenSER to provide registration where the actual calls will be routed through an external Asterisk server. I do not want media traffic passing through Mediaproxy unless I have a NAT related media problems with a certain client. But apparently current set up is forcing all media traffic to pass through the Mediaproxy server. Any advise or suggestion will be greatly appreciated. modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30 modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock" modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) route { # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6) use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

2 3

[Serusers] Memory Problems
by Ricardo Martinez 24 Apr '08

24 Apr '08

Hello. I'm getting weird memory problems with my SER distribution. I'm using a ser 0.9.7 with a Fedora Core release 6. I'm also using the LCR module to do some routing logics. Since a few months i'm getting Memory errors, like these : Apr 5 22:00:50 ser[21996]: ERROR: build_req_buf_from_sip_req: out of memory Apr 5 22:00:50 ser[21996]: ERROR: forward_request: building failed Apr 5 22:00:58 ser[21996]: ERROR: build_req_buf_from_sip_req: out of memory Apr 5 22:00:58 ser[21996]: ERROR: print_uac_request: no pkg_mem Apr 5 22:00:58 ser[21996]: ERROR: t_forward_nonack: failure to add branches Apr 5 22:01:00 ser[22000]: ERROR: do_action: memory allocation failure Apr 5 22:01:00 ser[22000]: ERROR:avpops:pushto_avp: SET_XXXX_T action failed Apr 5 22:01:00 ser[22000]: ERROR: do_action: memory allocation failure Apr 5 22:01:00 ser[22000]: next_gw(): ERROR: do_action failed with return value <-2> Apr 5 22:01:10 ser[21996]: ERROR: do_action: memory allocation failure Apr 5 22:01:10 ser[21996]: next_gw(): ERROR: do_action failed with return value <-2> Apr 5 22:02:04 ser[22000]: ERROR: build_req_buf_from_sip_req: out of memory Apr 5 22:02:04 ser[22000]: ERROR: print_uac_request: no pkg_mem Apr 5 22:02:04 ser[22000]: ERROR: t_forward_nonack: failure to add branches Apr 5 22:02:04 ser[22000]: ERROR: w_t_relay (failure mode): forwarding failed Apr 5 22:07:23 ser[22000]: ERROR: build_res_buf_from_sip_res: out of mem Apr 5 22:07:23 ser[22000]: ERROR: relay_reply: no mem for outbound reply buffer Apr 5 22:07:23 ser[22000]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 1105 It start to fail with a few calls to finally drop all calls with a total colapse of resources. I'm not sure where the problem could be located. Can someone help me here ? Among other things i do several AVP operations with the RURI to make some checks. Thanks Ricardo Martinez.-

3 6

[OpenSER-Users] how change method ??
by Amit Vijayvargiya 24 Apr '08

24 Apr '08

Hi, i want to change request type in openser how i can do this?? i want to change REFER method to INFO i can identified by this is_method("REFER") but how i can change it Thanks & Regards, Amit

2 1

[OpenSER-Users] Install Perl Module
by Marcio 23 Apr '08

23 Apr '08

Hi... Anybody could me help, to install perl module as is explained in 1.2 item of the variables enviroments would have put the output this cmds....into " Makefile.deps -> CROSS_COMPILE ?=" <output cmd "perl -MExtUtils::Embed -e ldopts">, <"perl -MExtUtils::Embed -e ccopts"> and <"echo "`perl -MConfig -e 'print $Config{installprivlib}'`/ExtUtils/typemap"> ??? or just put the output of these cmds into environments variables : PERLLDOPTS:, PERLCCOPTS: and TYPEMAP: and run the Makefile.....??? Does anyone know how...? Any hints.....?? thx...

2 1

[OpenSER-Users] BAsic openSER configuration. INVITES to a UA not looped back.
by Eloi Pecharroman 23 Apr '08

23 Apr '08

Hello, I have been unsuccessfully trying to setup OpenSER as a basic server in my private network. The idea is that clients can register to the virtual domain "sipserver.apt.local" and from then, INVITES will be sent to the appropiate address. I use the following openser.cfg # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 log_facility=LOG_LOCAL7 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 listen=udp:192.168.100.244:5060 listen=udp:sipserver.apt.local:5060 alias="sipserver.apt.local:5060" sip_warning=yes # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/lib/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; xlog("L_INFO","[ACHTUNG] saved $ru"); save("location"); exit; }; xlog("L_INFO","[ACHTUNG]request from $ru"); #lookup("aliases"); #if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); #}; # native SIP destinations are handled using our USRLOC DB #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # exit; #}; #append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } I can register fine, however, on INVITE, openSER forwards it internally instead of fw to the client registered until max_fw reaches 0 and an error "too many hops" is the final response. Log below: Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: [ACHTUNG] saved sip:192.168.100.244 Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: parse_headers: flags=ffffffffffffffff Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: parse_headers: flags=8000000 Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: parse_headers: flags=ffffffffffffffff Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: build_contact(): Created Contact HF: Contact: <sip:oslo_2_Route_1@sipserver.apt.local:5060>;expires=3600^M Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: parse_headers: flags=ffffffffffffffff Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: check_via_address(192.168.100.238, sipserver.apt.local, 0) Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: DEBUG:destroy_avp_list: destroying list (nil) Apr 18 15:36:16 barcelona /usr/sbin/openser[5022]: receive_msg: cleaning up Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: SIP Request: Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: method: <INVITE> Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: uri: <sip:oslo@sipserver.apt.local> Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: version: <SIP/2.0> Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: flags=2 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: Found param type 232, <branch> = <z9hG4bK4211644481>; state=16 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: end of header reached, state=5 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: Via found, flags=2 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: this is the first via Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: After parse_msg... Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: preparing to run routing scripts... Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: flags=100 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:parse_to:end of header reached, state=10 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DBUG:parse_to: display={}, ruri={sip:oslo@sipserver.apt.local} Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: get_hdr_field: <t> [32]; uri=[sip:oslo@sipserver.apt.local] Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: to body [<sip:oslo@sipserver.apt.local>^M ] Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: get_hdr_field: cseq <CSeq>: <985859914> <INVITE> Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: add_param: tag=3326383385 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:parse_to:end of header reached, state=29 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DBUG:parse_to: display={}, ruri={sip:oslo_2_Route_1@sipserver.apt.local} Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: flags=200 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: get_hdr_body : content_length=202 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: found end of header Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: find_first_route: No Route headers found Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: loose_route: There is no Route HF Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if host==us: 19==15 && [sipserver.apt.local] == [192.168.100.244] Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if port 5060 matches port 5060 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if host==us: 19==19 && [sipserver.apt.local] == [sipserver.apt.local] Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if port 5060 matches port 5060 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if host==us: 19==15 && [sipserver.apt.local] == [192.168.100.244] Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if port 5060 matches port 5060 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if host==us: 19==19 && [sipserver.apt.local] == [sipserver.apt.local] Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: grep_sock_info - checking if port 5060 matches port 5060 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: [ACHTUNG]request from sip:oslo@sipserver.apt.local Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: t_newtran: T on entrance=0xffffffff Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: flags=ffffffffffffffff Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: flags=78 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: t_lookup_request: start searching: hash=45820, isACK=0 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: RFC3261 transaction matching failed Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: t_lookup_request: no transaction found Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: parse_headers: flags=ffffffffffffffff Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: check_via_address(192.168.100.238, sipserver.apt.local, 0) Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: WARNING:vqm_resize: resize(0) called Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:tm:_reply_light: reply sent out. buf=0x813efd8: SIP/2.0 1..., shmem=0xb3e5d980: SIP/2.0 1 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:tm:_reply_light: finished Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: mk_proxy: doing DNS lookup... Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:sip_resolvehost2: no port, no proto -> do NAPTR lookup! Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: get_record: lookup(sipserver.apt.local, 35) failed Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:sip_resolvehost2: no valid NAPTR record found for sipserver.apt.local, trying direct SRV lookup... Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: get_record: lookup(_sip._udp.sipserver.apt.local, 33) failed Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:sip_resolvehost2: no valid SRV record found for _sip._udp.sipserver.apt.local, trying A record lookup... Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: check_via_address(192.168.100.238, sipserver.apt.local, 0) Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DBG:check_against_rule_list: using list dns Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:tm:set_timer: relative timeout is 500000 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: add_to_tail_of_timer[4]: 0xb4091bac (104400000) Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:tm:set_timer: relative timeout is 30 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG: add_to_tail_of_timer[0]: 0xb4091bc8 (133) Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:tm:t_relay_to: new transaction fwd'ed Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:tm:UNREF_UNSAFE: after is 0 Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: DEBUG:destroy_avp_list: destroying list (nil) Apr 18 15:36:16 barcelona /usr/sbin/openser[5024]: receive_msg: cleaning up What is necessary so the INVITES are sent to the client and not to the server? Regards,

3 3

[OpenSER-Users] [OpenSER] problem with my presence module
by piff62 piff62 23 Apr '08

23 Apr '08

Hello everybody, I write to you because I have an error with my presence module. I try to integrate a xcap server (openxcap) with my openser. I have configured my openser with module presence, presence_xml and mi_xmlrpc but i have theses error messages : ERROR:presence:mi_refreshWatchers: getting rules doc Apr 23 11:41:54 [7867] ERROR:mi_xmlrpc:default_method: command (refreshWatchers) processing failed. My openser.cfg contains : # -- mi_xmlrpc params -- modparam("mi_xmlrpc", "log_file", "/var/log/openser-xmlrpc.log") modparam("mi_xmlrpc", "port", 8080) # -- presence params -- modparam("presence", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("presence", "server_address", "sip:10.24.247.59.:5060") modparam("presence", "fallback2db", 1) # -- presence_xml params -- modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "integrated_xcap_server", 1) # -- xcap_client params -- modparam("xcap_client", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("xcap_client", "xcap_table", "xcap") modparam("xcap_client", "periodical_query", 1) modparam("xcap_client", "query_period", 50) I can't find why i have error ? Someone can help me or have already have this problem ? Thanks ! Thomas

1 0

[OpenSER-Users] [OpenSER] Problem with openser and openxcap integration
by piff62 piff62 23 Apr '08

23 Apr '08

Hello everybody, I work on a IMS prototype architecture. I have install a openser 1.3.1 and openxcap 0.9.9. I try to use openser with the XCAP server openxcap. I have follow the instruction on the openxcap homepage but I have already many problems. I have the error message : Apr 22 17:25:38 [12220] ERROR:presence:handle_subscribe: Missing or unsupported event header field value Apr 22 17:25:38 [12220] ERROR:presence:handle_subscribe: event= ua-profile Apr 22 17:25:38 [12220] ERROR:tm:t_newtran: transaction already in process 0xb5a073b0 I think i have errors in my config file openser.cfg but i can't found where. I understand that my openser don't support le event ua-profile but i can't found how to support it ? I precise that i have configured my eyebeam with : http://10.24.247.59:8000/xcap-root@mydomain Someone can help me please ? Thanks Thomas

2 2

[OpenSER-Users] Unregistered users able to make calls
by Aadilkhan Maniyar 22 Apr '08

22 Apr '08

Hi, I have a setup where in Openser is acting as the registrar and Asterisk handles the media. i.e. the INVITEs are relayed to the Asterisk server by Openser. When I try to make a call from an unregistered user the call goes thru. The user is present in the subscriber table but is not registered with Openser at that instant. Is this an expected behavior? If not what needs to be done so that an unregistered user is prohibited from making any calls. Any help would be appreciated. Thanks in advance. Regards, Aadil

5 6

[OpenSER-Users] UDP/TCP failover in OpenSER
by Antoine Fressancourt 22 Apr '08

22 Apr '08

Hello list, I have several questions about the support of TCP in OpenSER in general, and about the switch/ failover from UDP to TCP in particular. In my setup, I have several SIP clients registered to a single OpenSER, and they exchange presence messages on a peer-to-peer basis. When the presence document is complex, the size of the message sent exceeds a given limit, and I want my message to be sent over TCP. My clients are registered on the OpenSER, and the REGISTER has been sent over UDP. From now, when client A sends a NOTIFY using TCP to client B, the message is transported over TCP from A to the OpenSER, and is transported over UDP from the OpenSER to B. So here are my questions : - How can I tell OpenSER to relay the message to B using TCP ? - Do I have something particular to do while registering to tell OpenSER that the client is listening in UDP and TCP ? Thank you in advance for your answers, Antoine

5 10

[OpenSER-Users] Openser ignoring SIP version in the Request Line
by Aadilkhan Maniyar 22 Apr '08

22 Apr '08

Hi, When Openser receives a malformed INVITE with SIP version set to 10.0 in the Request Line, it just processes it as a normal INVITE. The Request Line of the INVITE message received looks like this: INVITE sip:ua1@domain.com SIP/10.0 According to RFC 3261, the Requests should contain SIP/2.0. Hence Openser should drop this request with an appropriate 4xx reponse. But this behavior is not seen. Do I have to modify my openser.cfg for validating the above INVITE? Any help would be appreciated. Thanks, Aadil

2 1

[OpenSER-Users] Call setup authentication through openLDAP
by Ahmed Huraimel 22 Apr '08

22 Apr '08

Dear all, I wounder if anyone successfully uses openSER 1.3.x with LDAP server (openLDAP). For me I work fine with Authenticating the uses while REGISTER request. However, I could not configure it to Authenticate call setup. in other words, an non register user can make a call setup. what comes to my mind is to use the same idea of authentication when RIGISTER request was issues with some modefications but when SIP proxy server send "407 Proxy Authentication Required" the client send ACK and stop. ================================================= # account only INVITEs if (is_method("INVITE")) { xlog("L_NOTICE","Processing by INVITE handler ...\n"); route(4); exit; #setflag(1); # do accouting } ..... route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } .......... rout[4] { if(is_present_hf("Authorization")) { # ldap search if (!ldap_search("ldap://sipaccounts/ou=sip,dc=mysip,dc=com?sn,userPassword?one?(cn=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("sn/$avp(s:username)"); ldap_result("userPassword/$avp(s:password)"); if(!pv_proxy_authorize("")) { proxy_challenge(""/*realm*/,"0"/*qop*/); exit; } route(1); } else { proxy_challenge("","1"); exit; } ================================================= do any one know how to authenticate call setup? do you think using RADIUS is better for authentication instead of LDAP authentication? regards, Ahmed ALALI

1 0

[Serusers] rewriting to a dns domain
by Rolf Unger 22 Apr '08

22 Apr '08

Hello list users, i want to use ser as a load balancing proxy for some PSTN gateways that I have in use. My idea is to use the rewritehost() core function and DNS SRV entries to accomplish this. But it seems that ser is doing a A record lookup if I simply replace the host with my domain "mynet.interx": if ( ... ) { rewritehost("mynet.interx"); t_relay(); } Ser is running on the same host (89.49.16.54) as the authorative nameserver for the mynet.interx domain. dig shows that the DNS files are configured in the right way for load balancing, but I just can't find a way to make ser as a dns client work like dig. If I use "_sip._udp.mynet.interx" it will complain about a bad name. Well he is partly right, underscores are not allowed in domain names. And if I use only the domain in the rewrite command it will contact the IP of the nameserver of this domain. Or in other words it will send the messages to itself. Using directly rewritehost("sip1.mynet.interx") doesn't help me, because it wouldn't do load balancing. How can I convince ser (version 0.9.6) to use the correct lookup method for the DNS? Any hints are appreciated ... Thanks, Rolf. ---------------------------------------------------------------- ; <<>> DiG 9.2.4 <<>> -t SRV _sip._udp.mynet.interx ;; QUESTION SECTION: ;_sip._udp.mynet.interx. IN SRV ;; ANSWER SECTION: _sip._udp.mynet.interx. 21600 IN SRV 0 50 5060 sip3.mynet.interx. _sip._udp.mynet.interx. 21600 IN SRV 0 50 5060 sip1.mynet.interx. ;; AUTHORITY SECTION: mynet.interx. 21600 IN NS mynet.interx. ;; ADDITIONAL SECTION: sip1.mynet.interx. 21600 IN A 89.49.16.50 sip3.mynet.interx. 21600 IN A 89.49.16.51 mynet.interx. 21600 IN A 89.49.16.54

3 3

[OpenSER-Users] imc module
by sasirekha jaganathan 22 Apr '08

22 Apr '08

1 0

[OpenSER-Users] Dispatcher Failover Hell.
by Douglas Garstang 22 Apr '08

22 Apr '08

I'm using OpenSER 1.2.2. I can't get the dispatcher to do failover. Here's my module parameters... modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "force_dst", 1) modparam("tm","fr_timer",3) Here's my dispatcher list file: 1 sip:119.42.149.76:5099 1 sip:119.42.149.79:5099 And, my routing logic is below. The first thing that I don't understand is that when I print $avp(i:271) I get "sip:119.42.149.76:5099". Where did the other IP address form the list file go? When I print $avp(i:273), I get "2" which is supposed to be the count of the number of items in the list. Where's the other IP gone? I only see one. The failover routing doesn't work. When OpenSER fails to connect to the first IP, it goes into failure_route(11) like it is supposed to, but it DOES NOT try the other IP address (the one that is missing from $avp(i:271)). I have confirmed this by running ngrep on port 5060. So... question is... why is the dispatcher only loading one entry from the dispatcher list file into the avp? Doug. route[10] { xlog("L_INFO","Inbound Routing"); t_on_failure("11"); ds_select_dst("1","4"); xlog("L_INFO","---> $avp(i:271) $avp(i:272) $avp(i:273)"); forward(); xlog("L_INFO","Trying first route $dd $od $rd $td $br"); t_relay(); } failure_route[11] { xlog("L_INFO","First route failed. Trying second"); if(t_check_status("408") ){ ds_next_dst(); t_on_failure("12"); t_relay(); } else { t_reply("501", "Not Implemented"); } } failure_route[12] { route(13); } route[13] { xlog("L_INFO","CRAP"); return; } ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

Re: [Serusers] how can we setup ser to route invite packet?
by Victor Pascual Ávila 22 Apr '08

22 Apr '08

Hi Jun Yin, On Tue, Apr 22, 2008 at 12:00 PM, <serusers-request(a)lists.iptel.org> wrote: > Message: 1 > Date: Mon, 21 Apr 2008 12:09:36 -0700 > From: "Jun Yin" <hansyin(a)gmail.com> > Subject: [Serusers] how can we setup ser to route invite packet? > To: serusers(a)lists.iptel.org > Message-ID: > <e776e0110804211209u6e8c81dcva09193074ca49ab5(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > My enviroment is like this: phone1---proxy1-----proxy2-----phone2. > phone1 register to proxy1, and phone2 register to proxy2, now I initiate > call from phone1 to phone2, I noticed that phone1 just resolved proxy2's > domain by itself and send invite to proxy2 directly, everything works fine. > My question is : How can I force phone1 to send invite to proxy1 and then > let proxy1 to route that invite packet to proxy2? Actually, I hope all > packet between phone1 and phone2 can go through proxy1 and proxy2, include > RTP. This behaviour was determined by phone's configuration or proxy's > configuration? -Signaling can be forced to go through the proxy, but not the media. If you want non-end-to-end media, use a media proxy. -Have a look at SIP trapezoid [1] [2] and decide what you want to go through the proxy (via, R, RR) and what end-to-end -Check your UAs' (i.e. phone) outbound proxy configuration. [1] RFC 3261, Section 16.12.1.1 [2] http://www.iptel.org/sip_trapezoid My two cents, -- Victor Pascual Ávila

1 0

[OpenSER-Users] failover with openser
by Robert McNaught 22 Apr '08

22 Apr '08

Hi, I have heard that there are two main methods to allow failover with openser - you can use heartbeat, which I believe is where a secondary machine assumes the primary machines IP upon the primary not responding to a ping. I understand the alternative is SRV records, which use DNS to failover when a machine goes down. What is the advantages/disadvantages of each? What are people actually using in production with OpenSER? Can anyone offer some advice on this? TIA Robert

4 4

[OpenSER-Users] cnt_avp () doesn't work?
by Miroslav HOSTINSKY 22 Apr '08

22 Apr '08

Hello, I have little problem with openser 1.3.1 and dispatcher (again). according to dispatcher manual: -- 1.3.11. cnt_avp (str) The name of the avp storing the number of destination addresses kept in dst_avp avps. -- So if I understand it correctly, in AVP (defined in cnt_avp) is stored number/quantity of active destinations in current destination group. but I have always initial number of destinations in this AVP (as defined in dispatcher "destination list" config file). If I remove destination from balancing list (failed GW), in this AVP is always same initial value. I think it should change (in this case decrement by 1). Is this correct? Thanks :) -- Miroslav Hostinsky

2 2

[OpenSER-Users] Avp_db_query with timestamp??
by Geir O. Jensen 22 Apr '08

22 Apr '08

I'm trying to get a timestamp from the sql server with the following: avp_delete("$(avp(s:rds_time))"); avp_db_query("SELECT time FROM usr_status WHERE cid='$ci';","$(avp(s:rds_time))"); HOWEVER, even if the time field is "2008-04-21 15:06:33" the avp is only filled with "2008". Any idea what's going on ?? - gojensen UNINETT

2 2

[OpenSER-Users] [Wesip] Ringtone application
by kriko 22 Apr '08

22 Apr '08

Hello! Have anyone tried the ringtone app on wesip page? I rewrote it, however mine crashes, but so does wesip one. I'm suspecting a bug somewhere else that those two application, but I'm having trouble debugging, since it doesn't show exactly where is the problem: ---- SipProcessor [SipProcessor[4]] - error processing event... java.lang.NullPointerException at com.voztele.sipservlet.deploy.rules.StandardSipServletMapping.evaluate(StandardSipServletMapping.java:39) at com.voztele.sipservlet.core.ConvergedHost.map(ConvergedHost.java:104) at com.voztele.sipservlet.core.SipHostMapper.map(SipHostMapper.java:123) at org.apache.catalina.core.ContainerBase.map(ContainerBase.java:967) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeSipRequest(ConvergedHostValve.java:267) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeSIP(ConvergedHostValve.java:126) at com.voztele.sipservlet.valves.ConvergedHostValve.invoke(ConvergedHostValve.java:104) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.valves.ErrorDispatcherValve.invoke(ErrorDispatcherValve.java:124) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:118) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedEngineValve.invoke(ConvergedEngineValve.java:114) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.processor.SipProcessor.invokeContainer(SipProcessor.java:198) at com.voztele.sipservlet.processor.SipProcessor.processRequest(SipProcessor.java:142) at com.voztele.sipservlet.processor.SipProcessor.process(SipProcessor.java:272) at com.voztele.sipservlet.processor.SipProcessor.run(SipProcessor.java:253) at java.lang.Thread.run(Thread.java:619) ---- How to discover what is blocking it from working? -- (\__/) (='.'=) (")_(")

1 0

[OpenSER-Users] Openser Mediaproxy
by Abdul 21 Apr '08

21 Apr '08

Hi friends, I was trying to install Openser and Mediaproxy. but when i am starting media proxy using /etc/init.d/mediaproxy star I am getting some error in syslog. Jan 22 12:18:27 hatifserver4 mediaproxy[14707]: warning: accounting is enabled but the accounting module is missing. accounting is not available! Jan 22 12:18:27 hatifserver4 mediaproxy[14707]: Listening for commands on local socket `/var/run/mediaproxy.sock' Jan 22 12:18:27 hatifserver4 mediaproxy[14707]: Listening for remote commands on `212.XXXXXXXXXXXXXx:25060' Jan 22 12:18:27 hatifserver4 mediaproxy[14707]: Remote commands are allowed from: 212.XXXXXXXXXXXXXXXXXX, 127.0.0.1 Jan 22 12:18:27 hatifserver4 mediaproxy[14707]: Using IP address `XXXXXXXXXXXXX' for the RTP/RTCP proxy Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: Traceback (most recent call last): Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: File "/usr/local/mediaproxy/proxydispatcher.py", line 65, in <module> Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: import dispatcher Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 39, in <module> Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: from accounting import accounting, StopRecordSerializer, UnserializeError Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: File "/usr/local/mediaproxy/modules/accounting.py", line 17, in <module> Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: from radius import * Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: File "/usr/local/mediaproxy/modules/radius.py", line 10, in <module> Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: import pyrad.client Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: ImportError Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: : Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: No module named pyrad.client Jan 22 12:18:27 hatifserver4 proxydispatcher[14710]: exiting. this is our Mediaproxy.ini file i am not using any kind of accounting or radius in INI file but i don't kow why it is shwoing accounting is enable [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock group = openser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = openser listen = XXXXXXXXXXXXXx allow = XXXXXXXXXXXXXXXXXXXXXXXXxx proxyIP = XXXXXXXXXXXXXXXXXXXXXX portRange = 20000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 accounting = none [Accounting] accounting = none --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

3 2

[OpenSER-Users] removing openser
by Robert McNaught 21 Apr '08

21 Apr '08

Hi, I wish to re-compile OpenSER 1.2.3 - I previously had openser 1.1 on the machine and then put on openser.1.2.3 - without a make uninstall command, how is it possible to recompile. Using the command 'openser -V' gives me the correct version, however, when I try and recompile it does nothing and says openser is up to date. I tried removing /usr/local/lib/openser to see if that would make it re-compile. Does anyone know how to get it to re-compile and start from scratch when it is already on the machine? Thanks Robert McNaught

3 3

[OpenSER-Users] kagoor voiceflow replacement
by Robert McNaught 21 Apr '08

21 Apr '08

Hi, Has anyone replaced the functionality of a kagoor voiceflow session border controller with openser? and is it possible? TIA Robert

6 16

[Serusers] how can we setup ser to route invite packet?
by Jun Yin 21 Apr '08

21 Apr '08

Hi, My enviroment is like this: phone1---proxy1-----proxy2-----phone2. phone1 register to proxy1, and phone2 register to proxy2, now I initiate call from phone1 to phone2, I noticed that phone1 just resolved proxy2's domain by itself and send invite to proxy2 directly, everything works fine. My question is : How can I force phone1 to send invite to proxy1 and then let proxy1 to route that invite packet to proxy2? Actually, I hope all packet between phone1 and phone2 can go through proxy1 and proxy2, include RTP. This behaviour was determined by phone's configuration or proxy's configuration? I'm using ser 0.9.6 works as proxy and linphonec works as phone. I also tried to use asterisk works as proxy but got same result. Thanks in advance. -- Rgds, Hans Yin Email: hansyin(a)gmail.com MSN: hansyin(a)hotmail.com Skype: hans_yin_vancouver

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[Serusers] How Many Nines?
by Jeremy A 20 Apr '08

20 Apr '08

Hello, I'm looking at a project which requires a higher than normal reliability for a SIP server. The project involves risk-of-life communications. The volume of traffic is quite low. I have been using SER in embedded applications for some time now without problems but I am now looking at a larger scale project. Assuming I can provide hardware server(s) with suitable characteristics, then : - How reliable is SER? - In each version? - In each module? Are there any well established high availability configurations for SER? Are there any risk-of-life deployments I can reference? We may well use SEMS for part of the project so the same questions apply. I'm not expecting 100% reliability, just a system that matches or exceeds present high reliability PABX deployments. If you are of the honest opinion that SER is not good enough, what SIP server products should I be looking at? Or is SIP not suitable full stop? Thanks in advance Jeremy

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[OpenSER-Users] Error when using avp_db_load: "avpops:dbrow2avp: wrong field types in dbrow"
by Noel R. Morais 19 Apr '08

19 Apr '08

Hi all, I'm trying to load some attributes from DB to an avp and check if the value is greater than 0 (zero). I'm trying to limit the max sessions per user@domain. But when I execute "avp_db_load(..)" I get the error: "avpops:dbrow2avp: wrong field types in dbrow". I'm using unixodbc with oracle as DBRMs. The table DDL is: UUID(VARCHAR2), USERNAME(VARCHAR2), DOMAIN(VARCHAR2), ATTRIBUTE(VARCHAR2), VALUE(VARCHAR2), TYPE(NUMBER) Follow bellow the source: if (method == "REGISTER") { if ( avp_db_load("$fu","$avp(s:maxsessions)/avp_user_attributes") && avp_check("$avp(s:maxsessions)", "gt/0/g") ) { save("location") ; return; } else { sl_send_reply("503", "Service Unavailable"); return; } } I've checked the AVP source and the field types seems to be ok. I don't what to do. Thanks in advance, Noel ps: Sorry about the poor English

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[OpenSER-Users] Radius with additional parameters
by Alex Massover 19 Apr '08

19 Apr '08

Hello! I see how I can receive additional parameters from Radius server via AVPs. But I wonder if it's possible to send additional parameters (headers) to Radius server in requests? For example I want to pass to Radius server P-Asserted header, in case that I need validate it? -- Best Regards, Alex Massover VoIP Development Manager Jajah Inc.

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[OpenSER-Users] kagoor voiceflow replacement
by Robert McNaught 18 Apr '08

18 Apr '08

Klaus, Broadworks is from Broadsoft and is a service provider platform for hosted VoIP and uses SIP http://www.broadsoft.com Robert

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[OpenSER-Users] mi_datagram module bind failure
by Sigrid Thijs 18 Apr '08

18 Apr '08

Hi, we're setting up OpenSER 1.3.0 on a FreeBSD system, with the mi_datagram module configured, and when we start up openser, we get the following error: /usr/local/sbin/openser[3139]: NOTICE presence:child_init: init_child [1] pid [3139] /usr/local/sbin/openser[3138]: ERROR:mi_datagram:mi_init_datagram_server: bind: Invalid argument /usr/local/sbin/openser[3142]: NOTICE presence:child_init: init_child [4] pid [3142] /usr/local/sbin/openser[3138]: CRITICAL:mi_datagram:pre_datagram_process: function mi_init_datagram_server returned with error!!! /usr/local/sbin/openser[3138]: ERROR:core:start_module_procs: pre-fork function failed for process "MI Datagram" in module mi_datagram In the openser configuration: modparam ("mi_datagram", "socket_name", "udp:[ip.add.ress.here]:5068") We checked that the port is not in use. Note that this is working fine when running the same openser version on a linux based system. We are also not having this issue when we specify a UNIX socket. Do you have any idea what could be wrong? kind regards, Sigrid

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[OpenSER-Users] Openser xmpp module configuration problem
by Gaoqiang Qian 18 Apr '08

18 Apr '08

Hi, I have now running a openser server 1.1.0 notls and have installed another openser 1.3.1 with xmpp module configured as the server-mode. Could you tell me how could I change my main sip server(openser 1.1.0 notls) configuation to configure it send sip message to the xmpp gw. Or is there anyway that I could just have one openser server(1.2.x above) along with xmpp module installed. And have it configured as SIP server + XMPP server + SIP-to-XMPP gateway? Could you show me some configuration example to achieve this if possible? Thanks in advance. BR, -- Gaoqiang Qian Automation System and Technology Helsinki University Of Technology GSM: +358415058138

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[OpenSER-Users] PUA/PUA_MI: More than one Publish sent
by Schumann Sebastian 17 Apr '08

17 Apr '08

Dear all I am using OpenSER 1.3.1 branch release R3936. I face some bad behaviour using pua and pua_mi modules. I publish presence states via the management interface. Sometimes, it happens that multiple PUBLISH message are sent by one request. Unfortunately, there is no "system" or oder when this occurs. It just happens after some publications. In case of an expiration (Expires=0, E-Tag .) it is not that bad, I receive the log message: ERROR:presence:update_presentity: No E_Tag match. In case of publication (Expires=-1, E-Tag .) it is very bad: After publication, he receives also two similar messages, and threats both PUBLISH messages as two different ones. Thus, creates two different presentity entries with different E-Tags. If the state is expired now via MI (Expires: -1, E-Tag .) only one of the E-Tags is removed. The second stays in the presentity table and the user still appears online. Any ideas what can be the reason and how to solve it? Best regards Sebastian

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[OpenSER-Users] Can't start usrloc using mysql connection (mysql:db_mysql_new_connection: driver error: Can't connect to MySQL server on ...)
by Pascal Maugeri 17 Apr '08

17 Apr '08

Hi I have the following error in connection with MySQL when I enable usrloc to use the DB (db_mode=1): [...] Apr 17 14:05:48 [32444] DBG:mysql:db_mysql_init: connection 'mysql:// openser:openserrw@192.168.1.4/openser' not found in pool Apr 17 14:05:48 [32444] DBG:mysql:db_mysql_new_connection: opening connection: mysql://xxxx:xxxx@192.168.1.4/openser Apr 17 14:05:48 [32444] ERROR:mysql:db_mysql_new_connection: driver error: Can't connect to MySQL server on '192.168.1.4' (111) Apr 17 14:05:48 [32444] ERROR:usrloc:register_udomain: failed to open database connection Apr 17 14:05:48 [32444] ERROR:registrar:domain_fixup: failed to register domain Apr 17 14:05:48 [32444] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 323 [...] My configuration file is: loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" [...] # ----- usrloc params ----- modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.1.4/openser ") I see that mysql server is started and I've managed at installation to create the openser DB (using openserdbctl create). Could you help me doing the troubleshooting of this ? What else should I verify ? Regards, Pascal

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Re: [OpenSER-Users] Can't start usrloc using mysql connection (mysql:db_mysql_new_connection: driver error: Can't connect to MySQL server on ...)
by Bogdan-Andrei Iancu 17 Apr '08

17 Apr '08

There are parameters that should point to the same DB as both openser and openserctl should operate on the same DB. Regards, Bogdan Pascal Maugeri wrote: > You're right Bogdan, I changed my configuration to connect to localhost: > > modparam("usrloc", "db_url", > "mysql://openser:openserrw@localhost/openser") > > Thanks for your prompt reply ! > > Now it works very well. > > just for curiosity, is it important that the DBHOST in the file > openserctl file match the host used in openser.cfg for db_url ? Are > these two parameters independent ? > > Regards, > Pascal > > > > > > > > > On Thu, Apr 17, 2008 at 2:32 PM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hi Pascal, > > try to connect with the mysql client (mysql) from the openser > machine to the mysql server machine and see if it works. > > Maybe your mysql server is not configured to listen on 192.168.1.4 > <http://192.168.1.4>, but only on 127.0.0.1 <http://127.0.0.1> > (for local connections). > > Regards, > Bogdan > > Pascal Maugeri wrote: > > Hi > I have the following error in connection with MySQL when I > enable usrloc to use the DB (db_mode=1): > > [...] > Apr 17 14:05:48 [32444] DBG:mysql:db_mysql_init: connection > 'mysql://openser:openserrw@192.168.1.4/openser > <http://openser:openserrw@192.168.1.4/openser> > <http://openser:openserrw@192.168.1.4/openser>' not found in pool > Apr 17 14:05:48 [32444] DBG:mysql:db_mysql_new_connection: > opening connection: mysql://xxxx:xxxx@192.168.1.4/openser > <http://xxxx:xxxx@192.168.1.4/openser> > <http://xxxx:xxxx@192.168.1.4/openser> > Apr 17 14:05:48 [32444] ERROR:mysql:db_mysql_new_connection: > driver error: Can't connect to MySQL server on '192.168.1.4 > <http://192.168.1.4> <http://192.168.1.4>' (111) > > Apr 17 14:05:48 [32444] ERROR:usrloc:register_udomain: failed > to open database connection > Apr 17 14:05:48 [32444] ERROR:registrar:domain_fixup: failed > to register domain > Apr 17 14:05:48 [32444] ERROR:core:fix_actions: fixing failed > (code=-1) at cfg line 323 > [...] > > My configuration file is: > > loadmodule "mysql.so" > loadmodule "sl.so" > loadmodule "tm.so" > loadmodule "rr.so" > loadmodule "maxfwd.so" > loadmodule "usrloc.so" > [...] > > # ----- usrloc params ----- > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "db_url", > "mysql://openser:openserrw@192.168.1.4/openser > <http://openser:openserrw@192.168.1.4/openser> > <http://openser:openserrw@192.168.1.4/openser>") > > > I see that mysql server is started and I've managed at > installation to create the openser DB (using openserdbctl create). > > Could you help me doing the troubleshooting of this ? What > else should I verify ? > > Regards, > Pascal > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

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[OpenSER-Users] global variables
by Hugo Koblmueller 17 Apr '08

17 Apr '08

high.*! just wondering if there is any possibility to use global variables in openser.cfg? like e.g. for configuration values, constants, .. another question: how can i access the configuration paramerters like "debug", "port", etc. thx & cheers -hugo

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[OpenSER-Users] carrierroute seg fault
by Helmut Kuper 17 Apr '08

17 Apr '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have "seg-fault" with carrierroute module: openser: svnrevision: 2:4013M on a 64 bit machine running CENTOS5 - -------------------------------------------------- openser-config: [..] loadmodule "carrierroute.so" modparam("carrierroute", "config_file", "/opt/app/voip/openser/etc/openser/carrierroute.conf2") [..] if(!cr_route("default", "0", "$rU", "$rU", "call_id")) { ... } - ----------------------------------------------------- carrierroute.conf2: domain 0 { ~ prefix +49441 ~ { ~ max_targets = 1 ~ target y.y.z ~ { ~ prob = 0.500000 ~ hash_index = 1 ~ status = 1 ~ comment = "GK-Proxy" ~ } ~ } } - ---------------------------------------------------- openser console: [...] Apr 17 10:54:50 [27937] INFO:carrierroute:add_carrier_tree: carrier default not found, add it Apr 17 10:54:50 [27937] INFO:carrierroute:add_tree: tree default has internal id 0 Apr 17 10:54:50 [27937] INFO:carrierroute:add_carrier_tree: created carrier tree: default, with id 1 and 1 trees Apr 17 10:54:50 [27937] INFO:carrierroute:load_config: loading domain 0 Apr 17 10:54:50 [27937] INFO:carrierroute:load_config: loading prefix 49441 Apr 17 10:54:50 [27937] INFO:carrierroute:load_config: loading target y.y.z Apr 17 10:54:50 [27937] INFO:carrierroute:load_config: adding route for prefix 49441, to host y.y.z, prob 0.500000, backed up: 0, backup: -1 Apr 17 10:54:50 [27937] INFO:carrierroute:add_route: adding prefix 49441, prob 0.500000 Apr 17 10:54:50 [27937] INFO:carrierroute:get_route_tree: domain 0 not found, add it Apr 17 10:54:50 [27937] INFO:carrierroute:add_domain: domain 0 has id 0 Apr 17 10:54:50 [27937] INFO:carrierroute:add_route_tree: tree default has 1 trees Apr 17 10:54:50 [27937] INFO:carrierroute:get_route_tree: created route tree: 0, with id 0 Apr 17 10:54:50 [27937] INFO:carrierroute:add_route: found route, now adding Apr 17 10:54:50 [27937] INFO:carrierroute:add_route_rule: no backed up rules Apr 17 10:54:50 [27937] INFO:carrierroute:rule_fixup: fixing tree 0 Apr 17 10:54:50 [27937] INFO:carrierroute:rule_fixup_recursor: rule with host y.y.z hash has hashindex 1. Apr 17 10:54:50 [27937] INFO:carrierroute:mod_init: module initialized, pid [27937] Apr 17 10:54:50 [27937] ERROR:core:pv_parse_spec: bad parameters Apr 17 10:54:50 [27937] INFO:carrierroute:carrier_fixup: carrier tree default has id 1 Apr 17 10:54:50 [27937] ERROR:core:pv_parse_spec: bad parameters [...] - --------------------------------------------- The seg-fault: Apr 17 10:57:20 [28437] DBG:carrierroute:get_route_tree_by_id: searching in carrier default, id 1 Apr 17 10:57:20 [28437] DBG:carrierroute:get_route_tree_by_id: tree default, domain 0 : 0 Segmentation fault (core dumped) Number in $rU was "+49441234" Any ideas? Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgHFfIACgkQ4tZeNddg3dx90wCfTms/1qSzzJtmGLUQUdUt81hH KCIAni+f23WQBeJTt5BRMVuJ1YtzEwKM =CCVU -----END PGP SIGNATURE-----

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[OpenSER-Users] dispatcher and auto reload of previously failed GWs
by Miroslav HOSTINSKY 17 Apr '08

17 Apr '08

Hello all, I have problem with dispatcher module in openser 1.3.1 I am not able to correctly setup scenario when failed GWs will be automatically reloaded back by openser, when they will be available. My openser.cfg: -- mpath = "/usr/lib/openser/modules" debug = 3 fork = no log_stderror = yes children = 2 check_via = no dns = off rev_dns = off port = 5060 loadmodule "maxfwd.so" loadmodule "sl.so" loadmodule "tm.so" modparam("tm", "fr_timer", 5) loadmodule "xlog.so" loadmodule "avpops.so" loadmodule "textops.so" loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_ping_from", "sip:proxy@192.168.245.201") modparam("dispatcher", "ds_probing_threshhold", 3) ######################################################################## # Request route 'main' ######################################################################## route { if ( method=="INVITE" || method=="ACK" || method=="BYE" || method=="OPTIONS" ) { ds_select_dst("1","0"); t_on_failure("1"); t_relay(); exit; } } failure_route[1] { if (t_check_status("408")) { xlog("L_INFO","Marking GW as failed...\n"); ds_mark_dst(); ds_next_dst(); t_on_failure("1"); t_relay(); } else{ t_reply("501","Not Implemented"); } } -- According to dispatcher readme file, this config should be sufficient (I think :)), but there is no OPTIONS generated by openser to the failed gateways. Removing of failed GWs works perfectly for me... Could someone help what is wrong with my config? Thanks.

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[OpenSER-Users] Openser can not load LDAP module
by Ahmed Huraimel 17 Apr '08

17 Apr '08

Hello, I am running Openser 1.3.X with MySQL authentiocation and i decided to LDAP instead of MySQL. I did the following: - Reconfigated the Makefile and i took ldap from openser Makefile in (exclude_modules?=) in order to include LDAP module ls /usr/local/lib/openser/modules/ acc.so benchmark.so diversion.so gflags.so mangler.so mysql.so pike.so speeddial.so uac.so alias_db.so cfgutils.so domainpolicy.so group.so maxfwd.so nathelper.so registrar.so sst.so uri_db.so auth_db.so db_berkeley.so domain.so h350.so mediaproxy.so options.so rr.so statistics.so uri.so auth_diameter.so dbtext.so enum.so imc.so mi_datagram.so path.so siptrace.so textops.so usrloc.so auth.so dialog.so exec.so lcr.so mi_fifo.so pdt.so sl.so tm.so xlog.so avpops.so dispatcher.so flatstore.so ldap.so msilo.so permissions.so sms.so uac_redirect.so - then i added the module in openser.cfg (loadmodule "ldap.so") - When i compile the configuration file i get the following: ERROR:core:sr_load_module: could not open module </usr/local/lib/openser/modules/ldap.so>: /usr/local/lib/openser/modules/ldap.so: undefined symbol: ber_pvt_opt_on Apr 16 17:26:45 [8172] CRITICAL:core:yyerror: parse error in config file, line 83, column 13-14: failed to load module i have no idea from where to start or what to check. can anybody help me? Ahmed ALALI

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[OpenSER-Users] NAT SIP OPTIONS ping
by Amit Nagpal 17 Apr '08

17 Apr '08

Hi, I am trying to configure OpenSER such that it pings its NATed clients with SIP OPTIONS packets instead of the default dummy UDP pings. But, no matter what I do, OpenSER always sends the usual UDP pings. My openser.cfg contains the following lines - sipping_bflag sipping_method sipping_from

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[Serusers] Stability 2.1
by Martin Hoffmann 16 Apr '08

16 Apr '08

Hi, we have a bit of an issue with TCP handling in SER 2.0. It appears that dead connections cause the executing TCP process to hang for a short while until it figures things out. Too many such requests and processing essentially stops. Now, I understand all of this has been fixed in SER 2.1 and all our problems will go away. Question is: Will we get new ones by upgrading. Or asked differently: Is there any experience with SER 2.1 and its statbility in a real world scenario? The boxes in question are edge proxies, so there is not much processing to be done other than basic forwarding and a bit of NAT handling. Can we risk it? Best regards, Martin

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Re: [OpenSER-Users] Failure route after parallel forking (was "Problem with redirection (302 Moved temporarly)")
by Antonio Reale 16 Apr '08

16 Apr '08

Sorry for resuming the post.. I can't understand how to handle failure messages after parallel forking. I need a particular configuration or OpenSER must handle them the same way of a single message? I'm using OpenSER 1.2.3-tls and also using debug mode I can't see nothing special. Thanks. Regards, Antonio. 2008/4/15, Antonio Reale <ant.reale(a)gmail.com>: > > Hi all, > I have a Linksys SPA2102 with 2 phones configured with the same > account/number. If I call that number both phones ring and all is OK. > The problem is that if I set a call forwarding on one of the two line on > the SPA only one phone rings. > Openser receives the 302 message from the SPA but the failure route that > contains t_check_status("302") is not called. > The forwarding works if only one line is connected. > How can I handle the 302 in this situation? > Thanks in advance. > > Regards, > Antonio. >

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[OpenSER-Users] scope of openser flags
by Hugo Koblmueller 16 Apr '08

16 Apr '08

high.*! i'm using openser v1.2 and am a bit confused about the scope of the flag. just for information about my setup: i use openser as gateway between OCSmediation server and asterisk. i learned that the message flags are transaction persistent, which means that if i set some flags while processing the INVITE request, these flags must be visible also when we process all other requests and replies in scope of the same transaction. and i also learned, that the flags which are set for INVITE are visible only when processing 100 (Trying), 180 (Ringing), and 487 (Request Cancelled). however, the flags are not visible when processing CANCEL, 200 (OK), and ACK. thats for the theory. now i have two problems: first (as i mentioned above) i experienced, that 'my' flags are not that visible as they should be. means e.g. i set flag 3 in request route and print it out in onreply_route, where it seems not to be set. see my code below: route { ... if (src_ip == 172.xx.xx.xx) { # relay request to * if (!t_relay("udp:172.xx.xx.xx:5060")) { sl_reply_error(); } } else { # relay request to OCS if (!t_relay("tcp:172.xx.xx.xx:5060")) { sl_reply_error(); } } if (method == INVITE) setsflag(3); xlog("L_INFO","$Cyb$rm: mf=$mf, bf=$bf, sf=$sf$Cxx\n"); exit; } onreply_route { xlog("L_INFO","$Cyr$rs($rr): mf=$mf, bf=$bf, sf=$sf$Cxx\n"); } the output is the following: 10(10569) INVITE: mf=0, bf=0, sf=8 7(10566) 100(Trying): mf=0, bf=0, sf=0 5(10564) 180(Ringing): mf=0, bf=0, sf=0 8(10567) 200(OK): mf=0, bf=0, sf=0 10(10569) ACK: mf=0, bf=0, sf=0 6(10565) BYE: mf=0, bf=0, sf=0 11(10570) 200(OK): mf=0, bf=0, sf=0 as you can see the flag 3 is only set, while processing the INVITE but after this is stays reset. any ideas which problem i'm facing here? my second problem is, that i need to identify the reply "200 OK" which is answered to the relayed INVITE message. as i mentioned above, this will not be covered by the flags scope anyways. Sad <http://www.voipuser.org/modules/PNphpBB2/images/smiles/icon_sad.gif> is there any possibility to have kind of (real) global variables in openser.cfg? thx & cheers -hugo

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[OpenSER-Users] SIP SIMPLE - XMPP interoperability issue
by Gaoqiang Qian 16 Apr '08

16 Apr '08

Hello everyone, I have setup the openser server with the very basic default configuration. Now I have to configure my openser server to connect with a xmpp envoironment on another host with a purpose of piloting the SIP SIMPLE - XMPP IOP. So does anyone here have successful experience of configure openser to connect with XMPP on different hosts? Or could someone provide me some instruction how to configure it. Thanks a lot in advance -- Gaoqiang Qian Automation System and Technology Helsinki University Of Technology GSM: +358415058138

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[OpenSER-Users] Server crash when permanent User Location is present in DB ?!
by Pascal Maugeri 16 Apr '08

16 Apr '08

Hi I've got a weird problem I do not manage to solve alone. I'm using v1.3.0 with mysql.so and and usrloc.so, on redhat enterprise v4. When there is a permanent user location in the database (added previously with openserctl ul add), the server does not start: - there is no log about any error, - after starting the server, there is no running openser process Is it a server bug or should I activate/configure something else ? Below my signature is my configuration file. Thanks in advance for your help Pascal # # $Id: openser.cfg 3542 2008-01-16 15:00:34Z miconda $ # # OpenSER basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=9 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=9 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #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" /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 listen=192.168.1.4:5060 port=5060 # Uncomment this to prevent the blacklisting of temporary not available destinations disable_dns_blacklist=yes alias="pascal.net" ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ #loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" #loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" #loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ #modparam("uri_db", "use_uri_table", 0) #modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- modparam("usrloc", "db_mode", 0) #modparam("usrloc", "db_mode", 1) #modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.1.4/openser ") #modparam("usrloc", "use_domain", 1) # ----- auth params ----- modparam("auth", "secret", "secretphrase") # ----- auth_db params ----- #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "user_column", "username") #modparam("auth_db", "password_column", "password") #modparam("auth_db", "domain_column", "domain") #modparam("auth_db", "db_url", "mysql:// openser:openserrw@192.168.1.4/openser") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authenticate if from local subscriber (uncomment to enable auth) ##if (!(method=="REGISTER") && from_uri==myself) ##{ ## if (!proxy_authorize("", "subscriber")) { ## proxy_challenge("", "0"); ## exit; ## } ## if (!check_from()) { ## sl_send_reply("403","Forbidden auth ID"); ## exit; ## } ## ## consume_credentials(); ## # caller authenticated ##} # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain /* uncomment this if you want to enable presence server and comment the next 'if' block NOTE: uncomment also the definition of route[2] from below */ ##if( is_method("PUBLISH|SUBSCRIBE")) ## route(2); if (is_method("PUBLISH")) { sl_send_reply("503", "Service Unavailable"); exit; } if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) ##if (!www_authorize("", "subscriber")) ##{ ## www_challenge("", "0"); ## exit; ##} ## ##if (!check_to()) ##{ ## sl_send_reply("403","Forbidden auth ID"); ## exit; ##} if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } # Presence route /* uncomment the whole following route for enabling presence NOTE: do not forget to enable the call of this route from the main route */ ##route[2] ##{ ## if (!t_newtran()) ## { ## sl_reply_error(); ## exit; ## }; ## ## if(is_method("PUBLISH")) ## { ## handle_publish(); ## t_release(); ## } ## else ## if( is_method("SUBSCRIBE")) ## { ## handle_subscribe(); ## t_release(); ## } ## ## exit; ##} branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

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[OpenSER-Users] OpenSer closes TCP connection if receives wrong data
by Iñaki Baz Castillo 16 Apr '08

16 Apr '08

Hi, if OpenSer receives a request via TCP with wrong data (for example Via missing) it closed automatically the connection. I'm just experimenting with SIP TCP and I'd like to know the reason for this behaviour, I know that many SIP TCP UAS's have not this behaviour. I also know that other UAS's dissallow empty lines before the message (so disracrd the message) while OpenSer do allow them. Any explanation/recomendation is welcome. Are those recommendation for a SIP TCP stack written in some RFC? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] Openser problem
by Bogdan-Andrei Iancu 16 Apr '08

16 Apr '08

Hi Norberto, most sure the problem is in routing the CANCEL request. On this topic, please see: http://www.voice-sistem.ro/downloads/2007.08.29-Admin-Course/ There is detailed explanation about CANCEL routing. Regards, Bogdan Norberto Monteiro wrote: > > Hello, > > I’m having a problem that is the following: > > I’m using rewriteport to send traffic to another proxy. Then When I > make a call and if I hung up the other phone still rings… > > What could be the problem? > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] Openser and Portabilling100
by Jaas 16 Apr '08

16 Apr '08

Hi all, I would like to know if there could be same problems with OpenSER and PortaBilling100 because I have to choose a billing system for the ISP I work. I don't see any document with interaction between these two systems... Alternatively, can you give me hints about a good and complete billing system? Thank you in advance

2 1

[OpenSER-Users] FreeSWITCH and OpenSER
by Mikael A. Bjerkeland 16 Apr '08

16 Apr '08

Hi, I am trying to look up information on the use of FreeSWITCH together with OpenSER. I am doing a configuration on my own without much experience on the configuration of OpenSER. In my setup OpensSER acts as a registrar and proxy. I have multiple FreeSWITCH servers for media, routing and PSTN termination, but I am not sure if my approach is the "right way". All SIP traffic goes to the OpenSER proxy which in turn routes the call with the LCR module to a FreeSWITCH server. This has proven to work fine in my limited testing environment. I am using ENUM on the FreeSWITCH server to look up dialed numbers. If the dialed number was found in our ENUM registry and belongs to the proxy, we call back to OpenSER: SIP UA (subscriber 56195122) -> INVITE 25929511 -> FreeSWITCH -> ENUM lookup -> Found -> Routed back to OpenSER -> SIP UA (subscriber (25929511) If the call is not found: SIP UA (subscriber 56195122) -> INVITE 95018615 -> FreeSWITCH -> ENUM lookup -> Not found -> Routed to PSTN gateway or SIP peer I am not using aliases in OpenSER as it seems that ENUM provides the same functionality, but please correct me if I'm wrong. I also made this decision because it seems easier to implement features like call forwarding (unconditional, busy, no answer and timed) in FreeSWITCH rather than in OpenSER. This brings me back to my current issue: Call transfer Outbound and inbound calling with ENUM lookups in FreeSWITCH to my test phones works fine. So does calls to the PSTN gateway. My problem has to do with transfers. I can do a an attended transfer from my Polycom phone when I'm in a call. The second call is set up and I can talk to the person I want to transfer to. I hit the transfer button. Nothing happens, and FS and OpenSER try to look up number_to_transfer_to@openser-registrar. The call is routed to OpenSER which complains that it cannot find the subscriber/alias, which makes sense as there isn't configured an alias. But this also happens when I transfer to a PSTN number. Do I need to take special actions on REFERs? Are there any examples of a working OpenSER + FreeSWITCH setup out there? I believe FreeSWITCH and OpenSER complement each other great but I found the lack of information on the two applications used together a showstopper. Has anyone implemented phone service based on OpenSER and FreeSWITCH who can explain their setup a bit? Regards, Mikael

2 3

[OpenSER-Users] Question about installing openser
by Pete Kay 15 Apr '08

15 Apr '08

Hi all, I am trying to get Openser 1.3.1 installed on Debian. It gave me a lot of errors when I try to "make" from the source. Could someone please tell me the list of packages I should install before building Openser? By the way, I haven't been able to find a step-by-step guides on installing Openser from scratch on Debian. If there is one exists somewhere on the Internet, I would greatly appreciate it if you can point me to it. I am currently working through a few of the how-tos trying to put pieces together. Thanks in advance for your help. Regards, Pete

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[OpenSER-Users] Problem with redirection (302 Moved temporarly)
by Antonio Reale 15 Apr '08

15 Apr '08

Hi all, I have a Linksys SPA2102 with 2 phones configured with the same account/number. If I call that number both phones ring and all is OK. The problem is that if I set a call forwarding on one of the two line on the SPA only one phone rings. Openser receives the 302 message from the SPA but the failure route that contains t_check_status("302") is not called. The forwarding works if only one line is connected. How can I handle the 302 in this situation? Thanks in advance. Regards, Antonio.

1 0

[OpenSER-Users] sipwise config error - save location?
by Robert McNaught 14 Apr '08

14 Apr '08

Please discard - fix for this was a curly bracket on line 313 - not sure if that was sipwise or me! Robert

1 0

[OpenSER-Users] sipwise config error - save location?
by Robert McNaught 14 Apr '08

14 Apr '08

Hi, Has anyone ever seen this error before? I created a config file using the sipwise utility: [root@ser1 openser]# openser -c 0(30469) parse error (314,1-3): syntax error 0(30469) parse error (314,1-3): ERROR: bad config file (2 errors) 0(30469) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy [root@ser1 openser]# 314 if(!save("location")) 315 { 316 317 xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); 318 sl_reply_error(); 319 exit; 320 } It complains about line 314 having a syntax error - I have another openser machine running with the same exact line and it is ok? Has anyone seen this before? TIA Robert

1 0

[OpenSER-Users] SIP Load Balanser for OpenSER
by Vladimir Romanov 14 Apr '08

14 Apr '08

Hi! I want test setup OpenSER in HA & performance cluster. For this I need SIP LoadBalancer. Anyone known were I can download vovida lbproxy? Vovida site not answer last time :( Maybe there exist other software SIP load balancers. -- Vladimir Romanov

2 1

[OpenSER-Users] Avoid Loops between Openser and Asterisk
by Cosimo Fadda 13 Apr '08

13 Apr '08

Hi everybody, I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail server. Suppose that an incoming calls arrives to Ser from Asterisk and the user is offline; Then Openser sends back the Invite to Asterisk that should activate the voicemail application but unfortunately it detects a Loop (482) and rejects the invite. I think I have to mangle the SIP message in Openser before the send to Asterisk again. This is what I currently do: if(!lookup("location")) { #send to voicemail if active but not registerd xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if($avp(s:vmail) == "1") { route(11); } else { route(20); } } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(9); } exit; .... route[11] { xlog("L_INFO", "Forwarding request to VM\n"); prefix("vm"); sethostport("asterisk-gw:5060"); #append_branch(); if (!t_relay()) { xlog("L_INFO", "Unable TO Forward the request to VM\n"); route(20); exit; } What can I do to avoid Loop Detection? Thanks in advance, Cosimo Fadda ___________

9 26

[Serusers] Asterisk with SER ...great solution
by satish patel 12 Apr '08

12 Apr '08

find solution on my website http://linuxbug.org/index_files/projects.html $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

1 0

[OpenSER-Users] Openser problem
by Norberto Monteiro 11 Apr '08

11 Apr '08

Hello, I'm having a problem that is the following: I'm using rewriteport to send traffic to another proxy. Then When I make a call and if I hung up the other phone still rings. What could be the problem?

1 0

[Serusers] Questions re. MySQL
by frq ser 11 Apr '08

11 Apr '08

Hi, Here's a set of basic questions for which I need your help. I want to play different announcements based on the caller's history. For simplicity, I want ann1_user1.wav to be played for the first call, and ann2_user1.wav for all subsequent calls from the 1st user. Similarly for 2nd user, the files would be ann*_user2.wav. 1) I want to have a basic record of the number of times a user has called the SER(ie, sent an INVITE). Which table and column in mySQL can I use for this purpose. Also, how can I write the integer using SER commands (ie something that I can put in ser.cfg file). There is an example ( http://www.iptel.org/write_custom_text_to_mysql_acc_table) which uses 'acc_db_request', but it can only write to sip_status column in acc table. How can I modify /write to to any general field in mysql? 2) Subsequently, how can I retreive this value from mysql through some command in ser.cfg file. I want to use this value in an 'if' statement and redirect the process flow accordingly. 3) I was thinking of rewriting the uri to make it 'ann1_user1@xxxxx' and sending the request to early_announce application so that the user hears different announcement based on username and their call history (ie whether this user has called before or not). However, I am not very clear about how to do this. Is there any simpler method to do this. The final aim of this whole exercise is to demonstrate call differentiation based on username and user history. So if someone has a better idea then please do share it. Thanks for the help and accept my apologies for squeezing so many questions into a single message. Regards, Frq

2 3

[OpenSER-Users] My application destroys registration
by kriko 11 Apr '08

11 Apr '08

Hi! I have a problem with wesip, it works but it destroys client registration. I added simple if statement, which executes app on wesip server (only on INVITE): #app server conn if(!as_relay_t("examplesvr")){ t_reply("500","No App Server"); } If you look at wesip log: http://files.myopera.com/kriko/code/seas_log.txt You can see that wesip doesn't process this request, since the app only responds to INVITE, looking closer at this line: --- SIP/2.0 481 Call Leg/Transaction Does Not Exist --- If you would like to look at app it is here (really simple thing): http://files.myopera.com/kriko/code/sample.java What am I doing wrong? Is it because I'm using t_reply in wrong place or there is something else. I don't think it is my app, since it is not being executed. -- (\__/) (='.'=) (")_(")

1 0

[OpenSER-Users] Openser And DISC Connection close
by Dilip 11 Apr '08

11 Apr '08

Hello Friends, When Openser want to connect with disc client at that time disc client close the connection and gives the folowing error. DEBUG:peer_state_machine: peer 0x2aaaaaaf33a8, state = 3, event=0 ERROR:peer_state_machine: event - state mismatch : peer=0x2aaaaaaf33a8, state=3, event=0 INFO:do_receive: timeout received Actually Openser and DISC can not ESTABLISH the connection and gives CLOSEWAIT in the netstat command. So what will be problem ,please guide me. Regards, Dilip

1 0

[Serusers] Problem related to exec_msg
by frq ser 11 Apr '08

11 Apr '08

Hi, I am trying to understand the 'exec_msg' command as described in Section 4 of SER Admin Guide. I am just trying to echo the user name (1001) of the caller on the screen. I use the config file given at the end. On receiving the INVITE message, SER DOES echo the user name, but I also get some warning and error messages as shown below: #> /opt/ser-sems/sbin/ser -f /xxx.../voip/ser_expt.cfg Listening on udp: 192.168.3.208 [192.168.3.208]:5070 tcp: 192.168.3.208 [192.168.3.208]:5070 Aliases: tcp: xxx:5070 udp: xxx:5070 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing exec - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 2803 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(2801) Main Route... 0(2801) Inside search from-1001... 1001 0(2801) child process 2805 exited normally, status=0 0(2801) INFO: dont_fork turned on, living on 0(2801) WARNING: SIGPIPE received and ignored 0(2801) ERROR: exec_msg: cmd echo '1001' failed. exit_status=-1, errno=10: No child processes 0(2801) WARNING: receive_msg: error while trying script Can someone tell me how I can get rid of these messages. My aim was just to get the user name displayed, and this has been achieved. I just want to understand the reason for these warnings and error messages. Thanks Frq PS: This is my config file: # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen=192.168.3.208 #2.13 # put your server IP address here port=5070 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" unix_sock="/tmp/ser_sock" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/opt/ser-sems/lib/ser/modules/mysql.so" loadmodule "/opt/ser-sems/lib/ser/modules/sl.so" loadmodule "/opt/ser-sems/lib/ser/modules/tm.so" loadmodule "/opt/ser-sems/lib/ser/modules/rr.so" loadmodule "/opt/ser-sems/lib/ser/modules/maxfwd.so" loadmodule "/opt/ser-sems/lib/ser/modules/usrloc.so" loadmodule "/opt/ser-sems/lib/ser/modules/registrar.so" loadmodule "/opt/ser-sems/lib/ser/modules/textops.so" loadmodule "/opt/ser-sems/lib/ser/modules/exec.so" route { log(1, "Main Route...\n"); # sl_send_reply("404", "User Not Found...:-("); # if (uri=~"^sip:200@") { # exec_msg("echo 'Frq'"); # } if (search("(f|From): .*1002@")) { log(3, "Inside search from-1002...\n"); exec_msg("echo '1002'"); }; if (search("(f|From): .*1000@")) { log(3, "Inside search from-1000...\n"); exec_msg("echo '1000'"); }; if (search("(f|From): .*1001@")) { log(3, "Inside search from-1001...\n"); exec_msg("echo 1001"); }; }

1 0

[OpenSER-Users] Recommendations for IM/Voice/Video client
by Adrian Georgescu 11 Apr '08

11 Apr '08

For those interested, you can monitor the progress of development for the multi party chat functionality based on MSRP here: http://www.msrprelay.org/ http://www.msrprelay.org/wiki/MSRPChatSoftwareDesign Adrian >>>>>>>>>>>> I've tryed Bria but in its web page it appears: IM Management - Send instant messages to groups - Use IM tabs for multiple IM conversations in one window - See IM alerts for incoming IM It's not as a IM conferencing in a server side, but maybe enought... Anyway, you can do a conference with IMC module and create contacts like "IM room 1", "IM room 2" in each client. Of course, for now there is no way to know which users are joining a conference. There are two RFC's about it: - SIP Event Package for Conference State http://tools.ietf.org/html/rfc4575 - SIP Call Control - Conferencing for User Agents http://tools.ietf.org/html/rfc4579 but AFAIK there are not yet client side implementation. -- Iñaki Baz Castillo

1 0

[OpenSER-Users] (no subject)
by Hamed al-Suhli 10 Apr '08

10 Apr '08

Hi I'm new to openser and also Linux server install and configuration, I'm profissional at windows and its network, but need to install server for openser on local small lan. also will be used as backup for data. what is the easiest server to install? -- hamed suhli http://sourceforge.net/projects/tarmeez

2 1

[OpenSER-Users] Recommendations for IM/Voice/Video client
by Robert McGilvray 10 Apr '08

10 Apr '08

This is a little off-topic, but figured it was a good place to ask. We're interested in deploying OpenSER coupled with a SIP client primarily used for IM, but also for voice and video conferencing. I'm looking for feedback and recommendations on clients from people that may do this already or have been involved with it. So far one of the better ones looks like Bria (a step up from Eyebeam) from CounterPath. One short coming though is the lack of IM conferencing. I could probably rig something up with IMC but our users would prefer to have a clicky GUI for it. This is for an enterprise and it'd be nice to be able to brand it. Thanks -Bob

3 2

[OpenSER-Users] forwarding responses to early in TM
by Taisto Qvist 10 Apr '08

10 Apr '08

Hi folks, When I am doing forking I am getting a puzzling result concerning forwarding of responses, and I'm wondering how script needs tweaking. When recieving a 6xx on one branch, the TM-module automatically generates CANCEL to cancel the remaining branches(I wouldnt mind if this was configurable), but the puzzling thing is that it forwards the 603 _before_ it has terminated the remaining client transactions. (It sends cancel but it doesnt wait for final responses of the invites its cancelling, just forwards 6xx directly.) A similar scenario happends when I let Timer C pop. My TM-module will send a final response on the server-side at the same time as it starts to cancel the client side. But since one of the client txns just might result in a 2xx(INV) I cant forward the final response until ALL client txns are done, if I am going to follow the rfc properly. I thought the TM module would keep track of all client txns for "me", but it seems it isnt, so is this something I should "manually" handle in my script? Regards Taisto Qvist

4 3

[Serusers] Automatic end of call
by inge 10 Apr '08

10 Apr '08

Hi all, In find some informations about an automatic end of call system. The aim is to cut the call after 2 hours. So it's not specificly link to a pre-paid system, but for all my ser user, I want to end the call after 2 hours. Is there a existing solution to do that ? Or maybe some idea to deal with this project ? Thank you !

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[OpenSER-Users] Accounting Time on dialogue SIP
by Marcio 10 Apr '08

10 Apr '08

Hi... Anybody could me help, I need account time within dialogue SIP( from reply of the Invite -> OK until the BYE-> finish dialogue) and to save for a SGBD (MySQL - That's autenthication's responsible)....... Does anyone know how...? Any hints.....?? thx... Marcio

4 4

[OpenSER-Users] Can't register on OpenSER using SIPp
by Vladimir Romanov 10 Apr '08

10 Apr '08

Hello All! I can't register in OpenSER using SIPp. SIPp versions tested 2.0.1 Windows (TLS), Latest from SVN on linux X-Lite registers w/o any problems. In pcap file & messages you can see 2 registration attempts. first form x-lite (successful) and second from SIPp. This looks like OpenSER and SIPp have some differences in MD5 authentication. -- Vladimir Romanov

1 0

[OpenSER-Users] Can't register on OpenSER using SIPp
by Vladimir Romanov 10 Apr '08

10 Apr '08

I can't register in OpenSER using SIPp. SIPp versions tested 2.0.1 Windows (TLS), Latest from SVN on linux X-Lite registers w/o any problems. In pcap file & messages you can see 2 regsitration attempts. first form x-lite (sucessfull) and second from SIPp. This looks like OpenSER and SIPp have some differences in MD5 authentification. -- Vladimir Romanov

1 0

[OpenSER-Users] Sip server tutorial
by kriko 10 Apr '08

10 Apr '08

Hi! I was assigned to explore OpenSer, however I'm more green on this stuff. Is there a good basic documentation/tutorial on how sip servers and systems around works (basic stuff like routing)? -- (\__/) (='.'=) (")_(")

4 4

[OpenSER-Users] Wants to reopen auth_diameter module
by Dilip 10 Apr '08

10 Apr '08

Hello Friends, There is a auth_diameter module in the openser,But now no one is working on that. I want to make it compatible with the opendiameter. So that we can implement AAA protocol in the complete manner. For that i want your help for the completion of this module. Regards, Dilip

2 1

[OpenSER-Users] survey: Who is using serweb?
by Klaus Darilion 10 Apr '08

10 Apr '08

Which version do you use? Does it work at all with current version of openser (1.3)? PS: this is to find out if serweb supported can be removed from openser regards klaus

10 14

[Serusers] good news for German ser fans: book
by Jiri Kuthan 09 Apr '08

09 Apr '08

I got today this book on my desk, courtesy of one of its authors. https://www.opensourcepress.de/index.php?26&backPID=178&tt_products=194 Whereas I was busy with a bunch of unrelated things, the book was circulating in the iptel office all day and has found its keen readers. Unfortunately still only in German lanugage. -jiri

5 5

[Serusers] select for canonical username (alias)?
by Stefan Sayer 09 Apr '08

09 Apr '08

Hello, is there a select to get the canonical username? or is 'canonical username' now a deprecated concept? I know there is the uid, but the problem is, with 36 characters uid is too long for my purpose (user id for voicebox). thanks Stefan -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

3 3

[Serusers] about URI request
by liu york 09 Apr '08

09 Apr '08

hi i want to distribute tow alieas users use different pstn gw with same area code. i add below in my cfg.but it doesnt work.any advice for me ? many many thx. if (uri=~"^sip:0060.*@(my\.test\.com)"){ strip(2); prefix("21"); rewritehostport("1.2.3.4:5060"); # This is where PSTN Gateway installed route(1); return; }; if (uri=~"^sip:0060.*@(your\.test\.com)"){ strip(2); prefix("22"); rewritehostport("5.6.7.8:5060"); # This is where PSTN Gateway installed route(1); return; };

2 1

[OpenSER-Users] Invalid cseq for aor - OpenSER stuck?
by Adrian A 09 Apr '08

09 Apr '08

I am running OpenSER 1.3.1 on a Debian 4.0 system. It seems that there's some sort of bug with OpenSER getting stuck and stops processing requests at least once a day. The logs only show the following line repeated every minute: ERROR:registrar:update_contacts: invalid cseq for aor <1302> At 1302 is a Cisco 7905 phone running the latest firmware. If I restart OpenSER then things get back to normal. Does anyone know what the problem might be?

2 2

[Serusers] SER 0.9.7-pre3 and DNS SRV
by Steven C. Blair 09 Apr '08

09 Apr '08

I have a SER 0.9.7-pre3 system which is working fine except for call transfers. It turns out that when the NOTIFY is sent after the REFER method the proxy performs a queried for only A records not SRV. This results in an IP address being returned which is a non-SIP host. This is because the domain name is being used by different services some of which use A records for resolving the name and some, like SIP, are suppose to use SRV. How can I make this version of SER perform an SRV lookup instead of an A record lookup when initiating SIP messages? Thanks, Steve

1 1

[OpenSER-Users] OpenSer and sip calls
by kriko 09 Apr '08

09 Apr '08

Hi! I'm trying to get openser working properly - it starts up, I can even regiter from softphone (linphone / twinkle) but I cannot make calls. OpenSer is running on 10.99.8.103:5060 (this is my machine, IP is bound to eth0), when I try to call to e.g.: sip:demovoice@10.99.8.48 I get some errors as in this log: http://files.myopera.com/kriko/code/openser.tar.gz Configuration is included too. Softphone clients are properly configured - it is the proxy that is causing me problems. Any kind of information is appreciated. -- kriko

2 1

[OpenSER-Users] Thoughts on openSER + Asterisk scenarios
by Gabriel Ortiz Lour 09 Apr '08

09 Apr '08

Hi all, We are working here on a solution for a large scale VoIP system to cover our entire state, using both openSER and Asterisk. The project is on an initial stage, basically deciding architecture, so I would like to share our thoughts here in the mailing list and hopefully get some feedback to get starting on the right foot. We have our own private network (internet only on a further first step, no NAT), and we think about two possible scenarios: 1: The SIP accounts would register on the local Asterisk and those would talk to a SER core to connect them all, something like "Asterisk: Oh! this account is not mine so I'll send it to the core to take care for me" and the SER core would try to reach the other endpoint or return some error code. 2: All SIP accounts would be registered on the SER core, using the local asterisk servers as mediaservers and backup SIP proxies, where the SIP accounts would be registering on them only when there is some problem on the link with the SER core (and here they would have to have their own local PSTN gateways) This is the two basic ideas, so we would like to have any thoughts about the "pros and cons" of them. We have also planned many PSTN gateways that would be used for LCR (making all PSTN calls = local) for the entire system, that should be considered on a failure event. Thanks, Gabriel Ortiz PS.: Any help/thoughts are very welcome, as we will share all the experience and development on this project with the FOSS community. We are a government IT company from Paraná, Brasil aligned with the Free Software movement, and the aim is to document and package every aspect of this project.

1 0

Re: [OpenSER-Users] [openser 1.3] loading module auth.so stop my server (bug?!)
by Pascal Maugeri 09 Apr '08

09 Apr '08

I'm still troubleshooting this issue and this morning I added more outputs in the source code of auth_mod.c. And I discovered: 1) when I set the configuration parameter "secret" of the module auth (modparam("auth", "secret", "secretphrase) ) in openser.cfg everything works fine ! 2) when I remove the configuration parameter "secret", the auth_mod.c calls the method generate_random_secret() as stated in the documentation. It enters in the loop for(i = 0; i < RAND_SECRET_LEN; i++) { LM_INFO("(3a)\n"); sec_rand[i] = 32 + (int)(95.0 * rand() / (RAND_MAX + 1.0)); } LM_INFO("(4)\n"); but never exit ! This is the reason why my server does not start and keeps blocked there. I modified the code above to print more logs, I can see in the logs the string (3a) once but I never (4): IMHO it is blocked somewhere in the first random computation. I'm running openser on Redhat Enterprise v4. Is there anything else I should configure in my environment ? Or I found a bug ? Regards, Pascal On Tue, Apr 1, 2008 at 12:45 PM, Pascal Maugeri <pascal.maugeri1(a)gmail.com> wrote: > Hi > > I'm still unable to have a working server 1.3 that loads the module > auth.so. > > It is weird because there is no error message in my log file also I set > the debug level to maximum. But after starting the openser service, it says > it is successful but there is no running process (pgrep openser does not > show anything, when it does if I remove the load auth.so)! > > How can I get more debugging info about the loading and initialization of > auth ? > > Can anybody share with me his or her configuration file ? I may miss > something in mine: > loadmodule "sl.so" > [...] > loadmodule "auth.so" > (I believe it is sufficient). > > Is there any known issue using Redhat Enterprise 4 ? > > Regards, > Pascal > > >

3 4

Re: [OpenSER-Users] Specifying IP Address for SIP Responses
by Bogdan-Andrei Iancu 09 Apr '08

09 Apr '08

Hi Victor, if a UAC wants to have asymmetric signalling, just send from IP_A:PORT_A and put in top most via IP_B:PORT_B - via is used for routing back the reply. regards, Bogdan Victor Cartes wrote: > Thanks, Iñaki. But It is not what I really need. > > I need to send to an UAS a SIP Request, like OPTIONS, and tell it to > send the reply to a different transport address (IP Addr + Port) than > the source address of the request. > > This is, basically, in order to perform NAT discovering and typing > from the server. > > Thanks in advance, > > Víctor > > > > > 2008/4/7, Victor Cartes <ietf.sip(a)tivatechnologies.com > <mailto:ietf.sip@tivatechnologies.com>>: > > Actually not. What I need the UAS exactly to do is to send the SIP > reply to a different IP address than the source of the SIP Request > that it is replying to. Just how it sends the replies to the UDP > port specified in top most Via HF. > > > 2008/4/7, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi Victor, > > Not sure I got your question correctly, but you mean like > something like "received" via param used to route back the > replies? > > Regards, > Bogdan > > Victor Cartes wrote: > > Hello to all, > > About SIP/UDP: Is there any way to specify the IP address > for SIP responses, in a similar fashion as port specified > on top most Via HF? > > Thanks in advance, > > Víctor > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > >

3 3

[OpenSER-Users] Sametime
by r007b17 09 Apr '08

09 Apr '08

Hi all, i have a lotus sametime server (SIP) and a Jabber server (XMPP). Is it possible messaging with each other using xmpp Module ? Thanks in advance. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 3

Re: [OpenSER-Users] CPL and Postgres problems.
by Bogdan-Andrei Iancu 09 Apr '08

09 Apr '08

Hmm....that is something strange - probably there is something mixed with the postgres data types..... Could you open a bug report for further investigation? Regards, Bogdan Ola Karlsson wrote: > It was TEXT and I changed it to BYTEA and that seemed to do the trick. > > /Ola > > On Mon, Apr 7, 2008 at 11:57 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > So, you only changed the column type (for cpl_bin) from BLOB to > BYTEA ? > > Regards, > Bogdan > > Ola Karlsson wrote: > > Bogdan.. > > > > I changed the column type in the database to bytea for cgi-bin and > > stuff started working , seems the decode only happened for BLOB type > > not for text. > > > > When debugging I got a \ instead of 0x01 så figured something was > > wrong getting stuff out of the database. > > > > /Ola > > > > On Mon, Apr 7, 2008 at 10:50 AM, Bogdan-Andrei Iancu > > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro> > <mailto:bogdan@voice-system.ro <mailto:bogdan@voice-system.ro>>> > wrote: > > > > Hello Ola, > > > > That is strange as the len of the script is taken from the > > postgres driver (via openser API) - the returned columns must be > > BLOB type. Check the types of the cpl_* columns; otherwise I > > suspect a postgres driver (in openser) issue. > > > > Best regards, > > Bogdan > > > > Ola Karlsson wrote: > > > > Hi Users! > > > > Have a problem with CPL scripts and postgres. > > I've tried to make the simplest possible script for > debugging > > which resulted in the following in the 'cpl_xml' column. > > > > '<cpl>\012 <incoming>\012 <reject status="reject" > > reason="I reject anonymous calls"/>\012 > > </incoming>\012</cpl>\012\012' > > > > Loading it with openserctl that gets translated to this > in the > > cpl_bin field. > > > > > '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I > > reject anonymous calls\000\000' > > > > The reason I figure that postgres could have something to do > > with this is that I've got this working before(before being > > quite a while ago though. ) with MySQL. > > > > Turning on debugging gives this information. > > > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_dest_user: trying to get user from new_uri > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_dest_user: trying to get user from R_uri > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_user_script: fetching script for user <olak> > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_user_script: we got the script len=0 > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script > > > > OpenSER says that the length of the script is 0 whatever > > script I load?. > > I changed that in the CPL-C module so we get the right > length > > of the compiled script but then it falls over on the sanity > > check for NODE_CPL = 001 , and if you skip that check it > > starts complaining about unknown nodes ( in my case 92 ) ... > > > > So my question is if anyone uses postgres and CPL, if so can > > you give me some pointers? > > Would be good to know if this happens to be a bug or an > > untested area in that case I have no problem fixing the > > problem just don't want to burn alot of time on something > > already fixed or me doing something stupid with > configuration. > > > > Regards > > /Ola Karlsson > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > <mailto:Users@lists.openser.org <mailto:Users@lists.openser.org>> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Serusers] Adding authentication through mySQL
by frq ser 09 Apr '08

09 Apr '08

Hi, I am trying to add user authentication through mySQL database to my SER (0.9.7-pre1). When I tried to run SER with the modifed ser.cfg file, it gave the error that it could not locate the 'mysql.so' module. Then I tried to 'make' the mysql module following the instructions given here: http://www.iptel.org/faq/why_does_ser_fail_to_start_with_an_error_load_modu… When I typed 'make modules=modules/mysql modules' at the SER source directory, it gives me the error message saying that it could not locate 'mysql.h' and 'errmsg.h' (complete log given at the end of this email). I've tried to locate these files on my computer but couldn't find them. Note that mySQL is working fine on its own. I need your suggestions as to how I should proceed. Thanks in advance for any help. Frq

2 3

[Serusers] ser sip-t or sip-i
by mashudi 09 Apr '08

09 Apr '08

Dear all, is ser can handle sip-t or sip-i invite message ? have some one got experience with sip-t or sip-i using ser ?

1 0

[OpenSER-Users] [Fwd: Re: Bug with OpenSER 1.3 Presence module : event handling.]
by Anca Vamanu 09 Apr '08

09 Apr '08

-------- Original Message -------- Subject: Re: [OpenSER-Users] Bug with OpenSER 1.3 Presence module : event handling. Date: Wed, 09 Apr 2008 11:20:04 +0300 From: Anca Vamanu <anca(a)voice-system.ro> To: Jeremy McNamara <jj(a)nufone.net> CC: Iñaki Baz Castillo <ibc(a)aliax.net>, users(a)openser.org References: <b769badb0710240220pe56ebe2mf617230fdb667586(a)mail.gmail.com> <200710241333.09880.ibc(a)aliax.net> <471F34B5.6040806(a)voice-system.ro> <47FBB0CE.4000909(a)nufone.net> Hi Jeremy, Firstly, no, the message: '"Missing or unsupported event header field value / ^Ievent= message-summary"' has nothing to do with presence. It is about another feature - notifications about whether the user has voice mail messages pending. Now, since presence does not work for you with xlite, I am wandering if you have clicked the option Show contact availability , or something like this, when editing the buddy details. It is at the end of the form. regards, Anca Vamanu Jeremy McNamara wrote: > Anca Vamanu wrote: > >> Hello, >> >> A major change in the presence structure occurred in openser 1.3. You >> now need to load two modules to have presence running: presence and >> presence_xml. >> Most probably you didn't load the presence_xml module and this is why >> the error message appears. >> >> > > Currently I am at Interop HotStage in Belmont, CA trying to deploy a > 'unified communication' situation to demonstrate for the Interop Vegas > show here in a few weeks. I am in charge of the OpenSER/Asterisk > components and presence is giving me a huge headache. > > Hopefully someone out there can show me what I am doing wrong. > > I am running Openser v1.3.1 from tarball on Ubuntu Gutsy, no xcap > server (haven't figured out if we really need one or not for our demo...) > > I have loaded both presence and presence_xml. As documented in the > reply (down below) if you notice there is a 'tab' character in the > event= presence, I too experience this, however mine is with > message-summary. > > I see the Subscribe/Publish messages coming into openser and they are > getting routed to the appropriate config block and processed as I have > seen documented, but every so often the "Missing or unsupported event > header field value / ^Ievent= message-summary" error pops up. > > Between two X-Lite softclients using "presence agent' method, we never > seem to get any presence notifications, I presume due to this error, > but I am unsure. > > All dependencies on both presence.so and presence_xml.so seem to be > met (verified with ldd) > > # ldd presence_xml.so > linux-gate.so.1 => (0xffffe000) > libxml2.so.2 => /usr/lib/libxml2.so.2 (0xb7e76000) > libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7d2c000) > libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0xb7d27000) > libz.so.1 => /usr/lib/libz.so.1 (0xb7d12000) > libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb7ced000) > /lib/ld-linux.so.2 (0x80000000) > > (identical for presence.so) > > > Here are the relevant portions of my config: > > loadmodule "presence.so" > modparam("presence", "db_url", "mysql://XXX:XXX@localhost/openser") > modparam("presence", "max_expires", 60) > modparam("presence", "server_address", "sip:valid.ip.address:5060") > > loadmodule "presence_xml.so" > modparam("presence_xml", "db_url","mysql://XXX:XXX@localhost/openser") > modparam("presence_xml", "force_active", 1) > > ... > > if(is_method("PUBLISH|SUBSCRIBE")) { > route(2); > } > > ... > > route[2] { > xlog("L_INFO", "Dealing with $rm F:$fu T:$tu\n"); > sl_send_reply("100","trying"); > if (!t_newtran()) { > sl_reply_error(); > exit; > }; > > append_to_reply("Contact: <sip:valid.ip.address:5060>\r\n"); > if(is_method("PUBLISH")) { > if($hdr(Sender)!= NULL) { > handle_publish("$hdr(Sender)"); > } else { > handle_publish(""); > } > t_release(); > } else { > if( is_method("SUBSCRIBE")) { > handle_subscribe(); > t_release(); > } > } > exit; > } > > > > > If anyone wants to help me troubleshoot and make this work, I will > make sure you get credit for assisting us here at Interop. > > > > > Jeremy McNamara > > > >>> El Miércoles, 24 de Octubre de 2007, Antoine Fressancourt escribió: >>> >>> I've now installed SVN version and same erro occurs to me: >>> >>> Oct 24 13:32:37 [4653] ERROR:presence:handle_subscribe: Missing or >>> unsupported event header field value >>> Oct 24 13:32:37 [4653] ERROR:presence:handle_subscribe: >>> event= presence >>> >>> The client is Twinkle and presence worked well in 1.2.X. >> > > > > >

1 0

[OpenSER-Users] Missing or unsupported event header field value - presence woes
by Jeremy McNamara 08 Apr '08

08 Apr '08

I am at Interop HotStage in Belmont, CA trying to deploy a 'unified communication' situation to demonstrate for the Interop Vegas show here in a few weeks. I am in charge of the OpenSER/Asterisk components and presence is giving me a huge headache. > Apr 8 11:34:12 openser openser[23798]: > ERROR:presence:handle_subscribe: Missing or unsupported event header > field value Apr 8 11:34:12 openser openser[23798]: > ERROR:presence:handle_subscribe: ^Ievent= message-summary Hopefully someone out there can help me figure out what is going wrong. I am running Openser v1.3.1 from tarball on Ubuntu Gutsy, no xcap server (haven't figured out if we really need one or not for our demo...) I have loaded both presence and presence_xml. If you notice there is what I believe to be a 'tab' character in the event= message-summary error response. I wonder if this tab is causing the issue. I see the Subscribe/Publish messages coming into openser and they are getting routed to the appropriate config block and processed as I have seen documented, but every so often the "Missing or unsupported event header field value and ^Ievent= message-summary" error pops up. Between two X-Lite softclients using 'presence agent' method, we never seem to get any presence notifications, I presume due to the error above, but I am unsure. All dependencies on both presence.so and presence_xml.so seem to be met (verified with ldd) # ldd presence_xml.so linux-gate.so.1 => (0xffffe000) libxml2.so.2 => /usr/lib/libxml2.so.2 (0xb7e76000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7d2c000) libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0xb7d27000) libz.so.1 => /usr/lib/libz.so.1 (0xb7d12000) libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb7ced000) /lib/ld-linux.so.2 (0x80000000) (identical for presence.so) Here are the relevant portions of my config: loadmodule "presence.so" modparam("presence", "db_url", "mysql://XXX:XXX@localhost/openser") modparam("presence", "max_expires", 60) modparam("presence", "server_address", "sip:valid.ip.address:5060") loadmodule "presence_xml.so" modparam("presence_xml", "db_url","mysql://XXX:XXX@localhost/openser") modparam("presence_xml", "force_active", 1) ... if(is_method("PUBLISH|SUBSCRIBE")) { route(2); } ... route[2] { xlog("L_INFO", "Dealing with $rm F:$fu T:$tu\n"); sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; append_to_reply("Contact: <sip:valid.ip.address:5060>\r\n"); if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) { handle_publish("$hdr(Sender)"); } else { handle_publish(""); } t_release(); } else { if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } } exit; } If anyone wants to help me troubleshoot and make this work, I will make sure you get credit for assisting us here at Interop. Jeremy McNamara > > > >

2 6

[Serusers] call recording app
by Mihaly Zachar 08 Apr '08

08 Apr '08

Hi all, I got a job now, I need to build a call recording application, what stores the rtp streams of the calls, what routed to the application. There is an existing routing logic (special sip proxy) what decides if the call should be recorded or not. If the call needs to be recorded, this proxy will route the call to this recorder application. This app needs to send through the call to the next sip hop, but it must store the RTP streams on the fly. What we have now: - we have our own rtpproxy, we can modify it easily if we need to store the streams. - we have our own B2BUA what controls the rtpproxy I just wondering if we can use SER + SEMS for this, or should we build our own system ? What should be the most effective way ? Can it be done with SEMS easily ? Thanks for the suggestions, recommendations. Misi

2 1

[OpenSER-Users] downgrading or removing openser
by Robert McNaught 08 Apr '08

08 Apr '08

Hi, I have googled and found some posts relating to this, but no definitive answers: how can I downgrade from openser 1.2.x to openser 1.1.x - do I just recompile openser 1.1 and it overwrites whatever is in 1.2? there does not seem to be a make uninstall option for make in 1.2. Recompiling does not give any errors, but wont there be modules, etc that are still left over from 1.2.x? Can anyone offer some advice? Robert

2 1

[OpenSER-Users] Specifying IP Address for SIP Responses
by Victor Cartes 08 Apr '08

08 Apr '08

Hello to all, About SIP/UDP: Is there any way to specify the IP address for SIP responses, in a similar fashion as port specified on top most Via HF? Thanks in advance, Víctor

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[OpenSER-Users] ERROR:presence:publ_notify: Could not send notify for presence
by gregor.sp 08 Apr '08

08 Apr '08

Hi all, I am using OpenXcap (0.99) and OpenSER (1.3), presence authorization is working and users receive NOTIFYs when users subscribe to presence. But after updating presence information with PUBLISH no NOTIFY are sent to clients. The debug messages the server shows when the server receives a PUBLISH are: ERROR:presence_xml:get_final_notify_body: while extracting the transformation node ERROR:presence_xml:pres_apply_auth: in function get_final_notify_body ERROR:presence:send_notify_request: in function apply_auth_nbody ERROR:presence:notify: sending Notify not successful ERROR:presence:publ_notify: Could not send notify for presence Somebody could help me with this? Thanks. Gregorio -- View this message in context: http://www.nabble.com/ERROR%3Apresence%3Apubl_notify%3A-Could-not-send-noti… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] Session not stop
by Sergey Babichev 08 Apr '08

08 Apr '08

Hi all! I have found out a problem at end of a call from voip. If a call the user finishes voip that session is not broken off. If from PSTN that session comes to the end successfully. PSTN Gateway -> Openser with mediaproxy -> VoIP This is log (grep mediaproxy): call from pstn to voip, voip user ended call Apr 8 12:45:40 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:40 ser1-2 mediaproxy[15711]: session F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x: 529/302/831 packets, 105720/58476/164196 bytes (caller/called/relayed) Apr 8 12:45:40 ser1-2 mediaproxy[15711]: session F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x: ended. Apr 8 12:45:40 ser1-2 mediaproxy[15711]: execution time: 0.44 ms Apr 8 12:45:41 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:41 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:42 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:42 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:44 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:44 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:48 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:48 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:52 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:52 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:56 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:56 ser1-2 mediaproxy[15711]: execution time: 0.17 ms Apr 8 12:46:00 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:00 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:04 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:04 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:08 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:08 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:30 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:30 ser1-2 mediaproxy[15711]: execution time: 0.17 ms Apr 8 12:46:30 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:30 ser1-2 mediaproxy[15711]: execution time: 0.16 ms Apr 8 12:46:31 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:31 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:33 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:33 ser1-2 mediaproxy[15711]: execution time: 0.16 ms Apr 8 12:46:37 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:37 ser1-2 mediaproxy[15711]: execution time: 0.15 ms This is log (grep mediaproxy): call from pstn to voip, pstn user ended call Apr 8 12:53:26 ser1-2 mediaproxy[15711]: session 109131A1-48011DD-A33BB2F1-2DDABEF8(a)x.x.x.x: ended. Apr 8 12:53:26 ser1-2 mediaproxy[15711]: execution time: 0.40 ms What's problem maybe? -- Best regards, Sergey Babichev

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[OpenSER-Users] SeasMessageEvent [ExpressMChannel[12]] - ERROR parsing a SEAS message event!!!
by bsb093 08 Apr '08

08 Apr '08

Hi all, I am trying to run OpenSER 1.2 with WeSIP. After the callee (shine_sip1) answer the call I get the following exception and the call hangs up after a BYE message from the caller to the calee. ExpressMessageChannel [ExpressMChannel[12]] - Done processing ACK sip:shine_sip1@192.168.2.101:5070 SIP/2.0 ExpressMessageChannel [ExpressMChannel[12]] - peerAddress = 192.168.2.100/5060 SeasMessageEvent [ExpressMChannel[12]] - ERROR parsing a SEAS message event!!! java.lang.StringIndexOutOfBoundsException: String index out of range: -243 at java.lang.String.checkBounds(Unknown Source) at java.lang.String.<init>(Unknown Source) at com.voztele.javax.sip.header.Via.<init>(Via.java:112) at com.voztele.javax.sip.header.ViaList.<init>(ViaList.java:43) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:247) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPRequest.<init>(SIPRequest.java:154) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:355) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Unknown Source) Exception in thread "ExpressMChannel[12]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Unknown Source) SeasMessageEvent [SeasRouter] - New message received, from 192.168.2.100:5060 to 192.168.2.100:5060 over UDP Please advice what can cause this exception. Thanks in advance, Shine -- View this message in context: http://www.nabble.com/SeasMessageEvent----------ExpressMChannel-12-----ERRO… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] authentication using openser
by bhargav.b＠renovau.com 08 Apr '08

08 Apr '08

hi all, i am trying to do authentication using openser at the front end and diameter as the back end.i have installed openser and opendiameter and they are running fine independently.but when i send needed avp's from openser towards opendiameter,the opendiameter client is clsing the connection.i donot understand why this is happening.if any body have any idea or if anybody provide me the refference to any study material regarding the integration of openser and diameter client,it would really help me a lot. seeking ur help regards bhargav.b

1 0

[OpenSER-Users] tee with openser
by Robert McNaught 07 Apr '08

07 Apr '08

Hi, This is probably a simple one - I have tried googling, but am not getting the answer to this. I am trying to catch what is displayed on screen and put it into a text file using tee so I can look at it. I can do 'free -h | tee dump.txt' and dump.txt is populated with data I can do 'asterisk -cvvvv | tee dump.txt' and dump.txt is populated with data when i try 'openser | tee dump.txt' and dump.txt does not get populated with the screen info. I have changed permissions to 666 on the dump.txt file incase that was the issue. Am I doing some silly? I thought that tee only copied what was displayed on screen. Thanks Robert

2 1

[OpenSER-Users] openser devel guide
by Daniel-Constantin Mierla 07 Apr '08

07 Apr '08

Hello, I want to announce the availability of the first version for openser devel guide, online at: http://www.asipto.com/pub/openser-devel-guide/ Written together with Elena-Ramona Modroiu, the tutorial approaches the important components of openser: shared memory, locking, pseudo-variables, transformations, mi commands, statistics, db api, timer, module development a.s.o. Hopefully, the guide will help new developers to understand easier the internals of openser and contribute back new code or improvements to existing code. Has a set of examples and references to the code to be able to learn quicker how to use the APIs. The document is under polishing process towards clean docbook and pretty printing (pdf) format, then the sources will be included in the openser repository to be more accessible for improvements. It will be a continuing work-in-progress document, please report any error you find there. It is also mirrored at: http://www.openser.org/docs/openser-devel-guide/ Cheers, Daniel -- http://www.asipto.com

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[OpenSER-Users] OpenSER Summit 2008 - Post facts
by Bogdan-Andrei Iancu 07 Apr '08

07 Apr '08

Hello everybody, The most important OpenSER event, community oriented, the OpenSER Summit 2008 marked a new point in OpenSER project history. Gathering more than 60 participants from both industry and community side, OpenSER Summit brought in to the lights major project people and industry representatives who help with the evolution of the project. Major topics that were discussed: A. how can OpenSER be used as a core component in industry proofed solutions and how/where to get the necessary knowledge B. what are the development directions that needs to be taken by the project - these were moderated open discussions. All this theoretical discussions were back-end up by the OpenSER Pavilion, were several companies showed the practical part of OpenSER software, via services, products and knowledge. All the material that were presented during the Summit are now available for download: A. via the summit agenda - http://www.openser.org/mos/view/OpenSER-Summit-2008/#agenda - here, by browsing the agenda, you can download a certain presentation; B. via the pub/ section of the web site - http://www.openser.org/pub/OpenSER-Summit-2008/ - gets all the presentations from here. Pictures and video captures will be soon available on the web site ;) Best regards, Bogdan

1 1

Re: [OpenSER-Users] CPL and Postgres problems.
by Bogdan-Andrei Iancu 07 Apr '08

07 Apr '08

So, you only changed the column type (for cpl_bin) from BLOB to BYTEA ? Regards, Bogdan Ola Karlsson wrote: > Bogdan.. > > I changed the column type in the database to bytea for cgi-bin and > stuff started working , seems the decode only happened for BLOB type > not for text. > > When debugging I got a \ instead of 0x01 så figured something was > wrong getting stuff out of the database. > > /Ola > > On Mon, Apr 7, 2008 at 10:50 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hello Ola, > > That is strange as the len of the script is taken from the > postgres driver (via openser API) - the returned columns must be > BLOB type. Check the types of the cpl_* columns; otherwise I > suspect a postgres driver (in openser) issue. > > Best regards, > Bogdan > > Ola Karlsson wrote: > > Hi Users! > > Have a problem with CPL scripts and postgres. > I've tried to make the simplest possible script for debugging > which resulted in the following in the 'cpl_xml' column. > > '<cpl>\012 <incoming>\012 <reject status="reject" > reason="I reject anonymous calls"/>\012 > </incoming>\012</cpl>\012\012' > > Loading it with openserctl that gets translated to this in the > cpl_bin field. > > '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I > reject anonymous calls\000\000' > > The reason I figure that postgres could have something to do > with this is that I've got this working before(before being > quite a while ago though. ) with MySQL. > > Turning on debugging gives this information. > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_dest_user: trying to get user from new_uri > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_dest_user: trying to get user from R_uri > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_user_script: fetching script for user <olak> > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_user_script: we got the script len=0 > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script > > OpenSER says that the length of the script is 0 whatever > script I load?. > I changed that in the CPL-C module so we get the right length > of the compiled script but then it falls over on the sanity > check for NODE_CPL = 001 , and if you skip that check it > starts complaining about unknown nodes ( in my case 92 ) ... > > So my question is if anyone uses postgres and CPL, if so can > you give me some pointers? > Would be good to know if this happens to be a bug or an > untested area in that case I have no problem fixing the > problem just don't want to burn alot of time on something > already fixed or me doing something stupid with configuration. > > Regards > /Ola Karlsson > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

2 1

[OpenSER-Users] sdp_mangle_ip()
by Travis Hegner 07 Apr '08

07 Apr '08

Hello All, I have briefly looked through the bug list but couldn't find anything related to my problem. I thought I would check here first to see if perhaps I have a configuration issue. I'm currently using the latest 1.3 version downloaded with svn. When attempting to call "sdp_mangle_ip()" I end up with a malformed sip packet. Upon inspection, I realized that the packet is correct except for the "Content-Length:" header, which seems to have the new (correct) content-length of the new packet written twice into the data stream. Here is what a snippet of the packet looks like: Content-Length: 156\r\n156\r\n Obviously, a correct packet should look like: Content-Length: 156\r\n Has anyone experienced this error before? Is there some parameter I should be setting for the mangler module, or another function I should call before this one? Should I roll up my sleeves and see if I can patch it myself? Any info is greatly appreciated... Travis

2 1

[OpenSER-Users] CPL and Postgres problems.
by Ola Karlsson 07 Apr '08

07 Apr '08

Hi Users! Have a problem with CPL scripts and postgres. I've tried to make the simplest possible script for debugging which resulted in the following in the 'cpl_xml' column. '<cpl>\012 <incoming>\012 <reject status="reject" reason="I reject anonymous calls"/>\012 </incoming>\012</cpl>\012\012' Loading it with openserctl that gets translated to this in the cpl_bin field. '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I reject anonymous calls\000\000' The reason I figure that postgres could have something to do with this is that I've got this working before(before being quite a while ago though. ) with MySQL. Turning on debugging gives this information. Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_dest_user: trying to get user from new_uri Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_dest_user: trying to get user from R_uri Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_user_script: fetching script for user <olak> Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_user_script: we got the script len=0 Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script OpenSER says that the length of the script is 0 whatever script I load?. I changed that in the CPL-C module so we get the right length of the compiled script but then it falls over on the sanity check for NODE_CPL = 001 , and if you skip that check it starts complaining about unknown nodes ( in my case 92 ) ... So my question is if anyone uses postgres and CPL, if so can you give me some pointers? Would be good to know if this happens to be a bug or an untested area in that case I have no problem fixing the problem just don't want to burn alot of time on something already fixed or me doing something stupid with configuration. Regards /Ola Karlsson

2 1

[OpenSER-Users] OpenSER on top of OpenIMSCore
by Habib, Yeasin 07 Apr '08

07 Apr '08

Hi, I want to install OpenSER server on top of OpenIMSCore. I am using the new VMWare image provided by Vladimir Romanov. I have tried to install it from the source from svn. But up to installation I was successful. But when I try to run the mysql database script it's not working. I am trying to run the script /usr/local/etc/openser/openserctlrc, but it's not working anyway. Because it doesn't seems like a shell script. If anybody installs OpenSER on top of OpenIMSCore VMWare image then please help me. If anybody has a working image of OpenSER then it will be really grateful if I have the link or tutorial of that image. I am looking forward to hearing from all of you. Best regards Yeasin Habib

2 1

[OpenSER-Users] How to detect lack of response to SIP nat ping?
by Mike Fedyk 07 Apr '08

07 Apr '08

Hi, I'm trying to come up with some code in openser that will log when a SIP UA does not respond to a SIP NAT ping. The best I can think of so far is using the sip timers module, but it is based on the INVITE method instead OPTIONS that is the nat ping default. Does anyone have any ideas that can point me in the right direction? Thanks, Mike

2 1

[OpenSER-Users] dlg_end_dlg with extra headers
by Alex Massover 07 Apr '08

07 Apr '08

Hi! I'm trying to use dlg_end_dlg with extra headers. As far as I understood I can send fifo command like: [root@SER openser]# openserctl fifo dlg_end_dlg 123 456 'Header: here\r\n' 400 Too few or too many arguments It doesn't seem to accept third extra-header parameter. Am I doing something wrong? -- Best Regards, Alex Massover VoIP Development Manager Jajah Inc.

2 1

[Serusers] SIP Registration fail
by William Zhang 06 Apr '08

06 Apr '08

Hi All, I am new to SER. I might just made some simple mistakes. So be easy on me:) I appreciate any inputs. Basically I was able to install and run SER 2.0.0 RC1 and make call between SIP clients SJPhone 1.65 with a helloworld configuration. Now I am trying to add the Authentication and MySQL. Here are what I did so far and no luck to get it work:( 1. Generate the configuration files with configuration file buildsystem and the content of the file is in the bottom of this message. 2. Create the SER database in MySQL server by running the ser_mysql.sh create command 3. Set SIP_DOMAIN variable: export SIP_DOMAIN="unknown.org" 4. Add domain and user with SER_CTL 2.0 with following commands ser_ctl domain add unknown.org ser_ctl user add 777(a)unknown.org -p 1212 The ser_ctl domain show and user show return: unknown.org unknown.org --------w--c-s 777(a)unknown.org uri 777(a)unknown.org 777(a)unknown.org credentials username=777 realm=unknown.org password=1212 777(a)unknown.org attr datetime_created=2008-04-06 03:25:43 I believe this should set the user and domain correctly to the database. But I also got some weired output with other command like ser_ctl ps, ser_ctl version. Same error: ser_ctl: (97, 'Address family not supported by protocol'): error'> And sercmd also give me the follow errors:( ERROR: connect_unix_sock: connect(/tmp/ser_ctl): Connection refused [111] 5. Run the ser with error to console and no fork. 6. Configure the SJPhone with above registration information. The client can not register and got 401 Unauthorized response. Below are the SIP packets contents: 2008-04-06 04:49:29.843 UDP LOCAL->192.168.0.182:5060 REGISTER sip:unknown.org SIP/2.0 Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc0a800b30000004947f8565900000d5f00000019;rport From: "unknown" <sip:777@unknown.org>;tag=4ff22665b92 To: <sip:777@unknown.org> Contact: <sip:777@192.168.0.179> Call-ID: C69D49CA6580482CAFB5D305213ACE0D0xc0a800b3 CSeq: 18 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 21:49:29 SIP.Network DEBUG 2008-04-06 04:49:29.843 UDP 192.168.0.182:5060->LOCAL SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc0a800b30000004947f8565900000d5f00000019;rport=5060 From: "unknown" <sip:777@unknown.org>;tag=4ff22665b92 To: <sip:777@unknown.org>;tag=3594b9925f930dcc8b38a6cda152d6b5.f97a Call-ID: C69D49CA6580482CAFB5D305213ACE0D0xc0a800b3 CSeq: 18 REGISTER Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.182:5060 "Noisy feedback tells: pid=14897 req_src_ip=192.168.0.179 req_src_port=5060 in_uri=sip:unknown.org out_uri=sip:unknown.org via_cnt==1" The contents of the ser.cfg file: # Debug level on logging debug=3 # memory debug log level memdbg=10 # memory statistics log level memlog=10 # Fork and create children (no=run single process for debugging purposes) fork=no # Log to stderr (only use this of fork=no) log_stderror=yes # Log to the defined syslog facility log_facility=LOG_LOCAL0 # IP address to listen to listen=192.168.0.182 # Port to listen on port=5060 # Aliases used by uri!=myself checks to identity locally destined messages alias=mytestser.org # Number of children to fork per port per protocol children=4 # Is this host multihomed? mhomed=no # Should dns resolving be used? dns=no # Should reverse dns resolving be used? rev_dns=no # Should reverse dns resolving be used? disable_core_dump=yes # *************Your local directives ***************** # *** Common # *** Specific # Load the stateless transaction module (sl_reply etc) loadmodule "/usr/local/lib/ser/modules/sl.so" # Load the transaction stateful module (t_relay() etc) loadmodule "/usr/local/lib/ser/modules/tm.so" # Load the record routing module. Needed for making sure messages go through our server loadmodule "/usr/local/lib/ser/modules/rr.so" # Load maxfwd for stopping loops loadmodule "/usr/local/lib/ser/modules/maxfwd.so" # Load the user location module to store and retrieve user locations loadmodule "/usr/local/lib/ser/modules/usrloc.so" # Load the registrar module to handle registrations loadmodule "/usr/local/lib/ser/modules/registrar.so" # Load the xlog module for advanced logging with parameters loadmodule "/usr/local/lib/ser/modules/xlog.so" # Load the ctl module for the control socket interface. loadmodule "/usr/local/lib/ser/modules/ctl.so" # Load the gflags, global flags module. loadmodule "/usr/local/lib/ser/modules/gflags.so" # Load textops used for searching/selecting text in messages and manipulating headers. loadmodule "/usr/local/lib/ser/modules/textops.so" # Load the sanity check module to do integrity checks on incoming messages. loadmodule "/usr/local/lib/ser/modules/sanity.so" # Load the mysql connector module. loadmodule "/usr/local/lib/ser/modules/mysql.so" # Load the auth module with all the core authentication functions. loadmodule "/usr/local/lib/ser/modules/auth.so" # Load the auth module with the database functions. loadmodule "/usr/local/lib/ser/modules/auth_db.so" # The uri_db module uses database to check the uri. loadmodule "/usr/local/lib/ser/modules/uri_db.so" # The avp_db module allows loading of user attributes from database. loadmodule "/usr/local/lib/ser/modules/avp_db.so" loadmodule "/usr/local/lib/ser/modules/avp.so" # Load domain module used to verify the local domains. loadmodule "/usr/local/lib/ser/modules/domain.so" # *** Your loaded modules # Flags set be various modules # We set database mode to 0 = do not save to database backend (keep in memory), 1 = write-through to database modparam("usrloc", "db_mode", 1) # ctl module params # by default ctl listens on unixs:/tmp/ser_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") # We turn on full lr (adds lr=on as uri parameter, not lr). # This should have higher compatibility with some user agents. modparam("rr", "enable_full_lr", 1) # Encrypt dialog cookies with this secret modparam("rr", "cookie_secret", "ChangeThisIfYouStoreSecretStuffInCookie") # The authentication related modules and which database to use. modparam("auth_db|usrloc|uri_db", "db_url", "mysql://ser:heslo@localhost/ser") # We can store some of our important attributes (inside the messages) avpflags dialog_cookie; # ************* Your local module parameters ***************** # *** Common # *** Specific # Main route, executed for all messages route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ sanity_check(); # Check whether we might be in a loop. if (!mf_process_maxfwd_header("10")) { sl_reply("483", "Too Many Hops"); exit; } if (msg:len > max_len) { sl_reply("513", "Message Overflow"); exit; } # Log all incoming messages to see start of processing if(is_gflag("29")){ xlog("L_NOTICE","INCOMING - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } # ------------------------------------------------------------------------ # Handling messages with Route headers (aka loose routing) # ------------------------------------------------------------------------ if (loose_route()) { if(is_gflag("31")){ xlog("L_NOTICE"," loose routed - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } route(RELAY); exit; } # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ else if (method!="REGISTER") { record_route(); }; # If the ACK has more hops before reaching the user agent, it will have a Route header # and be forwarded above. If next hop is a user agent, the Request URI will be the ip:port # found in Contact of the original OK from the user agent, but will not be loose routed. # Just relay the ACK to the user agent. if (method=="ACK") { if(is_gflag("31")){ xlog("L_NOTICE"," ACK relayed using ruri - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } route(RELAY); exit; } # check if the caller is from a local domain (f=from, d=domain, $f will be set) lookup_domain("$fd", "@from.uri.host"); # check if the callee is at a local domain (t=to, d=domain, $t will be set) # If we have no To tag (dialog creating messages), we use request uri, but # but if this is an in-dialog message (like BYE with last hop before UA here) # we use the To header. if((a)to.tag=="") { lookup_domain("$td", "@ruri.host"); } else { lookup_domain("$td", "@to.uri.host"); } # we dont know the domain of the caller and also not # the domain of the callee -> someone uses our proxy as # a relay if (!$t.did && !$f.did) { sl_reply("403", "Relaying Forbidden"); exit; } # ------------------------------------------------------------------------ # Message Type Processing Section # NOTE!!! It is assumed that all messages reaching this point are destined # for the local server (REGISTER) or for UAs registered locally. (see nonlocal.m4) # ------------------------------------------------------------------------ if (method=="REGISTER") { route(REGISTER); exit; } # Authenticate route(AUTHENTICATE); # All local, non-ACKs, non-REGISTER are assumed to be part of a potential # new dialog (standalone MESSAGE and NOTIFYs without dialog-creation will ex. also reach here) route(NEWDIALOG); # By now, the request uri should be correct, relay if(is_gflag("29")){ xlog("L_NOTICE","No route matched for the request\nCall-Id: %ci\n"); }; sl_reply("404", "No route matched"); } # end main route route[RELAY] { # Log all outgoing messages if(is_gflag("29")){ xlog("L_NOTICE","OUTGOING - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ # t_relay will act based on the current request URI, as well as any # overruling received parameters (i.e. for NATed user agents) # UDP to TCP and vice versa works as well. if (!t_relay()) { sl_reply_error(); }; exit; } route[REGISTER] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ # From lookup_domain, $t.did should be set if we found the domain as ours. if (!$t.did) { if(is_gflag("29")){ xlog("L_NOTICE","%rm From <%fu> Attempted REGISTER from non-local domain\nCall-ID: %ci\n"); }; sl_reply("403", "Attempting to register with unknown domain"); exit; } # We check that Authorization header in the message contains valid credentials found in the # table with the same name. if (!www_authenticate("$fd.digest_realm", "credentials")) { # $? represents the return error code from www_authenticate() if ($? == -2) { if(is_gflag("31")){ xlog("L_NOTICE"," REGISTER - %rm From <%fu> 500 Internal Server Error\nCall-ID: %ci\n"); }; sl_reply("500", "Internal Server Error"); } else if ($? == -3) { if(is_gflag("31")){ xlog("L_NOTICE"," REGISTER - %rm From <%fu> 400 Bad Request\n"); }; sl_reply("400", "Bad Request"); } else { # After www_authenticate(), $digest_challenge should be set if no credentials were found. # We append the challenge to reply and send 401 Unauthorized. if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("401", "Unauthorized"); } # We didn't have a valid set of credentials, so we exit. exit; } # www_authenticate set $uid, so let's store it. $authuid=$uid; # Check if the authenticated user is the same as the target user. # We use the uri in the To field to lookup the user's uid. If none is found, the To user is unknown. if (!lookup_user("$tu.uid", "@to.uri")) { sl_reply("404", "Trying to register unknon user (To)"); exit; } # $f.uid set by www_authenticate() should match the uid we should looked up if ($authuid != $tu.uid) { sl_reply("403", "Authentication and To-Header mismatch"); exit; } # We use the uri in From to look up the uid. if (!lookup_user("$fu.uid", "@from.uri")) { sl_reply("404", "Expect a known user in From"); exit; } # The looked up $fu.uid is checked against the Authentication user if ($authuid != $fu.uid) { sl_reply("403", "Authentication and From-Header mismatch"); exit; } if (!save_contacts("location")) { sl_reply("400", "Invalid REGISTER Request"); exit; }; exit; } # End route[REGISTER] route[AUTHENTICATE] { # RFC3261 says you are not allowed to challenge these methods if (method=="CANCEL" || method=="ACK") { return; } # We check that Authorization header in the message contains valid credentials found in the # table with the same name. if (!proxy_authenticate("$fd.digest_realm", "credentials")) { # $? represents the return error code from proxy_authenticate() if ($? == -2) { if(is_gflag("31")){ xlog("L_NOTICE"," AUTHENTICATE - %rm From <%fu> 500 Internal Server Error\nCall-ID: %ci\n"); }; sl_reply("500", "Internal Server Error"); } else if ($? == -3) { if(is_gflag("31")){ xlog("L_NOTICE"," AUTHENTICATE - %rm From <%fu> 400 Bad Request\n"); }; sl_reply("400", "Bad Request"); } else { # After www_authenticate(), $digest_challenge should be set if no credentials were found. # We append the challenge to reply and send 407 Proxy Authentication Required. if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("407", "Proxy Authentication Required"); } # We didn't have a valid set of credentials, so we exit. exit; } # proxy_authenticate set $uid, so let's store it. $authuid=$uid; # Use the uri in the From header and look up the uid. If not found, delete the uri attribute. if (!lookup_user("$fu.uid", "@from.uri")) { del_attr("$uid"); } # Check the found uid against the authenticated uid. if ($fu.uid != $authuid) { sl_reply("403", "Fake Identity"); exit; } # load the user attributes (preferences) of the caller. $fu.yourattr should now be set if set in the # user preferences table. load_attrs("$fu", "$f.uid"); } route[NEWDIALOG] { # If we don't know the user, it is not for a local user. if (!lookup_user("$tu.uid", "@ruri")) { # User is not known and request uri does not match PSTN dialplan. return; } if (!lookup_contacts("location")) { if(is_gflag("30")){ xlog("L_NOTICE"," NEWDIALOG - 480 User Offline\nCall-Id: %ci\n"); }; sl_reply("480", "User Offline"); exit; } # Relay it to the location(s) of the found user route(RELAY); exit; } # End route[NEWDIALOG]

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[OpenSER-Users] Res: Openser with freeradius not accounting BYE method
by Flavio Goncalves 05 Apr '08

05 Apr '08

Set the flag 2 and 3 in the loose route for BYE requests. if (loose_route()) { if (is_method("BYE")) { setflag(2); setflag(3); }; }; ----- Mensagem original ---- De: MagoDaViola <magodaviola(a)ig.com.br> Para: users(a)lists.openser.org Enviadas: Sábado, 5 de Abril de 2008 16:50:56 Assunto: [OpenSER-Users] Openser with freeradius not accounting BYE method I´m has a problem that my openser sometimes not accounting bye method and this cause a call without pay. See the openser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) ##-- # # Uncomment these lines to daemonize fork=yes log_stderror=no # (cmd line: -E) ##-- # # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes ##-- alias=sip.processavoip.net listen=201.32.70.20 #syn_branch=1 memlog=3 sip_warning=1 server_signature=yes reply_to_via=0 mhomed=0 disable_dns_blacklist=yes unix_sock="/tmp/openser.sock" check_via=0 # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 mpath="/usr/local/lib/openser/modules/" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "uri.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "acc.so" loadmodule "xlog.so" loadmodule "dbtext.so" loadmodule "uri_db.so" loadmodule "mi_fifo.so" loadmodule "group.so" # ----------------- setting module-specific parameters --------------- # -- mi-fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("usrloc", "domain_column", "fk_domains") modparam("usrloc", "user_column", "fk_accounts") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ## -- database params -- modparam("acc|usrloc|group|domain|uri_db|permissions", "db_url", "mysql://root:master@localhost/openser") # -- radius params -- modparam("acc|auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- radius permissions -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- group params -- modparam("group", "table", "accounts_groups") modparam("group", "user_column", "fk_accounts") modparam("group", "domain_column", "fk_domains") modparam("group", "group_column", "pk_accounts_groups") modparam("group", "use_domain", 0) ## -- acc params -- modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_flag", 2) #modparam("acc", "radius_extra", "via=$hdr(Via[*]); email=$avp(s:email)") modparam("acc", "log_level", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "radius_extra", "User-Name=$fU;Suffix=$fd;Sip-Translated-Request-URI=$ru") modparam("acc", "service_type", 15) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_flag", 1) ## -- acc params without radius -- #modparam("acc", "db_missed_flag", 3) #modparam("acc", "db_flag", 2) #modparam("acc", "db_table_acc", "call_history") #modparam("acc", "db_table_missed_calls", "call_history_missed") # -- uri_db params -- modparam("uri_db", "subscriber_table", "accounts") modparam("uri_db", "subscriber_domain_column", "fk_domains") modparam("uri_db", "subscriber_user_column", "pk_accounts") # -- domain params -- modparam("domain", "domain_table", "domains") modparam("domain", "domain_col", "pk_domains") # -- nathelper, mediaproxy nd registrar params -- modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # to configure nat at openser 1.2, 1.3 or 1.4 modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-clients") #modparam("registrar", "received_avp", "$avp(s:rcv)") # to configure nat at openser 1.2, 1.3 or 1.4 # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others #modparam("tm", "fr_inv_timer", 15 ) #Tempo maximo do ring antes de atender #modparam("tm", "fr_timer", 1) #modparam("tm", "wt_timer", 1) # ------------------------- request routing logic ------------------- # main routing logic route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Muitos Hops"); log(1, "486: Muitos Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message to big"); log(1, "513: Message too bi \n"); exit; }; ##-- # Eh feita a gravacao de todas as mensagens para se certificar que todas as mensagens # passem pelo proxy o que eh importante particularmente se as entidades estao usando # um protocolo de transporte diferente ##-- if (method!="REGISTER") { record_route(); }; route(6); ##-- # Secao de finalizacao da chamada ##-- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; ##-- # mensagens subsequentes dentro de um dialogo deveriam tomar um caminho pre-determinado # pela gravacao das rotas (record route) ##-- if (loose_route()) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { xlog("L_ERR", "Main Route: Fixing SDP to $fu $rm (loose route)\n"); if (client_nat_test("3") || isflagset(5)) { setbflag(6); use_media_proxy(); }; }; xlog("L_ERR", "Main Route: Loose Route relaying\n"); route(1); exit; }; if (!uri==myself) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); xlog("L_ERR", "Main Route: Relaing mesage to proxy destination \n"); route(4); route(1); }; ##-- # Verifica baseado na tabela de dominios se a parte host da R-URI comtem um dos dominios # quais o proxy eh responsavel ##-- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbinden..."); } exit; }; ##-- # A rotina abixo eh chamada para marcar o registro com NAT ##-- if (method=="CANCEL") { route(1); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="INVITE") { route(3); exit; }; ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); exit; }; ##-- # Verifica se encontra o usuario na base de localizacao ##-- if (!lookup("location")) { sl_send_reply("404", "Usuario nao encontrado"); exit; }; route(1); exit; } route[1] { ##-- # Envie isto agora; use "statefull fowarding" pois funciona de forma confiavel de UDP para TCP # Ativa a contabilizacao das chamadas ##-- xlog("L_ERR", "Route[1]: Relaying message\r\n"); setflag(2); ##-- # Ativa a contabilizacao das chamadas perdidas ##-- setflag(3); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ##-- # Tratamento padrao para as mensagens de Register ##-- sl_send_reply("100", "Registrando...."); xlog("L_ERR", "Route[2]: Trying Auth: $fu $si\n"); if (isflagset(5) && client_nat_test("7")) { xlog("L_ERR", "Route[2]: Client behind NAT - flag(6)\n"); setbflag(6); fix_nated_register(); force_rport(); # se quiser usar natping options descomente a linha abaixo # setbflag(7); }; if (!radius_www_authorize("")) { www_challenge("", "0"); xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; }; if (!check_to()) { xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; } else { xlog("L_ERR", "Route[2]: Authenticate: $fu $si\r\n"); }; consume_credentials(); if (!save("location")) { xlog("L_ERR", "Route[2]: Can't save register location\r\n"); sl_reply_error(); }; } route[3] { ##-- # Tratamento padrao para as mensagens de INVITE ##-- xlog("L_ERR", "Route[3]: INVITE: $fu -> $tu $ci\r\n"); # setflag(1); if (!allow_trusted()) { if (!radius_proxy_authorize("")) { proxy_challenge("", "0"); # Realm will be autogenerated exit; } else if (!check_from()) { sl_send_reply("403", "Usue from=ID"); xlog("L_ERR", "Route[3]: Error 403 $fu -> $tu\r\n"); exit; }; } consume_credentials(); ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); }; ##-- # VoIP Local ##-- if (uri=~"sip:[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao VoIP local: $fU -> $tu\n"); } else { ##-- # Nacional ##-- if (uri=~"sip:0[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Nacional: $fU -> $tu\n"); } else { ##-- # Internacional ##-- if (uri=~"sip:00[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Interacional: $fU -> $tu\n"); }; }; route(5); exit; }; if (!lookup("location")) { append_hf("P-hint: usuario nao encontrado\r\n"); sl_send_reply("404", "Usuario nao encontrado"); exit; }; append_hf("P_hint: usrloc applied\r\n"); t_on_failure("1"); route(4); route(1); } route[4] { ##-- # Secao de travessia do nat ##-- if (isflagset(5)) { setbflag(6); }; if (isflagset(5) || isbflagset(6) || isbflagset(7)) { use_media_proxy(); xlog("L_ERR", "Rota[4]: Travessia de NAT mediaproxy\n"); }; } route[5] { ##-- # Encaminhamento para a rede publica (PSTN) ##-- xlog("L_ERR", "Route[5]: Relaying to PSTN\n"); rewritehostport("201.32.70.21"); route(4); route(1); } route[6] { force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } failure_route[1] { if (t_check_status("487")) { exit; }; if (isbflagset(6) || isflagset(5)) { end_media_session(); }; onreply_route[1] { ##-- # Bloco de tratamento de respostas ##-- if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { use_media_proxy(); # search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } if (isbflagset(6)) { fix_nated_contact(); } } Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. MSN: a_schleder(a)hotmail.com Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular+ 55 54 9966 7591 alcindo(a)processa.org Fone+55 54 3286-1738comercial(a)processa.org -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

2 1

[OpenSER-Users] Openser with freeradius not accounting BYE method
by MagoDaViola 05 Apr '08

05 Apr '08

Untitled Document I´m has a problem that my openser sometimes not accounting bye method and this cause a call without pay. See the openser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) ##-- # # Uncomment these lines to daemonize fork=yes log_stderror=no # (cmd line: -E) ##-- # # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes ##-- alias=sip.processavoip.net listen=201.32.70.20 #syn_branch=1 memlog=3 sip_warning=1 server_signature=yes reply_to_via=0 mhomed=0 disable_dns_blacklist=yes unix_sock="/tmp/openser.sock" check_via=0 # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 mpath="/usr/local/lib/openser/modules/" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "uri.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "acc.so" loadmodule "xlog.so" loadmodule "dbtext.so" loadmodule "uri_db.so" loadmodule "mi_fifo.so" loadmodule "group.so" # ----------------- setting module-specific parameters --------------- # -- mi-fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("usrloc", "domain_column", "fk_domains") modparam("usrloc", "user_column", "fk_accounts") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ## -- database params -- modparam("acc|usrloc|group|domain|uri_db|permissions", "db_url", "mysql://root:master@localhost/openser") # -- radius params -- modparam("acc|auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- radius permissions -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- group params -- modparam("group", "table", "accounts_groups") modparam("group", "user_column", "fk_accounts") modparam("group", "domain_column", "fk_domains") modparam("group", "group_column", "pk_accounts_groups") modparam("group", "use_domain", 0) ## -- acc params -- modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_flag", 2) #modparam("acc", "radius_extra", "via=$hdr(Via[*]); email=$avp(s:email)") modparam("acc", "log_level", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "radius_extra", "User-Name=$fU;Suffix=$fd;Sip-Translated-Request-URI=$ru") modparam("acc", "service_type", 15) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_flag", 1) ## -- acc params without radius -- #modparam("acc", "db_missed_flag", 3) #modparam("acc", "db_flag", 2) #modparam("acc", "db_table_acc", "call_history") #modparam("acc", "db_table_missed_calls", "call_history_missed") # -- uri_db params -- modparam("uri_db", "subscriber_table", "accounts") modparam("uri_db", "subscriber_domain_column", "fk_domains") modparam("uri_db", "subscriber_user_column", "pk_accounts") # -- domain params -- modparam("domain", "domain_table", "domains") modparam("domain", "domain_col", "pk_domains") # -- nathelper, mediaproxy nd registrar params -- modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # to configure nat at openser 1.2, 1.3 or 1.4 modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-clients") #modparam("registrar", "received_avp", "$avp(s:rcv)") # to configure nat at openser 1.2, 1.3 or 1.4 # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others #modparam("tm", "fr_inv_timer", 15 ) #Tempo maximo do ring antes de atender #modparam("tm", "fr_timer", 1) #modparam("tm", "wt_timer", 1) # ------------------------- request routing logic ------------------- # main routing logic route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Muitos Hops"); log(1, "486: Muitos Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message to big"); log(1, "513: Message too bi \n"); exit; }; ##-- # Eh feita a gravacao de todas as mensagens para se certificar que todas as mensagens # passem pelo proxy o que eh importante particularmente se as entidades estao usando # um protocolo de transporte diferente ##-- if (method!="REGISTER") { record_route(); }; route(6); ##-- # Secao de finalizacao da chamada ##-- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; ##-- # mensagens subsequentes dentro de um dialogo deveriam tomar um caminho pre-determinado # pela gravacao das rotas (record route) ##-- if (loose_route()) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { xlog("L_ERR", "Main Route: Fixing SDP to $fu $rm (loose route)\n"); if (client_nat_test("3") || isflagset(5)) { setbflag(6); use_media_proxy(); }; }; xlog("L_ERR", "Main Route: Loose Route relaying\n"); route(1); exit; }; if (!uri==myself) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); xlog("L_ERR", "Main Route: Relaing mesage to proxy destination \n"); route(4); route(1); }; ##-- # Verifica baseado na tabela de dominios se a parte host da R-URI comtem um dos dominios # quais o proxy eh responsavel ##-- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbinden..."); } exit; }; ##-- # A rotina abixo eh chamada para marcar o registro com NAT ##-- if (method=="CANCEL") { route(1); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="INVITE") { route(3); exit; }; ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); exit; }; ##-- # Verifica se encontra o usuario na base de localizacao ##-- if (!lookup("location")) { sl_send_reply("404", "Usuario nao encontrado"); exit; }; route(1); exit; } route[1] { ##-- # Envie isto agora; use "statefull fowarding" pois funciona de forma confiavel de UDP para TCP # Ativa a contabilizacao das chamadas ##-- xlog("L_ERR", "Route[1]: Relaying message\r\n"); setflag(2); ##-- # Ativa a contabilizacao das chamadas perdidas ##-- setflag(3); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ##-- # Tratamento padrao para as mensagens de Register ##-- sl_send_reply("100", "Registrando...."); xlog("L_ERR", "Route[2]: Trying Auth: $fu $si\n"); if (isflagset(5) && client_nat_test("7")) { xlog("L_ERR", "Route[2]: Client behind NAT - flag(6)\n"); setbflag(6); fix_nated_register(); force_rport(); # se quiser usar natping options descomente a linha abaixo # setbflag(7); }; if (!radius_www_authorize("")) { www_challenge("", "0"); xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; }; if (!check_to()) { xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; } else { xlog("L_ERR", "Route[2]: Authenticate: $fu $si\r\n"); }; consume_credentials(); if (!save("location")) { xlog("L_ERR", "Route[2]: Can't save register location\r\n"); sl_reply_error(); }; } route[3] { ##-- # Tratamento padrao para as mensagens de INVITE ##-- xlog("L_ERR", "Route[3]: INVITE: $fu -> $tu $ci\r\n"); # setflag(1); if (!allow_trusted()) { if (!radius_proxy_authorize("")) { proxy_challenge("", "0"); # Realm will be autogenerated exit; } else if (!check_from()) { sl_send_reply("403", "Usue from=ID"); xlog("L_ERR", "Route[3]: Error 403 $fu -> $tu\r\n"); exit; }; } consume_credentials(); ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); }; ##-- # VoIP Local ##-- if (uri=~"sip:[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao VoIP local: $fU -> $tu\n"); } else { ##-- # Nacional ##-- if (uri=~"sip:0[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Nacional: $fU -> $tu\n"); } else { ##-- # Internacional ##-- if (uri=~"sip:00[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Interacional: $fU -> $tu\n"); }; }; route(5); exit; }; if (!lookup("location")) { append_hf("P-hint: usuario nao encontrado\r\n"); sl_send_reply("404", "Usuario nao encontrado"); exit; }; append_hf("P_hint: usrloc applied\r\n"); t_on_failure("1"); route(4); route(1); } route[4] { ##-- # Secao de travessia do nat ##-- if (isflagset(5)) { setbflag(6); }; if (isflagset(5) || isbflagset(6) || isbflagset(7)) { use_media_proxy(); xlog("L_ERR", "Rota[4]: Travessia de NAT mediaproxy\n"); }; } route[5] { ##-- # Encaminhamento para a rede publica (PSTN) ##-- xlog("L_ERR", "Route[5]: Relaying to PSTN\n"); rewritehostport("201.32.70.21"); route(4); route(1); } route[6] { force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } failure_route[1] { if (t_check_status("487")) { exit; }; if (isbflagset(6) || isflagset(5)) { end_media_session(); }; onreply_route[1] { ##-- # Bloco de tratamento de respostas ##-- if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { use_media_proxy(); # search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } if (isbflagset(6)) { fix_nated_contact(); } } -------------------------------------------------------------------------- Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. a.. MSN: a_schleder(a)hotmail.com b.. Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular + 55 54 9966 7591 alcindo(a)processa.org Fone +55 54 3286-1738 comercial(a)processa.org

2 1

[OpenSER-Users] How to modify the "Referred-by" field inside a REFER
by Amelie CAMELIERE 04 Apr '08

04 Apr '08

Hi everybody, Is it possible to modify the "Referred-by" field inside a REFER SIP message? In fact I want to modify the name displayed by the phone when I use a REFER. Thanks for your help Amélie

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[OpenSER-Users] Openser with RTPProxy in a different scenario
by morton.olson 04 Apr '08

04 Apr '08

Hi all! I have a problem with Openser an RTPProxy. I am using nathelper module to establish a statefull connection with RTPProxy in between the both clients. I player around with it now for a while but I cant get it to work. All the examples in the internet a for clients behind NAT, etc... Maybe one of you might help me. I am using Openser 1.3.0 with RTPProxy 20040107 on Debian Linux "Etch". My scenario looks like this: UA(public ip) ----- Firewall (public IP; 192.168.0.1/24) ---- Openser+RTPProxy (192.168.0.5/24) ----- UA (192.168.0.23/24) The Firewall forward Port 5060 to the Openser which is listening on 5060 as well. Also Ports from 60000-61000 are forwarded to the Openser aswell. (RTPProxy was compile with these Ports) I have no access to the firewall and it does not manipulate SIP Messages. Is there a chance to get this to work? If so, I willl post my configuration file to see what I did wrong. Thanks a lot. Greetz Martin -- View this message in context: http://www.nabble.com/Openser-with-RTPProxy-in-a-different-scenario-tp16349… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[Serusers] Asterisk + OpenSER integration successful
by satish patel 04 Apr '08

04 Apr '08

Dear ser guys.. I have done asterisk with openser integration after spending 2 and 3 days...not time to get nice cold beer... I am user openser front of Asterisk. All sip client register on OpenSER server and call to local extension without Asterisk. caz Asterisk is a media server which are useing only for PSTN calling. When any sip client dialing outside number means mobile number or any outside number Openser forward call to Asterisk and asterisk forward it on PSTN use the E1 or FXO line. ( In my case it is E1 ). OpenSER only user for SIP users or SIP proxy server it can only handle sip request. Asterisk is Media server which can handle IVR , confrance , VoiceMail, and Meetmee or PSTN application. when call come down from PSTN Asterisk forward it on OpenSER as per dialplan configuration. It can handle 1000 of call and you can use it on large production. [sip_client]------------->[Openser]--------->[Asterisk]----[PSTN] My Openser configuration file. #cat ser.cfg debug=6 fork=yes log_stderror=yes listen=<openser_ipaddress> # INSERT YOUR IP ADDRESS HERE port=5060 children=4 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # requests for Media server if(method=="INVITE") { if (uri =~ "sip:0[1-9]@*"){ route(3); break; } } lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3]{ if (uri =~ "sip:0[1-9]@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("<asterisk_ip>:5060"); route(1); break; } } #---------END------------# #Asterisk extension.conf [globals] SERADDRESS=203.124.16.109 [outgoing] exten => 5500,1,Playback(hello-world) exten => 5000,1,Answer() exten => 5000,2,Dial(SIP/5000) exten => 5000,3,Hangup() ; XXXX use to forward call on openser extensions. exten => _XXXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Explore your hobbies and interests. Click here to begin.

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[Serusers] How to set a global variable in ser.cfg and use it in following route() functions?
by lin9k＠yahoo.com 04 Apr '08

04 Apr '08

Hi, Is ser.cfg allow such feature? I'm thinking to use it to hold some variable that can be used for the life of ser.cfg. Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[Serusers] calling rewriteuser and rewriteuri from a module?
by lin9k＠yahoo.com 04 Apr '08

04 Apr '08

Hi, I want to provide a module of our own. How do I call rewriteuser and rewriteuri functions from a module? Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] 1.3.x package?? forLinux??
by Rey Mulero 04 Apr '08

04 Apr '08

Hello, I am new to OpenSER. I wanted to know if there was a package already compiled to support a P4 Intel machine. I was looking in the website but it really does not spell it out. I see file names openser-1.3.0-tls_linux_i386.tar.gz and openser-1.3.0-notls_linux_i386.tar.gz I am wondering what the difference is? What does tls or notls stand for? Your help is appreciated. Thank you Rey ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[Serusers] Asterisk + SER call forwading
by satish patel 04 Apr '08

04 Apr '08

Dear all I want to achive forwading call from SER to Asterisk system i have read about it on voip-info real-time integration of SER & asterisk but its base on mysql i dont have mysql database server i just want to use asterisk as PSTN media means all SIP register on SER and "0XXXXXXXXXXX" number goes thorugh asterisk Ser and asterisk runing on defferent machine. I have put this code in my ser.cfg.sample file.. [globals] SERADDRESS=XXX.XXX.XXX.XXX:5060 [context] exten => <yourexten>,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) In ser.cfg: if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("XXX.XXX.XXX.XXX:5060"); t_relay(); break; } } But it is not working u got error "User not Found" i dont know way this happending... $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

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[OpenSER-Users] re : Cert advisory and OpenSER V1.3.1
by Jignesh Gandhi 03 Apr '08

03 Apr '08

Hello, I recently upgraded from SER.8.14 to OpenSER 1.3.1 and had a quick question regarding the SIP CERT advisory suite of tests. Has OpenSER been tested against the SIP CERT advisory suite of test cases, that can be downloaded from http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/. This is the SIP CERT website http://www.cert.org/advisories/CA-2003-06.html thanks, Jignesh Gandhi Movius Interactive Corp

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[Serusers] SER for NAT client
by satish patel 03 Apr '08

03 Apr '08

Dear all I have a bit confusen about server and nat its stupid question but i want to clear it wil your suggestions.. I am going to impliment SER for my customers with domain base authentication. Now if my client behind the NAT then how my RTP make connection between two IP phone [SER] / \ / \ / \ [A] [B] In this case RTP between [A] to [B] right But when both [A] and [B] behind the same NAT router then what about RTP what is the path ??? What is the pass of RTP between [A] and [B] in this case [SER] | | | [NAT] / \ / \ [A] [B] $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Now you can chat without downloading messenger. Click here to know how.

2 3

[Serusers] SER and ptime ?
by Mike Trest - Personal 03 Apr '08

03 Apr '08

Hi, I have registering phones (Linksys) that use g711 or g729. I have a media question in a situation when I attempt to do pure-proxy without media gateway or proxy. Because these user are always messing with their adapter settings, I have variable call setup when I forward certain destinaitons because of ptime miss-match between Linksys and destination. Is it possible to accpet / reject / revise the inbound call when the "ptime" is not specifically 20ms or 30ms (or any specific value) before t_relay on to destination? I see the ptime value in SDP but before I embark on journey trying to "manage" call setup based on that, I want to know if others are working with this issue. ..mike..

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[OpenSER-Users] [Fwd: contact uri]
by Amit Vijayvargiya 03 Apr '08

03 Apr '08

2 1

[Serusers] How to return a string value of an exported functions from a module to be used in ser.cfg?
by lin9k＠yahoo.com 03 Apr '08

03 Apr '08

Hi, I am thinking to write a module and export a function to be called from ser.cfg. I need the exported function to return a string value to be used by another function in ser.cfg. How do I do that in ser.cfg? For example: in ser.cfg: function1(my_function()); in my_module: str *my_function(){ ... return my_string; } Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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[OpenSER-Users] Res: My first OpenSER post
by Flavio Goncalves 02 Apr '08

02 Apr '08

Hi Yakout, Check SerMyAdmin at http://sourceforge.net/projects/sermyadmin. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Yakout Esmat <yesmat(a)xtra.co.nz> Para: users(a)lists.openser.org Enviadas: Terça-feira, 1 de Abril de 2008 16:08:44 Assunto: [OpenSER-Users] My first OpenSER post Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] rtpproxy address filling
by Andres 02 Apr '08

02 Apr '08

Hi, I have a question regarding the way rtpproxy handles 'address filling'. After a session has been created, the rtpproxy pre-fills the caller and callee's addresses and we see that in the rtpproxy output like: pre-filling caller's address with 192.116.246.234:41000 pre-filling callee's address with 192.116.246.234:20000 Then when it sees the actual packets coming in from a different source port, it updates the address and we see it in the log like: callee's address filled in: 192.116.246.234:1024 (RTP) The audio then flows fine both ways. My question is, what would happen it the actual packets came in from a different IP while the rtpproxy was waiting between the state of 'pre-filling' and 'address filled' states? Will the rtpproxy accept such a change that includes a new IP? Will it ignore packets from a different IP and continue the session normally? Or will it abort the session completely? Thanks, Andres http://www.telesip.net

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[Serusers] Fwd: Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Stefan Thanks a lot for your prompt response. I can hear the announcement now. Previously my audio setup was not correct. Best Regards, Frq On Wed, Apr 2, 2008 at 12:11 PM, Stefan Sayer <stefan.sayer(a)iptego.com> wrote: > Hello, > > frq ser wrote: > > > Hi > > > > Thanks for your reply Stefan and my apologies for coming back to this > > issue after such a long time. As you suggested, I included SDP headers in my > > Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the > > announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 > > OK message, but no audio was being played. In the log, I got these messages > > (just including the important messages)... > > > > .. > > (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: > > dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying > > '/usr/local/lib/sems/audio/192.168.3.208/200.wav < > > http://192.168.3.208/200.wav>' (17358) DEBUG: getAnnounceFile > > (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' > > .. > > .. > > (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global > > address: 192.168.3.208 <http://192.168.3.208/> (17358) DEBUG: > > negotiate (AmSession.cpp:262): remote party doesn't support telephone events > > > > Seeing the last line, I then used an Xlite softphone (instead of sipsak) > > to contact SEMS. Again, the call gets established for about 10 seconds and > > then just gets disconnected and no audio can be heard. Nothing in > > > To me the logs look normal. seems like it plays the announcement and then > hangs up, as it should. the default_en.wav (or 200.wav, if you've copied it > there), is only a few seconds long. are you sure that audio is setup > correctly (volume etc), i.e. does the xlite test call work? if you call e.g. > sip:conference@iptel.org <sip%3Aconference(a)iptel.org>, do you hear > something? > > can you run tcpdump -i any or run wireshark to see whether packets flow > between SEMS and xlite? > > you could also PM me a wireshark dump so i could have a look at what's > going on. > > Stefan > >

1 0

Re: [Serusers] Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Thanks for your reply Stefan and my apologies for coming back to this issue after such a long time. As you suggested, I included SDP headers in my Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 OK message, but no audio was being played. In the log, I got these messages (just including the important messages)... .. (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying '/usr/local/lib/sems/audio/192.168.3.208/200.wav' (17358) DEBUG: getAnnounceFile (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' .. .. (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global address: 192.168.3.208 (17358) DEBUG: negotiate (AmSession.cpp:262): remote party doesn't support telephone events Seeing the last line, I then used an Xlite softphone (instead of sipsak) to contact SEMS. Again, the call gets established for about 10 seconds and then just gets disconnected and no audio can be heard. Nothing in the SEMS log (see below) suggests that there is any problem (as far as I can understand). I've checked that the file /usr/local/lib/sems/audio/200.wav is present and I can play it directly in Linux using play command. (The file 200.wav is just a copy of default_en.wav file) I just want to know why the announcement file is not being played and why the call between softphone and SEMS just gets disconnected after a while. I am sure it would be something very basic :( Thanks a lot for your help Frq

2 1

[OpenSER-Users] openser social networking events - April-June 2008
by Daniel-Constantin Mierla 02 Apr '08

02 Apr '08

Hello, in the period of time April - June 2008, there will be a series of openser social networking events organized in different places world wide, during the time some other Open Source or VoIP events take place in the same location. The events are most probably in the form of a dinner, organized together with local representatives of openser community. The events are intended to be rather small size, to allow each one to know the others, in a nice and friendly discussion environment. The primary goal is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to any of these events is free, everybody pays for its expenses. Anyone interested in openser, sip or voip in general is welcome to join. Next events are: - Bucharest, Romania, April 12, 2008, tied up with ROSDEV2008 (http://www.rosdev.ro - Romanian site only) - Orlando, FL, USA, April 22, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.edvina.net) - Barcelona, Spain, May 6, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.avanzada7.com/) - Vienna, Austria, May 23, 2008, on the way to AsteriskTag (http://www.asterisk-tag.org/wiki/Hauptseite) - Berlin, Germany, May 28, 2008, tied up with LinuxTag (http://www.linuxtag.org) - Amsterdam, Netherlands, June 3, 2008, tied up with VoN.x Europe (http://www.von.com/web/index.php) The hour and place for each event will be announced few days before as we know the number of attending people and we make the reservation. Please send email to me to reserve a place, although you are welcome to come at any time and without prior notice, you may find that there is no more available seat to join the table. If you live in the surroundings of these cities, and want to help with organization, you know a nice and quite place suitable for the event, let me know. Cheers, Daniel -- http://www.asipto.com

1 0

[OpenSER-Users] Building Carrierroute Tables
by Douglas Garstang 02 Apr '08

02 Apr '08

So, I was kind of wondering how those using carrierroute where populating their routes. I was hoping to build a cost table that contained the costs for all our routes and all our carriers, and then construct the carrierroute table as a view derived from the cost table. I guess my SQL (and those around me I asked about it) isn't that good, because I haven't been able to come up with a way to do it yet. Doug. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] My first OpenSER post
by Yakout Esmat 01 Apr '08

01 Apr '08

Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM

1 0

[OpenSER-Users] Problem activating auth and auth_db modules using openser 1.3.1
by Pascal Maugeri 01 Apr '08

01 Apr '08

Hi My 1.3.1 server does not start when I activate the modules auth and auth_db modules with the following configuration: loadmodule "auth.so" loadmodule "auth_db.so" [...] modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "db_url", "mysql://openser:openserrw@192.168.1.4 /openser") modparam("auth_db", "load_credentials", "") There is no error in the log file but no other module is loaded whenever I enable auth and auth_db. Is there anything else I should enable to configure these two modules ? When I disable the loading of these two modules everything works fine. Regards, Pascal

3 3

[Serusers] help required disabling DNS lookup
by geethanjali.csg＠wipro.com 01 Apr '08

01 Apr '08

Hi, I am trying to set up calls between 2 domains without using DNS lookup. UA -> domain1 -> domain2 -> UA I am using static forwarding route the all requests. Requests should be routed using this forward logic only. Both DNS lookup and forward logic are happening. Since I have not configured the DNS, an error "Unresolvable Destination" is seen by the UA. Ser log displays errors related to DNS lookup. I don't want the DNS lookup to happen. Requests should be routed using the forward logic only. Can someone help me solve this problem. Regards, Geethanjali C S Wipro Technologies The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

3 3

[OpenSER-Users] [openser 1.3] loading module auth.so stop my server
by Pascal Maugeri 01 Apr '08

01 Apr '08

Hi I'm still unable to have a working server 1.3 that loads the module auth.so. It is weird because there is no error message in my log file also I set the debug level to maximum. But after starting the openser service, it says it is successful but there is no running process (pgrep openser does not show anything, when it does if I remove the load auth.so)! How can I get more debugging info about the loading and initialization of auth ? Can anybody share with me his or her configuration file ? I may miss something in mine: loadmodule "sl.so" [...] loadmodule "auth.so" (I believe it is sufficient). Is there any known issue using Redhat Enterprise 4 ? Regards, Pascal

1 0

[OpenSER-Users] Problem in finding invite tsxn for ACK.
by Vikas CG. 01 Apr '08

01 Apr '08

Hi All, We were using the openser server as reg/proxy to test sip gateway. We observed that if we run openser in TCP mode it will add 2 record-routes one for TCP and UDP. But when we put hold UA-client i.e. when we send reinvite the openser server is adding only one record-routes But in ACK for reinvite it is adding 2 record-routes. I am also attaching the sip flow. It would be grateful if any body tells us is this behavior is correct? Because we are depending on record route to find the invite tsxn for ack. Thanks, Vikas C.G.

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