- sr-users - kamailio.org

[Serusers] Logging problem
by Zac Tolley 08 Nov '05

08 Nov '05

I have a sip server running purely as a proxy, all it does is forward requests onto a remote sip box. I want to log all the requests that go through it, and use this to create billing records, but I can't get anything out of the server. I have setup the logging level to 8, but only get a few messages at startup. I enabled the acc.so module, in hope of creating accounting information that way, and also nothing. I know this must be because all my requests are forwarded, but I should get something surely... Any ideas what I'm doing wrong? debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" ... loadmodule "/usr/local/lib/ser/modules/acc.so" ... # -- acc params -- modparam("acc", "log_level", 8) modparam("acc", "log_flag", 1) modparam("acc", "log_fmt", "mfctios") alias="acme.co.uk" route[0]{ # 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # record transaction using accounting module setflag(1); # Record route all messages to make sure that subsequent messages # go through sip proxy server. record_route(); # forward all requsts to --- forward (192.168.1.2, 5060 ); }

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[Users] OpenSER now in FreeBSD ports tree
by Jesus Rodriguez 08 Nov '05

08 Nov '05

Hello, OpenSER port has been committed in FreeBSD ports tree. The port will also appear as a package in the next packages build. There are two ways to install a port in FreeBSD. Both methods require to update the ports tree (with cvsup or tarball). Once the port tree is updated: If portupgrade tool is installed just do: portupgrade -N openser If portupgrade is not used or you prefer the usual/old method: cd /usr/ports/net/openser make install clean The port allows to select some compile options like TLS support and database support for MySQL and/or Postgres. Some OpenSER packages have been generated for different FreeBSD versions (4.x, 5.x and 6.0). To install the package version just do: pkg_add http://openser.org/pub/openser/latest/packages/FreeBSD/xxxx where xxx is the package version you want to install. Use that URL to know the existing packages. When the package appears in the next official build, you will only need to do: pkg_add -r openser to install the package... easy, right? ;) Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

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[Serusers] ser 0.9.4 radius_auth problem
by Pham Ngoc Khanh 08 Nov '05

08 Nov '05

Dear All, I have just followed the ser radius instructions to configure and install ser-0.9.4 from source in my redhat 9 machine. Whenever I enable the auth_radius module by inserting the following lines in my ser.cfg loadmodule "/usr/local/lib/ser/modules/auth_radius.so" modparam("auth_db", "password_column", "password") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") I got the error "bad config file (3 errors)" in the process of starting ser. But ser will start smoothly when I commented these lines. It takes me a lot of time. Pls help me to resolve this problem. Here is my full ser.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=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=9 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 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_radius.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") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) # -- 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 (!radius_www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); 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(); }; } Thank you. Best regards, Pham Ngoc Khanh

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[Serusers] "copying" the SIP traffic
by Ángel Gómez 08 Nov '05

08 Nov '05

Hi every one, i interested in making SER to send a "copy" of al the trafic than comes to another SIP server. I want SER to act like ever, but in adition send the messages to another server without waitting for a response. Sorry for my english :)

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[Serusers] Not able to download SER from IPTEL site
by rajesh.rajguru＠wipro.com 08 Nov '05

08 Nov '05

Hi, I am trying to access ftp://ftp.berlios.de/pub/ser/latest/ to download SER. After trying for many times also, I am not able to reach this site Has anybody faced this problem ? If yes, what is the solution ? >From where to download latest SER ? Please let me know at your earliest Regards, Rajesh Confidentiality Notice 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 confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

1 0

[Users] Call executing multiple blocks of code
by Brandon Price 08 Nov '05

08 Nov '05

I am having a problem with calls matching a certain pattern executing more than one block of code. I have a block of code that looks like if (method =~ "INVITE" && uri =~ "sip:\*12.*") { ... do some stuff }; and another like if (method=="INVITE" && uri =~ "sip:1[0-9]{10}@.*"){ ... do some stuff }; see full block here . The problem is that whenever *12 is dialed it executes both blocks, I thought the pattern matching is just wrong but when a number matching the second pattern is dialed then it doesn't execute the 1st block. Can someone please explain this behavior to me. --Sorry if this is a repost--

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[Serusers] SER + dispatcher + mediaproxy, and mediaproxy could not start.
by zhangshuai 08 Nov '05

08 Nov '05

Dear all, When we use only one mediaproxy which is on the same machine with SER and disable the dispatcher, this system works perfectly. But now we want to use the dispatcher on the machine A with SER(ip:218.97.252.38), and install a mediaproxy on a separater machine B(ip:218.97.252.40), then mediaproxy just can not start. Call signaling is all OK, but there is no voice. SDP section of invite message is not changed by the mediaproxy. The log file on machine A shows that: Nov 8 16:36:10 b2bua proxydispatcher[5188]: command request 8029CEE9-92A3-4E09-91B5-0F1EE955B343(a)218.97.252.39 218.97.252.39:8000:audio 218.97.252.39 sensates.com remote 203.86.79.190 remote X-Lite=20release=201103a info=from:38001@sensates.com,to:3801@sensates.com,fromtag:614201966,totag: Nov 8 16:36:11 b2bua proxydispatcher[5188]: error: couldn't forward command to 218.97.252.40:25060 (Connection refused) The log file on machine B shows that: Nov 8 11:46:10 localhost mediaproxy[4082]: mediaproxy started. Nov 8 11:46:10 localhost mediaproxy[4082]: Listening for commands on local socket is disabled Nov 8 11:46:10 localhost mediaproxy[4082]: Mediaproxy error: couldn't create command socket (Cannot assign requested address). Nov 8 11:46:10 localhost mediaproxy[4082]: exiting. Nov 8 11:46:10 localhost proxydispatcher[4085]: proxydispatcher started. Nov 8 11:46:10 localhost proxydispatcher[4085]: disabled from configuration Nov 8 11:46:10 localhost proxydispatcher[4085]: exiting. We set mediaproxy.ini (on machine B) as follow: [Dispatcher] start = no socket = /var/run/proxydispatcher.sock group = ser [MediaProxy] start = yes socket = None group = ser listen = 218.97.252.38:25060 allow = 218.97.252.38 proxyIP = 218.97.252.40 portRange = 35000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 accounting = off Thanks a lot.

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[Serusers] Auth_Radius, AVP_Radius, AVOPOS Hex Value Bug
by Lenir 08 Nov '05

08 Nov '05

Hello list, I'm using openser 1.0 and I'm using auth_radius and avp_radius to retrieve all my user's preferences via AVPs. Everything works very well except, that I cannot retrieve AVPs with value of i:0xhex. I'm trying to implement the ACL example in http://www.voice-system.ro/docs/avpops/ar01s08.html In my radius config I have the following: UserName Attribute op Value Sipuser SIP-AVP = #102#0x08 In my openser.cfg I have the following: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway xlog("L_N", " DOMESTIC CALL: from $fU to $rU\n"); if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25052) ERROR:auth_radius:extract_avp: invalid AVP numrical value '0x08' # ------------------------------------------------------------------------ So I tried to cheat, by copying i:102 to a temporary AVP i:302 with the N flag to convert the string to numerical value, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_copy("i:102","i:302/n") && avp_check("i:302", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; This is the debug message that I get from openser: 0(25135) ERROR:avpops:copy_avp: cannot convert str to int # ------------------------------------------------------------------------ And finally, I tried to force it by having the value as string, but try to use the ANDing op, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25140) ERROR:avpops:check_avp: value types don't match # ------------------------------------------------------------------------ It seems to me that the problem lies in auth_radius by not allowing extraction of AVPs with numerical values in hex. If we can get that work work, I think the rest should work fine without any alterations. Thanks for your help in advance! Lenir

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[Users] Auth_Radius, AVP_Radius, AVOPOS Hex Value Bug
by Lenir 08 Nov '05

08 Nov '05

Hello list, I'm using openser 1.0 and I'm using auth_radius and avp_radius to retrieve all my user's preferences via AVPs. Everything works very well except, that I cannot retrieve AVPs with value of i:0xhex. I'm trying to implement the ACL example in http://www.voice-system.ro/docs/avpops/ar01s08.html In my radius config I have the following: UserName Attribute op Value Sipuser SIP-AVP = #102#0x08 In my openser.cfg I have the following: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway xlog("L_N", " DOMESTIC CALL: from $fU to $rU\n"); if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25052) ERROR:auth_radius:extract_avp: invalid AVP numrical value '0x08' # ------------------------------------------------------------------------ So I tried to cheat, by copying i:102 to a temporary AVP i:302 with the N flag to convert the string to numerical value, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_copy("i:102","i:302/n") && avp_check("i:302", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; This is the debug message that I get from openser: 0(25135) ERROR:avpops:copy_avp: cannot convert str to int # ------------------------------------------------------------------------ And finally, I tried to force it by having the value as string, but try to use the ANDing op, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25140) ERROR:avpops:check_avp: value types don't match # ------------------------------------------------------------------------ It seems to me that the problem lies in auth_radius by not allowing extraction of AVPs with numerical values in hex. If we can get that work work, I think the rest should work fine without any alterations. Thanks for your help in advance! Lenir

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[Serusers] Multiple INVITE's in acc table when putting caller on hold
by Bjorn Asmul 08 Nov '05

08 Nov '05

Does anyone have a solution for eliminating multiple INVITE's in the accounting table (acc) as a result of the caller putting someone on hold? Here's my accounting section: ... modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) ... route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (method=="INVITE" || method=="ACK" || method=="BYE" || method=="CANCEL") { setflag(1); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; }

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[Serusers] [OTAnn] Feedback
by shenanigans 08 Nov '05

08 Nov '05

I was interested in getting feedback from current mail group users. We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband. Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast video and social networking such as favorite authors and an html editor. It?s free to join and any feedback would be appreciated. S. -------------------------------------------------------------------------------------------------------------------------------------- Broadband interface (RIA) + mail box saftey = <a href="http://SIP_Router_Users_List.roomity.com">SIP_Router_Users_List.roomity.com</a> *Your* clubs, no sign up to read, ad supported; try broadband internet. ~~1131397872002~~ --------------------------------------------------------------------------------------------------------------------------------------

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[Serusers] problem with rtpproxy
by rajesh volluru 08 Nov '05

08 Nov '05

hi I had installed ser with media and rtp proxy. Below is the output that i get when I start ser. However, iam not able receive any audio transmission when I connect through sip communicator when iam behind NAT. I am able to register with the server. 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Record-Route: 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/nathelper.so fix_nated_sdp 0(0) fixing /usr/local/lib/ser/modules/rr.so record_route 0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain(): Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Route: 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/tm.so t_on_reply 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 2(0) INFO: fifo process starting: 19946 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '57b7f7f7-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): tm 0(19941) DEBUG: callid: '57b7f7f7-19941(a)69.229.23.57' 0(19941) DEBUG: init_mod_child (1): usrloc 0(19941) get_connection(): Connection not found in the pool 1(0) DEBUG: init_mod_child (-1): tm 1(19945) DEBUG: callid: '57b7f7f7-19945(a)69.229.23.57' 1(19945) DEBUG: init_mod_child (-1): usrloc 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): auth_db 0(19941) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): auth_db 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): domain 0(19941) get_connection(): Connection found in the pool 0(19941) DEBUG: init_mod_child (1): nathelper 2(19946) DEBUG: init_mod_child (-2): auth_db 2(19946) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): domain 1(19945) DEBUG: init_mod_child (-1): nathelper 0(19941) rtpp_test: RTP proxy found, support for it enabled 1(19945) rtpp_test: RTP proxy found, support for it enabled 2(19946) DEBUG: init_mod_child (-2): domain 2(19946) DEBUG: init_mod_child (-2): nathelper 2(19946) rtpp_test: RTP proxy found, support for it enabled 2(19946) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(19946) DEBUG: register_fifo_cmd: new command (print) registered 2(19946) DEBUG: register_fifo_cmd: new command (uptime) registered 2(19946) DEBUG: register_fifo_cmd: new command (version) registered 2(19946) DEBUG: register_fifo_cmd: new command (pwd) registered 2(19946) DEBUG: register_fifo_cmd: new command (arg) registered 2(19946) DEBUG: register_fifo_cmd: new command (which) registered 2(19946) DEBUG: register_fifo_cmd: new command (ps) registered 2(19946) DEBUG: register_fifo_cmd: new command (kill) registered 2(19946) WARNING: no fifo_db_url given - fifo DB commands disabled! Let me know if you find any configuration error in SER. Please let me if you need any more information. I am trying to configure SER from long time. Please help me. Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here --------------------------------- Enjoy this Diwali with Y! India Click here

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[Serusers] avp_db_load fails with avp as source
by brett-ser-list 08 Nov '05

08 Nov '05

Hello All, I'm trying (desperately) to use avp_db_load with an avp: version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 12:28:21 Aug 16 2005 with gcc 3.3 modparam("auth_db","avps_column_str","callscope") modparam("avpops","avp_aliases","callgroups=i:200") modparam("avpops","avp_aliases","callscope=i:500") avp_db_load("$callscope","$callgroups/avps") ; In watching the debug, I see the avp loaded just fine from the database from the auth_db parameter: 3(1735) DEBUG:avpops:print_avp: p=0xf51c20d8, flags=3 3(1735) DEBUG: name=<callscope> 3(1735) DEBUG: val_str=<200> but the above avp_db_load line is resulting in: 3(1735) DEBUG:avpops:get_val_as_str: no avp found 3(1735) ERROR:avpops:load_avps: failed to get uuid I also notice that it doesn't even bother to hit the database after the auth_db hit. So unless the avp table has been cached (Which I have no idea if it has..) how would it know? Seems like error #1: no avp found? Seems like it doesn't like my syntax but according to the avpops documentation, it should be right: source = (sip_uri|avp_alias|str_value) ['/'('username'|'domain'|'uri'|'uuid')] note: avp_db_load("$callscope/uuid","$callgroups/avps") ; gives me: 0(0) ERROR:avpops:fixup_db_avp: source/flags "callscope" unknown! Look, what I really want to do is use the authentication credentials to load an avp. Isn't that was avp is for? I don't want the FROM or the TO or the [R]URI, but the actual AUTHentication username. Any ideas?? Thanks!! BTW, I've tried the newest SER DEVEL CVS, and OPENSER 1.0 with basically the same results. Some custom modules I have would also make me want to stick with 0.9.0. -Brett

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[Serusers] New Bee Information
by Toygun Mavinil 07 Nov '05

07 Nov '05

Hi, I want to use voip for a medium size subscriber, i suggest 10.000 i want to; subscribers talk each other with no charge some one can get 2 or 4 subscribing for faks or something else i want voicemail option for people who can not answer calls maybe conference call Call logs for incoming and outgoing Phonebook maybe first i thougt that ser can do this but i dont know if it can i use suse 9.3 /mysql/php on an intel desktop pc (not a server) with p.4 3ghz 1 Gbit DDR2-533 there are some programs i did not understand their roles; Radius? Asteriks? Sim? Which programs i should use to do them? Please give information, Thanks Toygun

1 0

[Serusers] Alias per SIP_DOMAIN? something broken here?
by Gerd Feiner 07 Nov '05

07 Nov '05

hi, i run an SER which supports multiple realms. all works fine, authentication, calls etc. except one thing: we are in the rare sitiuation of having the same number-block in two different cities. i have aliases for my users (username is areacode +number) which is only the number (without area-code) to support dialing without area-code. but in the case with the two same blocks in different area-codes the alias handling gets a bit confused: # serctl alias show 88999998 <sip:0471288999998@04712.sip.domain.de>;q=1;expires=459311264 <sip:0471188999998@04711.sip.domain.de>;q=1;expires=459311264 so i have two aliases for the same number. any of the numbers is in different SIP_DOMAINS, namely 04712.sip... and 04711.sip..., but even when i state the SIP_DOMAIN to use for the lookup stuff i get: # SIP_DOMAIN=04712.sip.domain.de serctl alias show 88999998 <sip:0471288999998@04712.sip.domain.de>;q=1;expires=459311207 <sip:0471188999998@04711.sip.domain.de>;q=1;expires=459311207 and of course the other way round with 04711. the realm/domain is stored within the alias, so there are two records: 88999998 04711.sip.domain.de sip:0471188999998@04711.sip.domain.de ... 88999998 04712.sip.domain.de sip:0471288999998@04712.sip.domain.de ... with different domains. now, what happens is: when a user from the 04711 area-code calls another user (for him in that same area), he/ she might end up reaching a user with the 04712 area-code. even though the INVITE messages and all state, that the particular (any) user is from the 04711 domain, or 04712 for that matter ... i also tried to name the alias 88999998(a)04712.sip... and so on, but then, it doesn't even find an alias at all. what did i miss? oh, and yes, both realms are defined via an alias-directive in SER.cfg so he treats both as local (or at least it should). any hint would be appreciated. brgds, g. feiner

2 2

[Users] Buffer overflow with Xlog
by Jose Maria Gonzalez 07 Nov '05

07 Nov '05

I have installed openser v1.0 and it is working perfectly. But I have a problem with the xlog module. Many times I've got the following error: Nov 7 11:28:34 localhost openser[1736]: xl_printf: buffer overflow -- increase the buffer size... The line that produces this error is this (it is inside a case stament): xlog("L_ERR","Has been selected the 1 index"); I have tried to increase the buffer size in the following form: modparam("xlog","buf_size",8192) No matters the size that I put on the line buf_size, I always get the same error. I thing that a buffer size of 8kb should be enough for this log message, isn't it? Thanks. -- Salu2: *************************************** José María González ***************************************

3 3

[Serusers] Incoming calls from behind NAT
by George Lambson 07 Nov '05

07 Nov '05

Is it possible for SER to accept inbound calls from a user that is not registered and still use the mediaproxy plugin?

1 0

[Users] Are pseudo variables supported in replace()?
by Federico Giannici 07 Nov '05

07 Nov '05

Are pseudo variables sopposed to work in replace() function? The pseudo variables documentation web page says that they are supported only by avpops and xlog, but the OpenSER 1.0.0 announcement web page says that they are supported by textops too... Who is right? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 6

[Serusers] core dump in SER 0.9.4
by Giuseppe Tricarico 07 Nov '05

07 Nov '05

Hi all I installed SER 0.9.4 on Fedora core 4, it works well but if I use include auth_radius module (use radiusclient-ng-0.5.1) it gives a "core dump" ... Does anyone have a similar problem? I tryed to solve the problem by compiling source distribution but nothing changes.... Best Regards Giuseppe

2 3

[Serusers] allow_trusted
by OT geomatics 07 Nov '05

07 Nov '05

I need a way to trust calls coming from a dynamic IP but with a domain-name. In fact, it's an asterisk on a dynamic IP with a dynamic dns. allow_trusted just permit to consider an IP as trustable, is there another way to do that with a domain name? Olivier

3 2

RE: [Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

Bogdan, It also works with -I$(LOCALBASE)/mysql/include/mysql George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, November 07, 2005 3:56 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > > George, > > my mistake - I meant "mysql" module, not "acc" - is any > difference between the Makefiles for this module? what was > the path you had to add? > > regards, > bogdan > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

RE: [Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

The Makefile for mysql module is identical in 0.9.5 and 1.0.0. I just had to add: -I/usr/src/mysql-4.1.10/include in DEFS, because this is where I found MySQL header files. George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, November 07, 2005 3:56 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > > George, > > my mistake - I meant "mysql" module, not "acc" - is any > difference between the Makefiles for this module? what was > the path you had to add? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >The Makefile for mysql module is identical in 0.9.5 and 1.0.0. > >What does the acc module have to do with this? Anyway, the > Makefile for > >acc module in 1.0.0 has > >DEFS+=-DSQL_ACC > >whereas in 0.9.5 it is commented out. > > > >I updated the -I line in modules/mysql and works fine. So I > guess the > >mysql module did not use the MySQL header files in 0.9.5. > > > >thanks a lot > > > >George > > > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Monday, November 07, 2005 2:45 PM > >>To: Papadopoulos Georgios > >>Cc: users(a)openser.org > >>Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > >> > >>Hi George, > >> > >>it seams that the mysql hdr files (devel package) are not found. > >>Please check with your package manager which is the correct path to > >>these files and update the "-I....." in modules/acc/Makefile > >> > >>BTW, is any difference between the makefile for acc module between > >>your > >>0.9.5 and 1.0.0 versions? > >> > >>regards, > >>bogdan > >> > >>Papadopoulos Georgios wrote: > >> > >> > >> > >>>Hello, > >>> > >>>I am trying to upgrade from 0.9.5 to 1.0.0 and I am having trouble > >>>compiling the MySQL module (worked fine with 0.9.5). > >>>I tried both openser-1.0.0_src.tar.gz and > >>> > >>> > >>openser-1.0.0-tls_src.tar.gz, > >> > >> > >>>but the error was the same. > >>>Am I missing something? > >>> > >>>gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -mcpu=ultrasparc > >>>-mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' > >>>-DARCH='"sparc64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' > >>>-D__CPU_sparc64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' > >>>-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 > >>> > >>> > >>-DUSE_MCAST > >> > >> > >>>-DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC > -DFAST_LOCK > >>>-DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 > >>>-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL > >>>-DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include > >>>-I/usr/local/include/mysql -I/usr/local/mysql/include > >>>-I/usr/include/mysql -c db_mod.c -o db_mod.o > >>>db_mod.c:35:19: mysql.h: No such file or directory > >>>db_mod.c: In function `mysql_mod_init': > >>>db_mod.c:87: warning: implicit declaration of function > >>>`mysql_get_client_info' > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>make: *** [db_mod.o] Error 1 > >>> > >>>thank you in advance > >>> > >>>George > >>> > >>> > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

1 0

RE: [Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

The Makefile for mysql module is identical in 0.9.5 and 1.0.0. What does the acc module have to do with this? Anyway, the Makefile for acc module in 1.0.0 has DEFS+=-DSQL_ACC whereas in 0.9.5 it is commented out. I updated the -I line in modules/mysql and works fine. So I guess the mysql module did not use the MySQL header files in 0.9.5. thanks a lot George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, November 07, 2005 2:45 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > > Hi George, > > it seams that the mysql hdr files (devel package) are not > found. Please check with your package manager which is the > correct path to these files and update the "-I....." in > modules/acc/Makefile > > BTW, is any difference between the makefile for acc module > between your > 0.9.5 and 1.0.0 versions? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >Hello, > > > >I am trying to upgrade from 0.9.5 to 1.0.0 and I am having trouble > >compiling the MySQL module (worked fine with 0.9.5). > >I tried both openser-1.0.0_src.tar.gz and > openser-1.0.0-tls_src.tar.gz, > >but the error was the same. > >Am I missing something? > > > >gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -mcpu=ultrasparc > >-mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' > >-DARCH='"sparc64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' > >-D__CPU_sparc64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' > >-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 > -DUSE_MCAST > >-DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK > >-DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 > >-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL > >-DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include > >-I/usr/local/include/mysql -I/usr/local/mysql/include > >-I/usr/include/mysql -c db_mod.c -o db_mod.o > >db_mod.c:35:19: mysql.h: No such file or directory > >db_mod.c: In function `mysql_mod_init': > >db_mod.c:87: warning: implicit declaration of function > >`mysql_get_client_info' > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >make: *** [db_mod.o] Error 1 > > > >thank you in advance > > > >George > > > >Disclaimer > >The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] documentation on nat_uac_test
by Maria Yndefors 07 Nov '05

07 Nov '05

Hi, Where can I find documentation on what the flags to nat_uac_test means? I have checked the admin guide and the nathelper module docs but I could not find anything. Regards /Maria

3 2

[Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

Hello, I am trying to upgrade from 0.9.5 to 1.0.0 and I am having trouble compiling the MySQL module (worked fine with 0.9.5). I tried both openser-1.0.0_src.tar.gz and openser-1.0.0-tls_src.tar.gz, but the error was the same. Am I missing something? gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"sparc64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_sparc64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include -I/usr/include/mysql -c db_mod.c -o db_mod.o db_mod.c:35:19: mysql.h: No such file or directory db_mod.c: In function `mysql_mod_init': db_mod.c:87: warning: implicit declaration of function `mysql_get_client_info' db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) make: *** [db_mod.o] Error 1 thank you in advance George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Users] Access to pseudo variables from within core
by Brandon Price 07 Nov '05

07 Nov '05

I have some questions about access to certain variables from within openser. The documentation states that I can access more of the pseudo variables like $fU and $fu from within more modules and the openser core. I am trying to use $hdr(From[0]) something that I can apparently still only do from within avpops or xlog. I need to have openser perform a certain action only if no SIP Display info exists if it does exist I do not want this action performed. For example.... if ("$hdr(From[0])") =~ "^\<sip:.*") {.... do something } else { do something else... }; Or if there is SIP Display info do A else do B. Any help greatly appreciated.

3 2

[Serusers] gflags
by Iqbal 07 Nov '05

07 Nov '05

Hi Can I just replace flag usage with gflag usage within my config, or are there any caveats Iqbal

3 5

[Serusers] ser+nat+siproxd+asterisk
by jeff kwong 07 Nov '05

07 Nov '05

Hi Guys! I just would like to share that I was able to get a working setup using SER as Softswitch, Asterisk as PSTN gateway and SIPROXD on my NAT Router. SIPROXD is an open source ALG and it effectively handles sip nat traversals. With it I dont have to run a seperate mediaproxy. When making calls from SIP UA to PSTN, RTP is as below: UA---NAT/SIPROXD---ASTERISK for 2 UA behind the same NAT: UA1--NAT---UA2 and for 2 UA behind different NATs: UA1--NAT1----NAT2---UA2 Thus there is less latency on signals and less traffic on SER. My question is, from the experience of other guys here, what do you think is the drawback or advantages of using SIPROXD together with SER to solve SIP NAT issues compared to other methods like using mediaproxy and rtpproxy?Will I still be able to do other SER features like accounting? Thanks! _jeff

8 11

[Serusers] Call logs
by Colin Jordan 07 Nov '05

07 Nov '05

Hi there I can not find where the call logs are in the ser database. What can I do to enable call information logging (time stamps, destination, exc)? I am using mysql. Best regards, Colin

1 0

[Serusers] Is SER a SIP PBX?
by gorand＠dvvti.com 07 Nov '05

07 Nov '05

Hey all you SER Guru's here is a noobie question that may seem dum, yet I am still going to ask it. I know SER is a SIP Proxy and a SIP Registrar and it works great. My question is can you use SER as a full blown SIP PROXY as ASTERISK? Does it handle call waiting? Call Park? Call Transfer? 3 way Conferencing? If so what are the key strokes that activates this in SER? Does SER Support Voicemail? Inquiring minds want to know? Also what is the username and password for Serweb? I tried admin and heslo and it's telling bad username and password on the screen. Thanks. Goran

4 3

[Users] Re: [Fwd: [Serusers] Vertical Service Codes]
by Daryl Sanders 06 Nov '05

06 Nov '05

Ok, so *98 seems to the standard for at least Bell South and Qwest and also works with all my ATAs by default, so now I'm off to tackle Caller-ID Blocking, Message Waiting Indicator (w/asterisk), and Distinctive Ring. Please feel free to share you comment/experiences If you've been down this road already. - Daryl On 11/6/05, Patrick J. Menjor <jmenjor(a)comcast.net> wrote: > *98 > > -------- Original Message -------- > Subject: [Serusers] Vertical Service Codes > Date: Sun, 6 Nov 2005 08:09:13 -0700 > From: Daryl Sanders <daryl.sanders(a)gmail.com> > To: SER Users <serusers(a)iptel.org>, users(a)openser.org > > > > Can SER detect star codes? I see that there are many codes already > defined in most ATAs. How would I make these work with SER? > > Is there a standard code for allowing users to check their voicemail > from their SIP-phone/ATA? I was considering also using 00 as the code > my users dial to access their voicemail? Would there any be any > conflict with 00 and other functionality such as international > calling? There are no DID/DDI that begin with 00 correct? > > Thanks! > - Daryl > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > >

3 2

[Serusers] SER Business List
by Rehan Ahmed 06 Nov '05

06 Nov '05

Hello All, I am wanting to buy some did's from Japan, i wanted to know if there is a Business Mailing list of SER, and if some one can point me to it. Thank You, Rehan

1 0

[Users] Vertical Service Codes
by Daryl Sanders 06 Nov '05

06 Nov '05

Can SER detect star codes? I see that there are many codes already defined in most ATAs. How would I make these work with SER? Is there a standard code for allowing users to check their voicemail from their SIP-phone/ATA? I was considering also using 00 as the code my users dial to access their voicemail? Would there any be any conflict with 00 and other functionality such as international calling? There are no DID/DDI that begin with 00 correct? Thanks! - Daryl

1 0

[Serusers] Clearing fr_inv_timer_avp after failure route
by Evan Borgstrom 06 Nov '05

06 Nov '05

Hey All, I have fr_inv_timer_avp pointing to an AVP named inv_timeout which users are allowed to set and everything is working fine except upon call forwarding from a failure route. Here's the situation... - Call comes in for user(a)domain.com - User at domain.com has inv_timeout set to 10s - User at domain.com also has forward on unavailable to a PSTN number - 10s of ringing to the user elapses - The failure route kicks in and pushes the PSTN URI to a new branch - avp_delete("inv_timeout"); is called - avp_write("i:3600", "inv_timeout"); is called - The PSTN number begins to ring - 10s later SER sends a CANCEL to the PSTN gateway How do I properly clear the users preset inv_timeout and reset it to 1 hour for the PSTN destination? Thanks, Evan

2 2

Re: [Serusers] avp_db_load fails with avp as source
by brett-ser-list 06 Nov '05

06 Nov '05

I keep coming back to that. However, a user can be a member of multiple calling scopes. for example 1)Houston 2)LongDistance 3)Information And I can't figure out how to do it nicely that way.. seems like maybe I'd have to create a seperate column for each callscope? Surely there sure be a different way.. I simply want to load a *list* of avps based on login credentials.. seems simple enough, right? -Brett ---------- Original Message ---------------------------------- From: "Greger V. Teigre" <greger(a)teigre.com> Date: Fri, 4 Nov 2005 08:10:04 +0100 >Brett, >I have never used avpairs with sql (I use radius), but I believe the >avp_db_load statement should be removed alltogether. The avps_column_str >instructs the auth module to load the value in the callscope column ON >SUCCESSFUL AUTHENTICATION, i.e. when the user is authenticated, the avp is >already loaded and ready to be used. >g-) > >----- Original Message ----- >From: "brett-ser-list" <brett-ser-list(a)worldcall.net> >To: <serusers(a)lists.iptel.org> >Sent: Friday, November 04, 2005 2:30 AM >Subject: [Serusers] avp_db_load fails with avp as source > > >> Hello All, >> I'm trying (desperately) to use avp_db_load with an avp: >> version: ser 0.9.3 (i386/linux) >> flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >> DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ >> main.c compiled on 12:28:21 Aug 16 2005 with gcc 3.3 >> >> modparam("auth_db","avps_column_str","callscope") >> modparam("avpops","avp_aliases","callgroups=i:200") >> modparam("avpops","avp_aliases","callscope=i:500") >> avp_db_load("$callscope","$callgroups/avps") ; >> >> In watching the debug, I see the avp loaded just fine from the database >> from the auth_db parameter: >> 3(1735) DEBUG:avpops:print_avp: p=0xf51c20d8, flags=3 >> 3(1735) DEBUG: name=<callscope> >> 3(1735) DEBUG: val_str=<200> >> >> but the above avp_db_load line is resulting in: >> 3(1735) DEBUG:avpops:get_val_as_str: no avp found >> 3(1735) ERROR:avpops:load_avps: failed to get uuid >> >> I also notice that it doesn't even bother to hit the database after the >> auth_db hit. So unless the avp table has been cached (Which I have no idea >> if it has..) how would it know? >> >> Seems like error #1: no avp found? Seems like it doesn't like my syntax >> but according to the avpops documentation, it should be right: >> source = (sip_uri|avp_alias|str_value) >> ['/'('username'|'domain'|'uri'|'uuid')] >> >> >> note: >> avp_db_load("$callscope/uuid","$callgroups/avps") ; >> gives me: >> 0(0) ERROR:avpops:fixup_db_avp: source/flags "callscope" unknown! >> >> Look, what I really want to do is use the authentication credentials to >> load an avp. Isn't that was avp is for? I don't want the FROM or the TO or >> the [R]URI, but the actual AUTHentication username. Any ideas?? >> >> Thanks!! >> BTW, I've tried the newest SER DEVEL CVS, and OPENSER 1.0 with basically >> the same results. Some custom modules I have would also make me want to >> stick with 0.9.0. >> >> -Brett >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > >

2 1

[Serusers] ser+asterisk example
by julien bos 05 Nov '05

05 Nov '05

Hi all, I am working with asterisk to active voicemail for ser. I am newbie for it. So can you give me an example of ser.cfg for do that? Thank you very much for your instruction.

3 4

Re: [Serusers] How to always remove auth credentials?
by brett-ser-list 05 Nov '05

05 Nov '05

I;m not entirely sure, but consume_credentials might take care of this.. Heh, or just get rid of the grandstreams! ick! ;) -Brett ---------- Original Message ---------------------------------- From: Federico Giannici <giannici(a)neomedia.it> Date: Sat, 05 Nov 2005 16:50:29 +0100 >I have found that some UAs (e.g Grandstream's ones) send the >Proxy-Authorization header even in ACKs messages. > >As we don't explicitly try to authenticate ACKs, we cannot use the >consume_credentials() function. In fact, it seems that we have to call >proxy_authorize() before we can use the consume_credentials(). But it's >contrary to RFC to call proxy_authorize() for ACKs and CANCELs methods! > >So, how can we prevent the Proxy-Authorization information to be >forwarded to subsequent hops? > > >Thanks. > >-- >___________________________________________________ > __ > |- giannici(a)neomedia.it > |ederico Giannici http://www.neomedia.it >___________________________________________________ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

3 2

[Serusers] Nat Problem
by harry gaillac 05 Nov '05

05 Nov '05

Hello, Is it possible to set host:port in the contact field when registration is sent ? Does nathelper allow this ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] How to always remove auth credentials?
by Federico Giannici 05 Nov '05

05 Nov '05

I have found that some UAs (e.g Grandstream's ones) send the Proxy-Authorization header even in ACKs messages. As we don't explicitly try to authenticate ACKs, we cannot use the consume_credentials() function. In fact, it seems that we have to call proxy_authorize() before we can use the consume_credentials(). But it's contrary to RFC to call proxy_authorize() for ACKs and CANCELs methods! So, how can we prevent the Proxy-Authorization information to be forwarded to subsequent hops? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 0

[Serusers] Value of max_len
by Federico Giannici 05 Nov '05

05 Nov '05

Is there a way to know what is the value assigned to the "max_len" core value? I wasn't able to find it looking at the source code... Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 0

[Serusers] where is the remove_hf function?
by Dale J. Robertson 05 Nov '05

05 Nov '05

I used and tested the remove_hf function in ser 8.14 on a test box running netbsd. After confirming that it operated to my satisfaction, I loaded the latest package on my soekris single board i386 computer running openbsd 3.7. When ser started up and parsed my config, It complained that it couldn't find the remove_hf function. Was this function removed from ser 9.4? or has it been relocated to an external module. Thanks, Dale

1 0

[Serusers] Windows Messenger charges user fees?
by Michael Kendall 04 Nov '05

04 Nov '05

Mark, I cannot tell you how much I appreciate your rare and insightful comments about Windows Messenger (WM). Valuable warnings of this kind are understandably difficult to find on the Microsoft Website, or anywhere else for that matter. The restrictions you describe are probably the reason that WM gets so little mention on the serusers list. I visited the Microsoft website describing WM, and all of the WM documents I find are dated 2001, as if Microsoft has abandoned the WM product, which I don't think is really the case. The documents did not mention any plan for me to pay a fee for WM use, with or without whiteboard and IM service. The site makes WM appear 100% free. Can you please direct me to a website or a phone number where I can find out the amount and conditions of such WM charges? During my startup phase, while I am evaluating the viability of my tutoring business model, it is probably acceptable for me to pay WM fees to Microsoft, and to restrict myself to the uPnP routers required by WM. But over the longer term, both of those limitations will present a serious problem, just as you have suggested. Microsoft's failure to include STUN support in WM is a serious error which limits its deployability. thanks for your help, Michael

2 2

[Serusers] Asterisk Sending SER 407
by Brett Nemeroff 04 Nov '05

04 Nov '05

Hey All, For the life of me, I can't figure this out. And I know it should be really simple. I can't get Asterisk to accept a call from SER. It appears that it's trying to authenticate the ORIGINAL caller and not the proxy itself. I've set asterisk up to use insecure=yes, but I still get a 407. I know this may be an Asterisk question, but I'm not sure if I need to be doing something different (like a consume credentials) on the SER side. Any ideas? -Brett

1 0

[Serusers] Line transfer and call waiting
by Christophe Irles 04 Nov '05

04 Nov '05

Hi, I wondering how to implement line transfer and call waiting. For the line transfer on the user side, it's necessary to create a special "ser.cfg" ? Or it's only a feature on the SIP phone/device ? How about the call waiting ? Need only a specific SIP phone/device ? Thanks, Christophe

2 1

[Serusers] serctl moni
by Richard C. Thompson 04 Nov '05

04 Nov '05

Can anyone tell me if serctl moni tells you the number of calls in progress? Transaction Statistics Current: 0 (38 waiting) Total: 2918 (0 local) Replied localy: 7014 Completion status 6xx: 0, 5xx: 26, 4xx: 546, 3xx: 0,2xx: 3410 Is it the number of waiting transactions? Thanks Richard C. Thompson

2 1

[Serusers] Checking Authorization Username == From: Username
by Russell James Bate 04 Nov '05

04 Nov '05

I have a requirement to ensure that Authorization Username == From: Username before a call is allowed. I would appreciate any advice, guidance or reference to other resources that may enable me to do this. Kind regards, Russell Bate Livewire International -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.6/152 - Release Date: 31/10/2005 The information transmitted including any attachments is intended only for the exclusive use of the person(s) or entity to which it is addressed and may contain confidential and privileged material that may be subject to legal privilege. Any perusal, review, retransmission, dissemination, distribution, reproduction or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient without the sender's prior consent is unauthorised and strictly prohibited. If you have received this message in error, please notify the sender immediately and delete the message from your computer.

2 1

[Serusers] SER performance when enabling presence & IM
by Vanlandschoote Michel 04 Nov '05

04 Nov '05

Hi again, yet another query about SER performance figures (somewhat related to my previous post): how many users can you put on your machine running SER stateful proxy when enabling SIP presence & IM? And which hardware (CPU & memory) are you using? Many thanks, Michel ************************************************************* Dit e-mail bericht inclusief eventuele ingesloten bestanden kan informatie bevatten die vertrouwelijk is en/of beschermd door intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit bericht (waaronder de volledige of gedeeltelijke reproductie of verspreiding onder elke vorm) door andere personen dan de geadresseerde(n) is verboden. Indien u dit bericht per vergissing heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit bericht te verwijderen. This e-mail and any attachment thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the sole use of the addressees. Any use of the information contained herein (including but not limited to total or partial reproduction or distribution in any form) by other persons than the addressees is prohibited. If you have received this e-mail in error, please notify the sender and delete its contents. *************************************************************

1 0

[Serusers] SER & RTPProxy performance figures
by Vanlandschoote Michel 04 Nov '05

04 Nov '05

Hi all, I'd like to know some real life performance figures for both SER and RTPProxy (with both applications running on separate machines): SER (as stateful SIP proxy & registrar): - how many calls-per-second and how many concurrent sessions do you handle per machine? - and of course, which hardware are you using (CPU & memory) RTPProxy: - how many streams do you handle per machine? - most important of all; which hardware (CPU & memory) - would be interesting if you also know: - which voice/video codecs are used? - what's the overall throughput in Mbps? Any feedback is highly appreciated. Michel ************************************************************* Dit e-mail bericht inclusief eventuele ingesloten bestanden kan informatie bevatten die vertrouwelijk is en/of beschermd door intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit bericht (waaronder de volledige of gedeeltelijke reproductie of verspreiding onder elke vorm) door andere personen dan de geadresseerde(n) is verboden. Indien u dit bericht per vergissing heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit bericht te verwijderen. This e-mail and any attachment thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the sole use of the addressees. Any use of the information contained herein (including but not limited to total or partial reproduction or distribution in any form) by other persons than the addressees is prohibited. If you have received this e-mail in error, please notify the sender and delete its contents. *************************************************************

1 0

[Serusers] SER+ASTERISK
by harry gaillac 04 Nov '05

04 Nov '05

Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-----192.168. ser 5060------- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] Mediaproxy Clarification
by Daryl Sanders 04 Nov '05

04 Nov '05

I would really appreciate any feedback or clarification on my understanding of Mediaproxy. It appears that Mediaproxy can be setup in such a way to allow load balancing based on domain. This seems to be the only native mechanism for getting any sort of pseudo geo-location out of it. My understanding is as follows... My users ATAs will query DNS which will direct them to a SIP proxy based on my SRV records. The SIP proxy (SER) machines will be running proxydispatcher.py and have the required mediaproxy module loaded. In addition, the SER machines may also have mediaproxy.py running locally, or mediaproxy.py may be on separate machines altogether. When a request for routing comes into the SIP proxy it is determined then if rtp proxying is necessary based on source IP. If rtp proxying is required, proxydispatcher.py send a DNS query to locate a suitable mediaproxy server. This is where SRV is used to decide (based on domain) which mediaproxy server is to be used. Given this, a separate domain would need to be setup for each geographic region that is be served by one or more mediaproxy servers. Then these domains are published to SRV to facilitate the lookups performed by each proxydispatcher.py instance. Please let me know if I'm understanding all of this correctly. Thanks! - Daryl

2 1

[Serusers] Ser and Asterisk
by Aster 04 Nov '05

04 Nov '05

SerUser, I am trying to use SER and Asterisk on the same host/system. When i try to make ser talk to asterisk, it failed and didn't want to talk to it. But if i put asterisk on different system then ser has no problem talking to asterisk. Here is flow of the traffic: UA ----------- SER ----------------<if PSTN> ------------ Asterisk ------------- Gateway. <if another user> ---------- UA Could someone please give me an idea how to make ser talk to asterisk on the same host. TIA

3 4

[Serusers] Failure route and CANCEL
by H Quintana 04 Nov '05

04 Nov '05

Hi, I'm doing some tests using the failure route feature. I send a call to an offline gateway and after the timeout, I'm sending the call to an online Gateway. It works fine if the call is not canceled, but when I did so before the offline route timeout, SER is not stopping and is sending the call to the online route. It seems that in this case the Reply status is NULL because I'm getting: "Entering failure_route. STATUS: <null>" in the log. Any ideas? Thanks for your help, Humberto ================================ failure_route[2] { xlog("L_INFO", "Entering failure_route. STATUS: %rs\n"); if (t_check_status("487")) { xlog("L_INFO", "Call cancelled\n"); break; } if (t_check_status("408") ) { xlog("L_INFO", "Time out. Gateway offline\n"); break; } rewritehost("a.b.c.d"); append_branch(); if (!t_relay()) { sl_reply_error(); break; } break; } __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

[Serusers] sipc unavailable for commercial users
by Michael Kendall 03 Nov '05

03 Nov '05

Mark, You suggested that I try sipc as an alternative to Windows Messenger. Sipc sounds like a great program since it has add-ons for voip, video, and whiteboard, as well as support for STUN. Sipc is also free for educational and research users. But I am not sure that my for-profit tutoring service would qualify as an educational organization. So I contacted Sipquest.com about pricing for a commercial license. They sent me to 3Com, who is supposed to be using sipc in their VCX product for "medium to very large businesses". The 3Com saleguy said he had never sold sipc to anyone, but he would pass my name to a company called Parallel Technologies which is considering creating a SIP server and might be planning to use sipc as a client. In summary, it appears that the commercial version of sipc is not available right now. I will try to get a free educational copy of sipc, just to evaluate it. thanks Michael

1 0

[Serusers] SRC Port change
by Ritesh Jalan 03 Nov '05

03 Nov '05

Hi Pls. help us in resolving the problem that the SIP proxy server changes the Source and Destination port while replying for message. Pls. find the below message for the Reference In below message A cisco ata device (which is behind NAT with IP 192.168.10.52) is trying to register it self with SIP proxy server IP 172.16.101.10 with SRC port 5060 and DST port 5060, but when SIP proxy server replies back to the message the port number gets changed i.e SRC port 53492 and DST port 5060. Is there any way by which we can fix this port num. No. Time Source Destination Protocol Info 532 152.672945 172.16.101.251 172.16.101.10 SIP Request: REGISTER sip:172.16.101.10 Frame 532 (469 bytes on wire, 469 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.280623000 Time delta from previous packet: 0.007492000 seconds Time since reference or first frame: 152.672945000 seconds Frame Number: 532 Packet Length: 469 bytes Capture Length: 469 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Cisco_b5:15:f0 (00:04:c1:b5:15:f0), Dst: Dell_17:ea:82 (00:14:22:17:ea:82) Destination: Dell_17:ea:82 (00:14:22:17:ea:82) Source: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.251 (172.16.101.251), Dst: 172.16.101.10 (172.16.101.10) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x68 (DSCP 0x1a: Assured Forwarding 31; ECN: 0x00) 0110 10.. = Differentiated Services Codepoint: Assured Forwarding 31 (0x1a) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 455 Identification: 0x01f8 (504) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 249 Protocol: UDP (0x11) Header checksum: 0x1e63 [correct] Source: 172.16.101.251 (172.16.101.251) Destination: 172.16.101.10 (172.16.101.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 435 Checksum: 0x6c94 [correct] Session Initiation Protocol Request-Line: REGISTER sip:172.16.101.10 SIP/2.0 Method: REGISTER Resent Packet: True Suspected resend of frame: 34 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK9749dfca14c0ffb2 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 Call-ID: 3431569068(a)192.168.10.52 CSeq: 1 REGISTER Contact: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 Contact Binding: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 URI: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp> SIP Display info: 23456 SIP contact address: sip:23456@192.168.10.52:5060 User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A) Content-Length: 0 No. Time Source Destination Protocol Info 533 152.674292 172.16.101.10 172.16.101.251 SIP Status: 401 Unauthorized (0 bindings) Frame 533 (432 bytes on wire, 432 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.281970000 Time delta from previous packet: 0.001347000 seconds Time since reference or first frame: 152.674292000 seconds Frame Number: 533 Packet Length: 432 bytes Capture Length: 432 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Dell_17:ea:82 (00:14:22:17:ea:82), Dst: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Destination: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Source: Dell_17:ea:82 (00:14:22:17:ea:82) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.10 (172.16.101.10), Dst: 172.16.101.251 (172.16.101.251) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x60 (DSCP 0x18: Class Selector 3; ECN: 0x00) 0110 00.. = Differentiated Services Codepoint: Class Selector 3 (0x18) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 418 Identification: 0x0000 (0) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x9988 [correct] Source: 172.16.101.10 (172.16.101.10) Destination: 172.16.101.251 (172.16.101.251) User Datagram Protocol, Src Port: 53492 (53492), Dst Port: 5060 (5060) Source port: 53492 (53492) Destination port: 5060 (5060) Length: 398 Checksum: 0x0ad1 [correct] Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: True Suspected resend of frame: 480 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;received=172.16.101.251;branch=z9hG4bK9749dfca14c0ffb2 Call-ID: 3431569068(a)192.168.10.52 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 CSeq: 1 REGISTER WWW-Authenticate: DIGEST realm="CISCO", nonce="4369a82d", qop="auth", algorithm=MD5 Content-Length: 0 Ritesh Jalan Senior Engineer - Business Solution Net4India Ltd. D-25 Secto3 Noida - 201301 India INDIA Tel: 91-120-5323500 Fax: 91-120-5323520 URL: http://www.net4.in _____ This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation. _____

2 1

[Serusers] Packets fragmentation and message reduction
by Federico Giannici 03 Nov '05

03 Nov '05

I found a situation where some SIP phones worked and some don't. After a while I found that, because of the added bytes of the "Proxy-Authorization:" header, the messages generated by some sip phones were larger than 1480 bytes and this caused the connection with the PSTN gateway to not work. I tried to forwad the message with the t_relay_to_tcp() function, but by the ngrep output it seems that it still uses UDP (but I may be wrong). Anyway, what can be done in this situations? What about a SER module that rewrite all header field names with their compact form? Often, the message is only a few bytes more than 1480, so this could be a solution... What do you think? Bye. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

5 7

[Serusers] SERWEB Username/password
by gorand＠dvvti.com 03 Nov '05

03 Nov '05

I finally got Serweb to work. Thanks to all who contributed to that! Now I am running into a very stupid problem. I can't log in! What is the username and password? Or do i have to create it or edit certain files? Please let me know. Thanks Goran

1 0

[Serusers] FW: EmbeddedObj.o Sipstack Make error
by administrator 03 Nov '05

03 Nov '05

_____ From: administrator [mailto:administrator@hellasfon.com] Sent: Wednesday, October 26, 2005 11:49 AM To: 'b2bua(a)vovida.org' Subject: EmbeddedObj.o Make error Hello everybody, I am trying to install b2bus from cvs and I get the following error: make[1]: Leaving directory `/root/vocal/sip/sipstack' make[1]: Entering directory `/root/vocal/sip/sipstack' g++ -Wall -fPIC -Wno-deprecated -D_REENTRANT -DUSE_PTHREADS -g -I../../build -I../../build/../sdp/sdp2 -I../../build/../util -I../../build/../util/threads -I../../build/../util/logging -I../../build/../util/crypto -I../../build/../util/statistics -I../../build/../util/snmp -I../../build/../util/signals -I../../build/../util/behavior -I../../build/../util/io -I../../build/../util/services -I../../build/../util/transport -I../../build/../util/config -I../../build/../util/dnssrv -I../../build/../util/deprecated -I../../build/../util/adt -I../../build/../contrib/libxml2.Linux.i686 -I../../build/../contrib/libxml2.Linux.i686/include/libxml -I../../build/../contrib/libxml2.Linux.i686/include -DVOCAL_USE_DEPRECATED -DVOCAL_USING_PENTIUM -DOLD_PROV -c -o obj.debug.Linux.i686/EmbeddedObj.o EmbeddedObj.cxx BaseUrl.hxx:146: error: specialization of βtemplate<class _Key> struct __gnu_cxx::hashβ in different namespace /usr/lib/gcc/i386-redhat-linux/4.0.1/../../../../include/c++/4.0.1/ext/hash_ fun.h:71: error: from definition of βtemplate<class _Key> struct __gnu_cxx::hashβ EmbeddedObj.cxx: In member function βData Vocal::EmbeddedObj::doReverseEscape(const std::string&)β: EmbeddedObj.cxx:213: warning: comparison between signed and unsigned integer expressions make[1]: *** [obj.debug.Linux.i686/EmbeddedObj.o] Error 1 make[1]: Leaving directory `/root/vocal/sip/sipstack' make: *** [sip] Error 2 I user SER(0.9.4) and freeRadius (latest) on Fedora Core 4. Please Help ASAP. Thank you in advance Yiannis Marios

3 4

[Serusers] conference server question
by Jorge C 03 Nov '05

03 Nov '05

Hello everybody, I have a question about the conference server. I wanted how the mixer part is done. Specifically, how it could be implemented. For example, should the input streams (and so the input samples) from different participants be interleaved to build the output stream? or should they be added in linear way? are there other alternatives? Thanks in advance!

1 0

[Serusers] multiple avp tables
by Iqbal 03 Nov '05

03 Nov '05

Hi Is it possible to have multiple avp tables, if so how is it done, is it via mulitple modparam entries as and when needed Iqbal

2 1

[Users] Max-Forward - new functionalities
by Bogdan-Andrei Iancu 03 Nov '05

03 Nov '05

Hi all, I just committed some new functionalities in maxfwd modules. Shortly: - more detailed reporting via the return code of the functions - success cases and error cases - new function - is_maxfwd_lt(x) - checks if the value of the MF is less then x. Is useful when you forward via a path with a known number of hops. Ex: avoid fwd to another proxy requests with MF=0 (which will be rejected for sure). for more details, please see http://www.openser.org/docs/modules/1.0.x/maxfwd.html regards, bogdan

1 1

[Users] problem with uac_replace_from
by Brandon Price 03 Nov '05

03 Nov '05

I am using uac_replace_from to set the Sip display info. The problem is that it only replaces the display when one exists and is passed. It doesn't create one when one doesn't exist. So for example if I use ... uac_replace_from("batman",""); then place a call from one of my registered user agents to another then the callerid name value is displayed properly as "batman", however if I place a call from the pstn to one of my user agents which comes in with no Sip display into set then it is not generated as batman and added. I need sip display info to be replaced if it does exist, and generated if it doesn't. Any help will be greatly appreciated.

3 4

[Serusers] Need help to implement into my ser.cfg compatibility with sems or voicemail
by Antonio Membrives 03 Nov '05

03 Nov '05

Hi to all, sorry for this mail, but, I need some help to implement some voicemail compatibility, above I put my ser.cfg, this config work in Fedora 3 + MySQL + Ser 0.9.3 and work fine, the server have 2 NIC and can talk with user into my lan and out my land. But I read some documents of SEMS and try some configs and don't work. If anyone can waste some time to see this code and help me a gratefully her with some beer if visit my city for tourist :D Sems are 0.9.0 and the config are above. Sorry for my horrible English and thanks for the help. Antonio Membrives ####---------------------- ser.cfg ---------------------------------### debug=3 fork=yes log_stderror=no listen=82.159.201.35 listen=192.168.10.240 alias=telintcable.es alias=82.159.201.35 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db","use_domain",1) modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("nathelper", "rtpproxy_disable", 1) 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("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registar@telintcable.es") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("tm", "fr_timer", 10) modparam("tm", "fr_inv_timer", 15) modparam("tm", "wt_timer", 10) modparam("acc", "log_flag", 2) #1 modparam("acc", "log_level", 2) modparam("acc", "db_flag", 2) #1 modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsp") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("82.159.201.35:5060;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 (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; setflag(1); # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; 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")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Not Found"); acc_db_request("Llamada perdida","missed_calls"); break; }; sl_send_reply("404", "Usuario Not Found"); break; }; setflag(3); 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("telintcable.es","subscriber")) { www_challenge("telintcable.es","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; m_dump(); #msilo.so } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("telintcable.es","subscriber")) { proxy_challenge("telintcable.es","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if(uri=~"sip:100.*@telintcable.es" || uri=~"sip:100.*@82.159.201.35"){ log(1, "LOG: Contactando con el modulo de conferencia\n"); if(!t_write_req("/tmp/am_fifo","conference")){ t_reply("500","Error contacting sems conference"); }; log(1, "LOG: Cerrando conexion con el modulo de conferencia\n"); break; }; if(method=="BYE" || method=="CANCEL"){ log(1,"LOG: vm end - inicio\n"); if(!t_write_req("/tmp/am_fifo","bye")){ log(1,"LOG: No se ha podido contactar con vm\n"); t_reply("500","Error contacting sems"); }; log(1,"LOG: vm end - final\n"); }; if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Usuario no conectado"); acc_db_request("Llamada perdida","missed_calls"); break; }; sl_send_reply("404", "Usuario no encontrado"); break; }; setflag(3); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { 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(); }; } ### ------------------- sems.config ---------------------------------#### # $Id: sems.conf.sample,v 1.21 2005/01/05 16:48:04 sayer Exp $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # # # whitespaces (spaces and tabs) are ignored # comments start with a "#" and may be used inline # # example: option=value1, value2 # i like this option # ################################## # global parameters # ################################## # optional parameter: fork={yes|no} # # - specifies if sems should run in daemon mode (background) # (fork=no is the same as -E) fork=yes # optional parameter: stderr={yes|no} # # - debug mode: do not fork and log to stderr # (stderr=yes is the same as -E) stderr=no # optional parameter: loglevel={0|1|2|3} # # - sets log level (error=0, warning=1, info=2, debug=3) # (same as -D) loglevel=1 # optional parameter: fifo_name=<filename> # # - path and file name of our fifo file (same as -i) fifo_name=/tmp/am_fifo # optional parameter: ser_fifo_name=<filename> # # - path and file name of Ser's fifo file (same as -o) ser_fifo_name=/tmp/ser_fifo # optional parameter: socket_name=<filename> # # - path and file name of our unix socket #socket_name=/tmp/am_sock # optional parameter: ser_socket_name=<filename> # # - path and file name of Ser's unix socket #ser_socket_name=/tmp/ser_sock # optional parameter: send_method=<filename> # # - sets which method will be used to contact Ser: # FIFO or unix socket server. # - values: fifo (default), unix_socket # # - PLEASE NOTE THAT: if you use 'fifo', make sure that # fifo_name and ser_fifo_name have been set. if you use 'unix_socket', # make sure that socket_name and ser_socket_name have been set. # #send_method=unix_socket send_method=fifo # optional parameter: plugin_path=<path> # # - sets the path to the plug-ins # - may be absolute or relative to CWD plugin_path=/usr/local/lib/sems/plug-in/ # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=82.159.201.34 # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 # optional parameter: rtp_low_port=<port> # # - sets port of rtp lowest server #rtp_low_port=1024 # optional parameter: rtp_high_port=<port> # # - sets port of rtp highest server #rtp_high_port=65535 ################################## # module specific parameters # ################################## # sample voicemail configuration (inline) config.voicemail=inline # optional parameter: rec_file_extension=<ext> # # - sets the file extension which will be used # to record messages. # - example 1: wav # - example 2: mp3 # # WARNING: you must compile the mp3 plug-in # in order to use 'mp3' as default # extension. You will need the lame # encoder for that. # See plug-in/mp3/ReadmeMP3.txt for # more information. # rec_file_extension=wav # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for # - the file to be played is searched in the following order: # <announce_path>/<domainname>/<username>.wav # <announce_path>/<domainname>/<language>/<default_announce> # <announce_path>/<domainname>/<default_announce> # <announce_path>/<language>/<default_announce> # <announce_path>/<default_announce> # where <language> is taken from the body of P-Language header # of the request (if any). announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # parameter: max_record_time=<seconds> # # - maximum record time max_record_time=30 # parameter: accept_delay=<x> # # - delays accepting of the call for x seconds # - default value is 0 accept_delay=0 # parameter: email_template_path=<filename> # # - email templates path # # See the README file in <sems-src>/plug-in/voicemail # for more information on the syntax used. # - template to be used is selected in the following order: # <email_template_path><domain>_<language>.template # <email_template_path><domain>.template # <email_template_path>default_<language>.template # <email_template_path>default.template # where <language> is taken from the body of P-Language header # of the request (if any). # email_template_path=/usr/local/etc/sems/ # end of configuration section for voicemail module config.voicemail=end # sample announcement configuration (inline) config.announcement=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # end of configuration section for announcement module config.announcement=end # sample isdngw module configuration (external file) #config.isdngw=/etc/isdngw.conf # sample ivr module configuration (inline) config.ivr=inline # # # The IVR checks for a script with the named of the callee # (<local part in r-uri>.py for python, <local part in r-uri>.pl for perl) # in the directory <ivr_script_path><domain>, then for # <ivr_script_path><domain><ivr_script_file>. If this is not found, # <ivr_script_path>/<local part in r-uri>.py if searched, # and if this is not found, <ivr_script_path>/<ivr_script_file> is used. # # So with a call to sayer(a)iptel.org and #ivr_script_path=/etc/ivr and #ivr_script_file=ivr.py # these files are checked: #/etc/ivr/iptel.org/sayer.py #/etc/ivr/iptel.org/ivr.py #/etc/ivr/sayer.py #/etc/ivr/ivr.py # #parameter: ivr_script_path: # path to ivr scripts # ivr_script_path=/etc/ivr/ # default script file: this will be executed if <user>.py does not exist. # ivr_script_file=ivr.py # parameter : tts_caching # y or n # text will be read from waves already synthesized and # cached in cache_path tts_caching=y # parameter : tts_cache_path # path to cache waves # path must exist! tts_cache_path=/tmp/wavs # end of configuration section for ivr module config.ivr=end # sample conference configuration (inline) config.conference=inline # parameter: default_announce=<filename> # # - sets the full pathed name of the default announce WAV file. # Will be played to lonely users. default_announce=/usr/local/lib/sems/audio/first_participant.wav join_sound=/usr/local/lib/sems/audio/beep.wav drop_sound=/usr/local/lib/sems/audio/beep.wav # end of configuration section for conference module config.conference=end # example configuration for number reader config.number_reader=inline number_path=/usr/local/lib/sems/audio/ prolog_file=welcome_to_number_reader.wav epilog_file=thanks_calling_number_reader.wav # end of number_reader configuration config.number_reader=end # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end

1 0

[Users] MySql accounting in OpenSer 1.0
by Hilco de Lathouder 03 Nov '05

03 Nov '05

2 1

RE: [Serusers] Packets fragmentation and message reduction
by Vitaly Nikolaev 03 Nov '05

03 Nov '05

MTU 1500 is kind a everywhere (almost) if you increase it on your interface/router then your ISP will find out that your packet too long and split it.. or block In my case I removed all information fields from SIP header -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Arek Bekiersz Sent: Thursday, November 03, 2005 4:11 AM To: Federico Giannici; SER Users Subject: Re: [Serusers] Packets fragmentation and message reduction Hi Federico, Had the same problems. Solutions that work: - you have somewhere MTU limit of 1500 bytes. Check where you have this limit, it is possible you will be able to send bigger messages after changes - use SER's consume_credentials() function to remove Authorization or Proxy-Authorization HF after use - configure all phones to use and describe in SDP only 1 or 2 codecs (decide about your system codec, delete the rest from UAs) - write your small function (in example in nathelper module) to edit SDP and remove unused codecs when messages go through proxy Arek, ----- Original Message ----- From: "Federico Giannici" <giannici(a)neomedia.it> To: "SER Users" <serusers(a)lists.iptel.org> Sent: Thursday, November 03, 2005 1:24 AM Subject: [Serusers] Packets fragmentation and message reduction > I found a situation where some SIP phones worked and some don't. > > After a while I found that, because of the added bytes of the > "Proxy-Authorization:" header, the messages generated by some sip phones > were larger than 1480 bytes and this caused the connection with the PSTN > gateway to not work. > > I tried to forwad the message with the t_relay_to_tcp() function, but by > the ngrep output it seems that it still uses UDP (but I may be wrong). > > Anyway, what can be done in this situations? > > What about a SER module that rewrite all header field names with their > compact form? Often, the message is only a few bytes more than 1480, so > this could be a solution... > > What do you think? _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] mathing prefix
by Iqbal 03 Nov '05

03 Nov '05

Hi Anyone know regex to match 0061 andf not 00611, basically match australia, and not australia mobile, have thought of 0061[2-9], but if I need to do this for all countries its a pain, since they all have their own logic. iqbal

1 0

[Serusers] Problem in SER+SEMS conference with NAT
by Ravic Costa 03 Nov '05

03 Nov '05

Hello, We have configured SER 0.9.4 and SEMS last cvs version in order to implement a conference funcionallity. Everything is working fine when the clients are using public IPs. When a client starts a conference, he listens a message that in on a wav file. When a client is behind NAT, when he tries to start a conference, he doen's listen to any message and in the soft phone (X-Lite) it appears a message saying "connected". We have captured the traffic with tcpdump, on the machine that ser and sems is running, and we've noticed that the RTP packets that the client behind NAT send are well received, however, there is only one RTP packet sent by the SER to the Client, and the Client does not get that message, maybe because he is behind NAT. Any ideia? In the ser.cfg file, to the conference funcionallity we have : _________________________________________________________ route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # select messages to redirect: if ( method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" ){ # switch to stateful mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; # prevent timeout on the other side: t_reply("100","Trying - just wait a minute !"); # actively absorb ACKs if (method == "ACK") { t_relay(); break; } if (method=="INVITE"){ if (nat_uac_test("19")) { setflag(6); } # redirect the call to the 'conference' plug-in # if the URI begin with 100 if (uri=~"sip:100.*@") { # assumes that Sems configuration parameter 'socket_name=' # has been set to /tmp/am_sock if(!t_write_req("/tmp/am_fifo","conference")) { t_reply("500","error contacting sems"); }; break; route(4); route(1); }; } else if (method=="BYE" || method=="CANCEL") { # Sems should already know which plug-in is handling that # call. "bye" is no plug-in name. It is a reserved name which # tells Sems to terminate the call. if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; }; }; } _______________________________________________________ Best regards, Ravic Costa

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[Serusers] Redirect server
by harry gaillac 03 Nov '05

03 Nov '05

Hello, I need your advises Here is my problem : I setup asterisk ser : Asterisk 5050-- |--Firewall---|----- SIP User agents SER 5060--- My client use SER as outbound proxy and asterisk as sip registrar. SER is able to forward SIP register request to asterisk however . How SER can inform SIP User agents to send register requests to asterisk. May I have to configure a redirect server to advise SIP User agents of the sip register server (asterisk)? can you help me for ser.cfg if possible ? Regards ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] accounting on mysql
by Colin Jordan 03 Nov '05

03 Nov '05

Hi there Can you please help me to setup accounting on mysql. I need to log the call transactions. Best regards, Colin

1 0

[Serusers] problem with ser registration
by rajesh volluru 03 Nov '05

03 Nov '05

hi I get Process Request Timedout ever whenever i try to register with SER. I get following messages on starting ser. 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 1(0) DEBUG: init_mod_child (-1): tm 1(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57' 1(0) DEBUG: init_mod_child (-1): usrloc 1(0) get_connection(): Connection not found in the pool 2(0) INFO: fifo process starting: 10317 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(10313) DEBUG: init_mod_child (1): tm 0(10313) DEBUG: callid: '3fbfd771-10313(a)69.229.23.57' 0(10313) DEBUG: init_mod_child (1): usrloc 0(10313) get_connection(): Connection not found in the pool 1(10316) DEBUG: init_mod_child (-1): auth_db 1(10316) get_connection(): Connection not found in the pool 0(10313) DEBUG: init_mod_child (1): auth_db 0(10313) get_connection(): Connection not found in the pool 1(10316) DEBUG: init_mod_child (-1): domain 1(10316) DEBUG: init_mod_child (-1): nathelper 1(10316) rtpp_test: RTP proxy found, support for it enabled 0(10313) DEBUG: init_mod_child (1): domain 0(10313) get_connection(): Connection found in the pool 0(10313) DEBUG: init_mod_child (1): nathelper 0(10313) rtpp_test: RTP proxy found, support for it enabled 2(10317) DEBUG: init_mod_child (-2): auth_db 2(10317) get_connection(): Connection not found in the pool 2(10317) DEBUG: init_mod_child (-2): domain 2(10317) DEBUG: init_mod_child (-2): nathelper 2(10317) rtpp_test: RTP proxy found, support for it enabled 2(10317) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(10317) DEBUG: register_fifo_cmd: new command (print) registered 2(10317) DEBUG: register_fifo_cmd: new command (uptime) registered 2(10317) DEBUG: register_fifo_cmd: new command (version) registered 2(10317) DEBUG: register_fifo_cmd: new command (pwd) registered 2(10317) DEBUG: register_fifo_cmd: new command (arg) registered 2(10317) DEBUG: register_fifo_cmd: new command (which) registered 2(10317) DEBUG: register_fifo_cmd: new command (ps) registered 2(10317) DEBUG: register_fifo_cmd: new command (kill) registered 2(10317) WARNING: no fifo_db_url given - fifo DB commands disabled! Can anyone please have a look at above information and let me know what is wrong with ser? Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here --------------------------------- Enjoy this Diwali with Y! India Click here

2 1

[Serusers] SRC Port Change
by Ritesh Jalan 03 Nov '05

03 Nov '05

Hi Pls. help us in resolving the problem that the SIP proxy server changes the Source and Destination port while replying for message. Pls. find the below message for the Reference In below message A cisco ata device (which is behind NAT with IP 192.168.10.52) is trying to register it self with SIP proxy server IP 172.16.101.10 with SRC port 5060 and DST port 5060, but when SIP proxy server replies back to the message the port number gets changed i.e SRC port 53492 and DST port 5060. Is there any way by which we can fix this port num. No. Time Source Destination Protocol Info 532 152.672945 172.16.101.251 172.16.101.10 SIP Request: REGISTER sip:172.16.101.10 Frame 532 (469 bytes on wire, 469 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.280623000 Time delta from previous packet: 0.007492000 seconds Time since reference or first frame: 152.672945000 seconds Frame Number: 532 Packet Length: 469 bytes Capture Length: 469 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Cisco_b5:15:f0 (00:04:c1:b5:15:f0), Dst: Dell_17:ea:82 (00:14:22:17:ea:82) Destination: Dell_17:ea:82 (00:14:22:17:ea:82) Source: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.251 (172.16.101.251), Dst: 172.16.101.10 (172.16.101.10) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x68 (DSCP 0x1a: Assured Forwarding 31; ECN: 0x00) 0110 10.. = Differentiated Services Codepoint: Assured Forwarding 31 (0x1a) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 455 Identification: 0x01f8 (504) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 249 Protocol: UDP (0x11) Header checksum: 0x1e63 [correct] Source: 172.16.101.251 (172.16.101.251) Destination: 172.16.101.10 (172.16.101.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 435 Checksum: 0x6c94 [correct] Session Initiation Protocol Request-Line: REGISTER sip:172.16.101.10 SIP/2.0 Method: REGISTER Resent Packet: True Suspected resend of frame: 34 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK9749dfca14c0ffb2 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 Call-ID: 3431569068(a)192.168.10.52 CSeq: 1 REGISTER Contact: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 Contact Binding: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 URI: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp> SIP Display info: 23456 SIP contact address: sip:23456@192.168.10.52:5060 User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A) Content-Length: 0 No. Time Source Destination Protocol Info 533 152.674292 172.16.101.10 172.16.101.251 SIP Status: 401 Unauthorized (0 bindings) Frame 533 (432 bytes on wire, 432 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.281970000 Time delta from previous packet: 0.001347000 seconds Time since reference or first frame: 152.674292000 seconds Frame Number: 533 Packet Length: 432 bytes Capture Length: 432 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Dell_17:ea:82 (00:14:22:17:ea:82), Dst: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Destination: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Source: Dell_17:ea:82 (00:14:22:17:ea:82) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.10 (172.16.101.10), Dst: 172.16.101.251 (172.16.101.251) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x60 (DSCP 0x18: Class Selector 3; ECN: 0x00) 0110 00.. = Differentiated Services Codepoint: Class Selector 3 (0x18) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 418 Identification: 0x0000 (0) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x9988 [correct] Source: 172.16.101.10 (172.16.101.10) Destination: 172.16.101.251 (172.16.101.251) User Datagram Protocol, Src Port: 53492 (53492), Dst Port: 5060 (5060) Source port: 53492 (53492) Destination port: 5060 (5060) Length: 398 Checksum: 0x0ad1 [correct] Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: True Suspected resend of frame: 480 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;received=172.16.101.251;branch=z9hG4bK9749dfca14c0ffb2 Call-ID: 3431569068(a)192.168.10.52 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 CSeq: 1 REGISTER WWW-Authenticate: DIGEST realm="CISCO", nonce="4369a82d", qop="auth", algorithm=MD5 Content-Length: 0 Ritesh Jalan

2 1

[Serusers] Anyone can help me with sems
by Antonio Membrives 03 Nov '05

03 Nov '05

The scenario are ser 0.9.3 + sems 0.9.0 + mediaproxy + mysql, al instances are in the same machine The code in my ser.cfg if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { acc_db_request("Llamada perdida","missed_calls"); if(!t_write_req("/tmp/am_fifo","voicemail")){ t_reply("500","Error contacting sems voicemail"); }; #t_reply("404", "Usuario no conectado"); break; }; } else if(method=="BYE" || method == "CANCEL"){ if(!t_write_req("/tmp/am_fifo","bye")){ t_reply("500","Error contacting sems"); }; }; sl_send_reply("404", "Usuario no encontrado"); break; setflag(3); when i call a offline user, the softphone make a call but I don't ear any message or received any notification, anyone can help me? this is /var/log/messages Nov 3 11:04:56 dns2 ser[23838]: Maxfwd module- initializing Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially 110592 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially 110592 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142 Nov 3 11:04:56 dns2 ser[23847]: INFO: fifo process starting: 23847 Nov 3 11:04:56 dns2 ser[23847]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... Nov 3 11:05:07 dns2 ser[23848]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:08 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:10 dns2 ser[23851]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:14 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:23 dns2 mediaproxy[3158]: command delete 3924182bdc181a37 info= Nov 3 11:05:23 dns2 mediaproxy[3158]: command execution time: 0.20 ms

1 0

[Serusers] Accounting
by harry gaillac 03 Nov '05

03 Nov '05

Hello, Does acc module allow these requests to a radius server according to rfc2866 ? A Talk Start information is sent when SIP Server receives 200 OK for a INVITE request. A TAlK Stop information when SIP Server receives BYE. How may i configure it ? if acc module can't calculate acct-session-time why does this module provide radius support ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 8

[Serusers] SEMS + SER performance question
by Ryan Pagquil 03 Nov '05

03 Nov '05

Hi, How many simultaneous connection can SER + SEMS can handle? i deployed SER + SEMS in Pentium 2 300MHz with 128MB memory, how many connections can this handle? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] Ser+Media proxy
by KM 03 Nov '05

03 Nov '05

I have installed SER 0.9.3.2 and media proxy 1.2.1 on the same machine. The machine is on public network. Calls work ok except when I have a UA on a LAN behind NAT calling to a UA that is in the same LAN. In this case I get : +----------+-------------------------------+--------------------------+| username | contact | received |+----------+-------------------------------+--------------------------+| 22000001 | sip:22000001@192.168.1.7:5060 | sip:xxx.xxx.xxx.xxx:5060 || 22000002 | sip:22000002@192.168.1.6:5060 | sip:xxx.xxx.xxx.xxx:2557 |+----------+-------------------------------+--------------------------+The number that takes up port 5060 on the NAT server,which is 22000001 can call the other UA but not the other way round.22000002 can only receive calls.Please help me,KM

2 3

[Serusers] about rtp mixing
by Jorge C 03 Nov '05

03 Nov '05

Hello everybody, I have a question related with conference servers. Do you now how is done the mixing in a Conference? Specifically, how input RTP streams are mixed into output string? Should I need to mix in a linear way, interleaving samples of different users? Thanks in advance!!

1 0

[Serusers] Response Code 408 = "Request Timeout" or "ISUP 18 no user responding"
by Dmitry Isakbayev 03 Nov '05

03 Nov '05

Hi, I am trying to figure out how to interpret response code 408 in a failure_route block. It seem that there is a couple of different cases. If the code means that a PSTN gateway is not responding (the tm's fr_timer has expired), I want to try another PSTN gateway. On the other hand, if the code means that the user is not responding (the tm's fr_inv_timer has expired), I want to stop trying and forward the code to the UAC. How can I tell what response code 408 actually means? Thanks! Dmitry

1 0

[Serusers] P-Asserted-Identity
by Atle Samuelsen 02 Nov '05

02 Nov '05

Hi guys, and girls, Male and Female, do anybody know if anybody has made a implimentation of 3325 for ser? or, is anybody makeing some form of support for this? -Atle

3 8

[Serusers] SER behind a router
by Ángel Gómez 02 Nov '05

02 Nov '05

Hi, I am trilling to make SER workin conectec to internet via a router, the router only forwards the mensages to SER, the messages who go to the 5060 port. With this config i get SER working well locally, but when i try to register from the outside i can't. For more info I'm using sipsak for testing, and the version of ser is 0.9.3. Last thing: P L E A S E H E L P

2 1

[Users] re: ISP charging for voip
by Yair Hakak 02 Nov '05

02 Nov '05

hello all, not sure if everyone here is familiar with the madison river case: http://informationweek.smallbizpipeline.com/60405214<http://www.tmcnet.com/tmcnet/articles/2005/voip-call-blocking-fcc-madison-r…> not exactly the same thing, because they were blocking instead of charging (they are a small rural CLEC who, like almost all phone companies, charge less than cost for access and make their profit on minutes, and voip takes away their minutes), but the principle is the same, and the FCC nailed them. However, meanwhile the FCC chairman has changed, and who knows if the general way of thinking is still the same... at very least, it's good precendent. as someone who works for a telecom regulator (i wont tell you which one and where) i'd be very interested in what happens around the world. are there countries where ISP's are not allowed to filter or otherwise interfere in traffic? I know there are companies that sell traffic filtering systems to ISP's so they can prioritize and/or slow the performance of certain customers, certain apps (filesharing and the like), etc. it's relatively trivial to configure such systems to introduce massive packet loss to anything on port 5060, for example, or deprioritize UDP packets, or things of that nature. -yair

1 0

RE: [Serusers] voip charging by ISP's
by Vitaly Nikolaev 02 Nov '05

02 Nov '05

I believe It is unlawful act and there must be some regulation, I am not talking here that it will become known very well and this provider will lost some customers... and some other provider will claim that they voip friendly and get some more customer on it :-) ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Cesc Sent: Wednesday, November 02, 2005 1:32 PM To: Nils Ohlmeier Cc: serusers(a)lists.iptel.org; Roger Lewau; serusers-bounces(a)lists.iptel.org; Iqbal Subject: Re: [Serusers] voip charging by ISP's Well, i think the ISP may care if they are also providing similar bussiness. They won't probably be able to charge neither the voip user or voip provider, but they can do some harm. Identifying the rtp streams and sip signalling is not sooo difficult ... then, block a few messages, add delay and jitter to the rtp (or udp in general ...) and you've got a bad service, aparently coming from the voip provider. There is software to detect and do all the above (one was presented in VON this fall) ... There was an article in IEEE Spectrum (sept, october, nov ... can't remember). And all the tricks Nils mentions ... hey, maybe the advanced user, but my mom, no way! And then you need software that supports all those tricks ... that is even further away. In this sense, i think carriers have the winning side ... Regards, Cesc On 11/2/05, Nils Ohlmeier <lists(a)ohlmeier.org> wrote: On Wednesday 02 November 2005 14:28, Iqbal wrote: > I agree, I mean the next step would be to charge ebay and yahoo per > transaction, just wondering how easy it would be for ISP to setup a > monitor and a billing model on this I think the ISP should not care what it is transporting. The snail mail also does not look into what they are devlivering to charge you differently (except express delivery but thats another story). As soon as an ISP starts to look into the traffic from the user, the users will invent uncounted ways of fooling the ISP (use non-standard ports; use standard ports for the "wrong" service; use proxies; encrypt the traffic; and finally combine all this). All I can say to the ISP: happy debugging! And in the end I would be interested in how a court would judge about the outcoming bills (who has to prove what?) :-) Nils _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Manipulate To-Header
by Stefan Prelle 02 Nov '05

02 Nov '05

Hi all, is it possible to manipulate the From-Header (similiar to strip() and prefix()), before relaying a INVITE message? If so, how do I do this? Regards, Stefan

2 1

[Users] voip charging by ISP's
by Iqbal 02 Nov '05

02 Nov '05

Hi i was looking into a model where ISP may charge (SBC is already thiking of charging Vonage :-)), now if an ISP wanted to charge, all they would need is to extract the INVITE, BYE and all the stuff in the middle, to do some billing per VoIP provider sending traffic on there network, is anyone aware of any sniffers which could do this not from a proxy, but from the network level at a ISP end. Iqbal

5 7

[Users] TLS packages for debian unstable
by Bogdan-Andrei Iancu 02 Nov '05

02 Nov '05

Hi all, i just committed specs for debian to generate both tls and non-tls packages - thanks to Klaus for support ;). I also generated the packages for debian unstable - see : http://openser.org/pub/openser/latest/packages/deb-unstable/ in the next days I hope packages for different versions of debian will be uploaded. regards, bogdan

1 0

[Serusers] Finding IP address of VoIP gateway sending the 5xx
by Ramin Nikaeen 02 Nov '05

02 Nov '05

Valued Friends, I am trying to process 5xx response codes and to resend the INVITE to an alternative VoIP gateway upon receiving an 5xx response from a VoIP gateway. So, I need to somehow figure out the IP address of the VoIP gateway from which SER receives the 5xx response. I tested and I receive 503 response codes only in onreply_route and not in failure_route at all. The 503 response does not include any "contact" fields that Would indicate the IP address of the VoIP gateway from which The 503 response is being received at SER. Do you know how I can figure out the IP address of the VoIP gateway from which SER is receiving the 503 response?! Thanks ramin

2 1

[Serusers] SIP/2.0 477 error on sending to next hop occurred
by rajesh volluru 02 Nov '05

02 Nov '05

hi can anyone explain me why i get the error mentioned in below trace. Please let me know how can i resolve this. SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Call-ID: 21b297bc85242e507848f6c297cf4c3f(a)69.229.23.54 CSeq: 1 REGISTER From: "test11" <sip:test11@iptel.org:5060;transport=tcp>;tag=6626965 To: "test11" <sip:test11@iptel.org:5060;transport=tcp>;tag=321967ba5deb3e7f484a2df055818800-a111 Via: SIP/2.0/TCP 69.229.23.54:5060;branch=z9hG4bK67a6e74fa0f95b090b70ff5d3d969e73;rport=6719 Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=17793 req_src_ip=69.229.23.54 req_src_port=6719 in_uri=sip:69.229.23.57:5060;transport=tcp out_uri=sip:69.229.23.57:5060;transport=tcp via_cnt==1" Content-Length: 0 Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here

3 2

[Serusers] Newbie question Best way to strip headers out of sip invites?
by Dale J. Robertson 02 Nov '05

02 Nov '05

I am looking for the best way to remove headers from INVITES sent to my PSTN gateway. The invites from my UA are too long and return '513's when they include authentication. My provider doesn't support BYOD so getting them to change maxlength is not an option. there are several headers that are sent by my ua that are not sent by the ua they have provided. when I strip the extra headers out using textops subst function it leaves an empty line which the pstn gateway doesn't seem to like very much. Is there some other method to strip headers? or maybe my regular expression syntax is wrong? the re I use to select the Date: header for removal is '^Date:.*$' replaced with // . I am testing my re's using sed. I must admit I know nearly nothing about using regular expressions. Thanks, Dale

2 1

[Serusers] Accounting
by harry gaillac 02 Nov '05

02 Nov '05

Hello, Does acc module allow these requests to a radius server according to rfc2866 ? A Talk Start information is sent when SIP Server receives 200 OK for a INVITE request. A TAlK Stop information when SIP Server receives BYE. How may i configure it ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

3 3

[Users] Help with MySQL acc query
by Daryl Sanders 01 Nov '05

01 Nov '05

I'm trying to come up with a query on the acc table that will return all INVITES that don't have a matching BYE. I'm basically looking for sessions that are currently open. - Daryl

6 11

[Users] Upgrading to 1.0.0
by Brandon Price 01 Nov '05

01 Nov '05

After upgrading from 0.9.5 to 1.0.0, openser won't start and I get a bunch of "!!!attention: from v1.0.0+ use 'return' instead of 'break'" in /var/log/messages. I figured, ok must be a syntax change so I replaced all instances of break; with return; and tried again. The messages persist. I have included sample messages and corresponding block of code. Thx in advance. Nov 1 10:43:42 ser1 openser: parse error (291,1-2): overwritting request routing table Nov 1 10:47:05 ser1 last message repeated 2 times line 291 is the closing } for route block 0. The same openser.cfg starts and works with 0.9.5. what the heck?

3 4

[Users] RE: [Serusers] voip charging by ISP's
by Vitaly Nikolaev 01 Nov '05

01 Nov '05

Well, I do not see a problem all depend on kind of network that you use, for example if ISP use cisco switches they can assign one of the ports on it as MONITOR for the port with interesting data (Vonage uplink :)) Then you connect something like call analyzer or simple unix server with ngrep running :) and make some nice filter that will catch all invites/byes that you need, or may be not invites but actually detect voip rtp streams and log it Ngrep can easily be changed to dump all SIP requests into database and then you can do what you want with it. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On Behalf Of Iqbal Sent: Tuesday, November 01, 2005 12:55 PM To: serusers(a)iptel.org Cc: users openser.org Subject: [Serusers] voip charging by ISP's Hi i was looking into a model where ISP may charge (SBC is already thiking of charging Vonage :-)), now if an ISP wanted to charge, all they would need is to extract the INVITE, BYE and all the stuff in the middle, to do some billing per VoIP provider sending traffic on there network, is anyone aware of any sniffers which could do this not from a proxy, but from the network level at a ISP end. Iqbal _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] RE: [Asterisk-Users] chan_exosip2
by harry gaillac 01 Nov '05

01 Nov '05

Thanks for reply, Does Chan_exosip2 is stable where may i find help ? Chan_sip from asterisk doesn't support IM and presence via SIMPLE. I whish to use either asterisk or openpbx to provide telephony features with SER to relay IM and presence SIMPLE openpbx or asterisk | | ----SER--- | | Sip agent sip agent What do you advise me ? Regards Harry --- Joshua Colp - Asterlink <joshnet(a)nbnet.nb.ca> a écrit : > Hello Harry, > > This is rather the wrong list to ask this... since > this is Asterisk, not > OpenPBX.org > > Chan_exosip2 though is something I'm basically > designing to have 3 operating > modes. > > Full server: Most closely resembles chan_sip in that > it acts as a B2BUA > Partial proxy: Extensions are mapped to SIP URIs and > it acts as a proxy. > Gateway: No authentication occurs (this is > presumably done outside by a SIP > proxy), incoming calls just get thrown into a > context. Outgoing calls are > down via SIP URI. > > Joshua Colp > > -----Original Message----- > From: asterisk-users-bounces(a)lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On > Behalf Of harry gaillac > Sent: Tuesday, November 01, 2005 10:19 AM > To: asterisk-users(a)lists.digium.com > Subject: [Asterisk-Users] chan_exosip2 > > Hello, > > I read roadmap on www.openpbx.org. > Does chan_exosip2 will be able to provide a real sip > proxy ? > What about asterisk solutions ? > > Harry > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users(a)lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users(a)lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] dev/null
by Iqbal 01 Nov '05

01 Nov '05

Hi Anyone know howto /dev/null inbound requests, from users. Reason for this is that I have a few random setups, which I no longer support, and the user has xlite and is just sending through REGISTER requests, I'd much rather they just get dumped, prefer to do it within ser, rather than some iptables/fw block. I guess I could have some blacklist table, and if From= blacklist then break...suggestions. Iqbal

3 3

[Serusers] UA's under same NAT
by deviator 01 Nov '05

01 Nov '05

Hello! i have 0.9.5 openser + mediaproxy 1.4.2 + serweb + mysql and so on, all works fine. private and public ua's works, no problems. But i have one question. Is it possible to send rtp traffic directly to each other if a callee and a caller are under same nat ? not through mediaproxy. p.s. huge thanks to the developers of ser/openser. Nice software! RESPECT!

2 1

[Serusers] Serweb not login
by Toygun Mavinil 01 Nov '05

01 Nov '05

Hi, İ cannot use serweb, it does not work My system is: Suse 9.3 Apache 2 Php 4 Ser 0.9.4 with mysql support Serweb 0.9.4 Ser Works with mysql support but serweb does not The login screen comes to the browser and if i use wrong user/pass it gives out wrong username or password but İf i use correct user/pass as admin - heslo it gives no errors and does nothing. Login screen comes again without any error First it worked for 5 minutes i could login but then i could not do anything else because it said "session expired, please login again" i can not also use serweb logging, pear log is not installed as pear package is installed Toygun Mavinil Mavinil Bilgisayar Sistemleri Toptan ve Perakende Satış Tel: 0 212 873 97 00 Faks: 0 212 873 97 03 Adres: Adakent Cd. No:28 Ak Apt. D.16 Beylikdüzü İstanbul Web: <http://www.mavinil.com.tr> www.mavinil.com.tr <http://www.adslbilgi.com> www.adslbilgi.com

1 0

[Users] MySQL 2-way replication with SER
by Daryl Sanders 01 Nov '05

01 Nov '05

Is anyone using MySQL replication 2-way replication (dual-master, multi-master) in your SER/OpenSER setup. How well does it work? Any pitfalls? - Daryl

3 2

RE: [Users] "uri == myself" matches when port is not port is different
by Vitaly Nikolaev 01 Nov '05

01 Nov '05

I also went thought the same situation and found the best solution (fastest) to keep second application (in my case b2bua) on different IP (alias) -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Tavis P Sent: Monday, October 31, 2005 7:34 PM To: users(a)openser.org Subject: [Users] "uri == myself" matches when port is not port is different This is a somewhat unique situation, but with my config i have some call features that are hosted on another SIP aware application running on the same system. In a branch_route block i am checking to see if destination is myself to avoid loops, however when i direct a message to, say "127.0.0.1:5090" and OpenSER is configured to listen on port 5060 only the "if (uri == mysql)" statment matches even though the port is different. It seems to me that this behaviour is not intuative given the use of the statment, can anyone else comment on this? _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Users] Tracker system
by Bogdan-Andrei Iancu 01 Nov '05

01 Nov '05

Hi everybody, I just finished configuring and testing the project tracker hosted on SorceForge. My apologies for the "un-explained" emails on the devel list :) The purpose of the tracker will be of course to have a centralized system to keep trace of : 1) bug reports 2) support requests 3) feature requests 4) documentation reports 5) patch submission If needed, other sections can be added. All developer have admin rights on this tracker (to create new categories or groups, etc). let's use it and start registering all known pending issues (discussed so far on the mailing list) in order to avoid forgetting them and to have a big picture of the current status. All changes regarding the status of the register items will be emailed to the devel mailing list. There are couple of items already registered by me - all project developers: please review them and update them accordingly if necessary! For example, for the enum patch, I think Juha will be the best person to assign it to. regards, bogdan

1 1

Re: [Users] Detecting runaway calls
by Daryl Sanders 01 Nov '05

01 Nov '05

Thanks for the info guys! It sounds like I need to do a little reading up on cseq to determine if this will even work, or find a PSTN gateway provider that supports Session-Timers. - Daryl On 10/27/05, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > by sending re-INVITEs from the middle of the path you will increase the > cseq number differently on each side...so you will need to synchronize > the cseq value when some in-the-dialog requests are passing through your > proxy....and that's quite complicated.... > > regards, > bogdan > > Daryl Sanders wrote: > > >Would it be possible to fake a REINVITE then check the response to > >determine if a call is still in session? Just brainstorming... > > > >- Daryl > > > > > >

7 16

[Users] lcr module
by Richard Z 01 Nov '05

01 Nov '05

Hi, I am trying to develop some features which requires to call load_gws and next_gw in a failure route. For example, in normal routing block, it forwards the call to a subscriber's ip phone locations. If nobody answer, it tries to call a pstn number. In the failure route, i called load_gws and next_gw. However it generates an error "next_gw(): No ruri_user AVP". Looking at the code, there seems to be an assumption that first time call to next_gw always happens in the normal routing block. But in my case, it calls next_gw first time in a failure route. I even tried to manually add the new uri's username to avp i:1402. It still won't work. Is it possible to call next_gw first time in a failure route? Thanks, Richard

2 3

[Users] Call not torn down when CANCEL sent before INVITE message is finished being processed
by Tavis P 01 Nov '05

01 Nov '05

With the latest CVS build of OSER, when an INVITE is sent followed by a CANCEL, before OSER has finished processing the initial INVITE, the outbound call is not torn down, and SER routes the CANCEL message to itself. Below is an ngrep trace of the exchange. I have a development cluster of mediaproxy servers setup and one of them was taken offline but left in the loop which increased the time it took for a message to be processed which is how i came across this situation (it took some seconds before the INVITE was able to be transmitted to the nexthop_proxy) -- # U 2005/10/04 01:44:43.134010 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-34d67b8f. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 101 INVITE. Max-Forwards: 70. Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:43.135736 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-34d67b8f;rport=5070;received=SIP_UAC. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>;tag=688bd636d693eb2ab143ccbaf0db2c7a.8474. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="hq.galnet.ca", nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e", qop="auth". Content-Length: 0. . # U 2005/10/04 01:44:43.147960 SIP_UAC:5070 -> OSER_PROXY:15061 ACK sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-34d67b8f. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>;tag=688bd636d693eb2ab143ccbaf0db2c7a.8474. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 101 ACK. Max-Forwards: 70. Contact: tavis <sip:200110005334@192.168.99.66:5070>. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 0. . # U 2005/10/04 01:44:43.154728 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="6867ef34181ba7cedbcbf8dd869f53ab",qop=auth,nc=00000001,cnonce="750e43e5". Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:43.648366 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="6867ef34181ba7cedbcbf8dd869f53ab",qop=auth,nc=00000001,cnonce="750e43e5". Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:44.155113 SIP_UAC:5070 -> OSER_PROXY:15061 CANCEL sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 CANCEL. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="512097281b4fba990adf2cc6c273f033",qop=auth,nc=00000002,cnonce="750e43e5". User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 0. . # U 2005/10/04 01:44:44.285994 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 483 Too many hops. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>;tag=688bd636d693eb2ab143ccbaf0db2c7a.32f9. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 CANCEL. Content-Length: 0. . # U 2005/10/04 01:44:44.648305 SIP_UAC:5070 -> OSER_PROXY:15061 INVITE sip:6046946333@OSER_PROXY:15061 SIP/2.0. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="200110005334",realm="hq.galnet.ca",nonce="4341dfb7e0cfaa398a6f425c02262325f943b20e",uri="sip:6046946333@OSER_PROXY:15061",algorithm=MD5,response="11a15f9ef4fdb1c9ab05566cad92d67e",qop=auth,nc=00000002,cnonce="750e43e5". Contact: tavis <sip:200110005334@192.168.99.66:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 287. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 192.168.99.66. t=0 0. m=audio 16410 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:44.688471 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.99.66:5070;branch=z9hG4bK-249e2787;rport=5070;received=SIP_UAC. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Content-Length: 0. . # U 2005/10/04 01:44:44.689012 OSER_PROXY:15061 -> NEXTHOP_PROXY:5060 INVITE sip:16046946333@NEXTHOP_PROXY SIP/2.0. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bK11ea.6c311116.0. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 15. Contact: tavis <sip:200110005334@SIP_UAC:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 288. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:45.396833 OSER_PROXY:15061 -> NEXTHOP_PROXY:5060 INVITE sip:16046946333@NEXTHOP_PROXY SIP/2.0. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bK11ea.6c311116.0. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. To: <sip:6046946333@OSER_PROXY:15061>. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. CSeq: 102 INVITE. Max-Forwards: 15. Contact: tavis <sip:200110005334@SIP_UAC:5070>. Expires: 240. User-Agent: Sipura/SPA3000-3.1.5(GWb). Content-Length: 288. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 25915303 25915303 IN IP4 192.168.99.66. s=-. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2005/10/04 01:44:46.090526 NEXTHOP_PROXY:5060 -> OSER_PROXY:15061 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bK11ea.6c311116.0. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 174. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 NEXTHOP_PROXY. t=0 0. m=audio 19004 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. # U 2005/10/04 01:44:46.104387 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 176. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. # U 2005/10/04 01:44:46.190376 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 176. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. ## U 2005/10/04 01:44:46.702880 OSER_PROXY:15061 -> SIP_UAC:5070 SIP/2.0 183 Inband Information. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, REGISTER. Via: SIP/2.0/UDP 192.168.99.66:5070;rport=5070;received=SIP_UAC;branch=z9hG4bK-249e2787. Record-Route: <sip:OSER_PROXY:15061;ftag=6e3e1f53c3f352efo0;lr=on>. To: <sip:6046946333@OSER_PROXY:15061>;tag=40032033. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=6e3e1f53c3f352efo0. Call-ID: ec5b9223-f98ffe5f(a)192.168.99.66. X-SHOUT-Channel: INBOUND 1:15:21:1. CSeq: 102 INVITE. Contact: sip:anonymous@NEXTHOP_PROXY. User-Agent: SHOUT 3.3.1p3v71. Content-Type: application/sdp. Content-Length: 176. . v=0. o=- 0 0 IN IP4 NEXTHOP_PROXY. s=SIP call via SHOUT. c=IN IP4 OSER_PROXY. t=0 0. m=audio 35328 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000.

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Re: [Users] Insert name into from header of SIP messages
by Brandon Price 01 Nov '05

01 Nov '05

Following that logic should I not be able to just replace the display and leave the uri as is with... uac_replace_from("$avp(s:cnam)",""); I have of coarse already trid that with no success. On Oct 28, 2005, at 3:38 PM, Norman Brandinger wrote: > If you break your callerid into 2 parts, you should be able to use > something like the following: > > AVP display: "John Doe" > AVP uri: "sip:18005551212@example.com" > > uac_replace_from("$avp(s:display)","$avp(s:uri)"); > > The above information has been documented at http://www.openser.org/ > docs/modules/1.0.x/uac.html > > Regards, > Norm > > Brandon Price wrote: > >> We are storing cnam (callerid name) values in a mysql database for >> use with openser. I have openser pulling the values from the >> database using avpops but I need to know how to insert the value >> into the from so that it will be passed to my sip user agents. >> Related code posed below >> >> modparam >> ("avpops","db_scheme","cnam_scheme:username_col=phonenumber;table=cna >> m_cache;value_col=cnam;value_type=string") >> modparam("avpops", "avp_aliases", "cnam=s:cnam") >> >> avp_db_load("$from/cnam_scheme","cnam/$cnam_scheme"); #sample >> query -- select cnam from cnam_cache where phonenumber='13143212222' >> >> this should now store that value in $cnam. >> now how do I put that value for example "John Smith" in the from >> portion of sip messaging so that it displays on the users sip >> phone, or phone attached to an ata? >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >

4 5

[Serusers] Fwd: DIDX.org - DIDXchange
by Rehan Ahmed 01 Nov '05

01 Nov '05

---------- Forwarded message ---------- From: Rehan Ahmed <rehan33(a)gmail.com> Date: Nov 1, 2005 8:02 AM Subject: DIDX.org - DIDXchange To: serusers(a)lists.iptel.org Hello All, I wanted to know if someone here is using did numbers from DIDX.org and able to make them work on SER yet. Rehan

1 0

[Serusers] when I include mysql module in ser.cfg, UA receive "483, too many hops", any advice?
by fishman7788 01 Nov '05

01 Nov '05

I have use ser_mysql create to setup database,and can connect to ser,can add a user using serctl add,but UA just can't receive correct register message,any suggestion? Thanks regargs free

1 0

[Serusers] ser.cfg parse error
by Ma Zhiyong 01 Nov '05

01 Nov '05

Hi All, After I installed the latest CVS version, I found SER could't be started by serctl utility. It shows # serctl start Starting SER : 0(14877) read 3820494812 from /dev/urandom 0(14877) seeding PRNG with 656407867 0(14877) test random number 38195676 # 0(14877) parse error (1,1-2): parse error .......... It seems that my ser.cfg file has wrong format or other problem. Any one has the same matther before? Does that mean my ser.cfg file encounters "parse error" at line 1 and coloum 1 to 2? Thanks.

1 0

Re: [Serusers] Error with acc.so and db funtions
by zhangshuai 01 Nov '05

01 Nov '05

Dear Antonio Membrives, Did you see the makefile of acc module and recompile acc.so? >Hi again i have this lines in ser.cfg > >modparam("acc", "log_flag", 1) >modparam("acc", "log_level", 2) >modparam("acc", "db_flag", 1) >modparam("acc", "db_missed_flag", 3) >modparam("acc", "log_missed_flag", 3) >modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") >modparam("acc", "report_ack", 0) >modparam("acc", "log_fmt", "miocfsp") > >for store missing calls in serweb, but when I launch the ser services >received 3 errors and see this in /var/log/messages > >Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_flag> not found >in module <acc> >Oct 31 13:50:00 dns2 ser: parse error (73,19-20): Can't set module parameter > >Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_missed_flag> >not found in module <acc> >Oct 31 13:50:00 dns2 ser: parse error (74,19-20): Can't set module parameter > >Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_url> not found >in module <acc> >Oct 31 13:50:00 dns2 ser: parse error (76,20-21): Can't set module parameter > >I have fedora 3+ser 0.9.3 , any idea of this. > >Thanks. > >Antonio Membrives > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > = = = = = = = = = = = = = = = = = = = =

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[Users] "uri == myself" matches when port is not port is different
by Tavis P 01 Nov '05

01 Nov '05

This is a somewhat unique situation, but with my config i have some call features that are hosted on another SIP aware application running on the same system. In a branch_route block i am checking to see if destination is myself to avoid loops, however when i direct a message to, say "127.0.0.1:5090" and OpenSER is configured to listen on port 5060 only the "if (uri == mysql)" statment matches even though the port is different. It seems to me that this behaviour is not intuative given the use of the statment, can anyone else comment on this?

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[Serusers] Please note the "HOW TO" for MYSQL connect issue
by Dhiman D Chowdhury 31 Oct '05

31 Oct '05

Dear friends, If you are like me faced with an issue describe below and could not find the answer for it. Please note the recommendation below. setup: Redhat Linux 9.0 and Mysql 3.23 Issue: 1. while trying to connect from shell prompt; #mysql Eorr 1045: Access denied for root@localhost (using password: No). Note: you will see this message in different times perhaps when using: # mysql.sh create or similar commands. Recommendation: There is a bug in Mysql version 3.23: Please visit the url below: http://bugs.mysql.com/bug.php?id=11822 To overcome the issue: Simply use the following command: mysql -h 127.0.0.1 -u root -p [issue it as needed] Thanks Dhiman

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[Users] problems with mediaproxy
by Gustavo Hernandez Baratta 31 Oct '05

31 Oct '05

Hi all! I´m trying to setup mediaproxy but making calls this error occur with proxydispatcher: Starting SER MediaProxy server: mediaproxyListening for commands on local socket `/var/run/proxydispatcher.sock' Default mediaproxy server is on `XX.XXX.XXX.XXX:25060' : : ---------------------------------------- Exception happened during processing of request from Traceback (most recent call last): File "/usr/lib/python2.2/SocketServer.py", line 458, in process_request_thread self.finish_request(request, client_address) File "/usr/lib/python2.2/SocketServer.py", line 253, in finish_request self.RequestHandlerClass(request, client_address, self) File "/usr/lib/python2.2/SocketServer.py", line 514, in __init__ self.handle() File "/usr/local/mediaproxy/modules/dispatcher.py", line 303, in handle result = self.execute(command) File "/usr/local/mediaproxy/modules/dispatcher.py", line 347, in execute proxy = MediaProxy('domain://%s' % domain).address File "/usr/local/mediaproxy/modules/dispatcher.py", line 578, in __init__ self.address = proxy File "/usr/local/mediaproxy/modules/dispatcher.py", line 583, in __setattr__ if name in readonly and self.__dict__.has_key(name): TypeError: 'in <string>' requires character as left operand ---------------------------------------- : ---------------------------------------- Exception happened during processing of request from Traceback (most recent call last): File "/usr/lib/python2.2/SocketServer.py", line 458, in process_request_thread self.finish_request(request, client_address) File "/usr/lib/python2.2/SocketServer.py", line 253, in finish_request self.RequestHandlerClass(request, client_address, self) File "/usr/lib/python2.2/SocketServer.py", line 514, in __init__ self.handle() File "/usr/local/mediaproxy/modules/dispatcher.py", line 303, in handle result = self.execute(command) File "/usr/local/mediaproxy/modules/dispatcher.py", line 347, in execute proxy = MediaProxy('domain://%s' % domain).address File "/usr/local/mediaproxy/modules/dispatcher.py", line 578, in __init__ self.address = proxy File "/usr/local/mediaproxy/modules/dispatcher.py", line 583, in __setattr__ if name in readonly and self.__dict__.has_key(name): TypeError: 'in <string>' requires character as left operand ---------------------------------------- Ideas? Thanks a lot, gus

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[Users] version 1.0.0 released
by Daniel-Constantin Mierla 31 Oct '05

31 Oct '05

Hi everybody, the version 1.0.0 has been released. The branch 'rel_1_0_0' is created and must be used by anybody wanting the get v1.0.0 out from CVS (cvs co -r rel_1_0_0 sip-server). There are a lot of new features in this release, starting from TLS, pseudo-variables in core and continuing with enhancements in avpops and other module. This release is the first open source sip server with TLS support, although there is still to work on it (e.g., multi domain support) it allow secure sip signaling communications between servers and clients. An unified format of pseudo-variables has been adopted in core and many modules, basic functionality for future dialog support is present in RR module, avpops includes functions for string operations (concatenation, prefix, suffix, substitute), arithmetic and bitwise operations, among others. A new module, uac_redirect, has been introduced to handle redirect replies (3xx) on server, with proper accounting and filter rules. See the full list of new features at: http://www.openser.org/release-1.0.0.php A roughly change log can be browsed online at: http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/ChangeLog?view=mar… There are two categories of released packages: - classic openser, without TLS support - the name of the package does not contain the -tls- word - TLS-ed version of openser - the name of the package contain the -tls- word The tarball with the sources can be downloaded from: http://openser.org/pub/openser/1.0.0/src/ The basic binary packages are deployed to the download site: http://openser.org/pub/openser/1.0.0/bin/ http://openser.org/pub/openser/1.0.0/packages/ We are still collecting other packages from contributors which already volunteered, so expect new distributions to be represented there in the next days. The module documentations are up-to-date, the site and tutorials need a quick review in the next days.

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[Serusers] Too many hops
by Philip Jones 31 Oct '05

31 Oct '05

I am using a dedicated server to host SER, when I use the actual IP address of server i/f I connect no problem but, using the domain name, the client times out with "Too many hops". Is there a known solution to this? I would really prefer to use the domain name. Phil

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[Serusers] REFER with Replaces header via FIFO - Invite drops Replaces
by Brian Roy 31 Oct '05

31 Oct '05

Short and sweet: I am attempting to REFER and existing call to another UA using a REFER with an imbedded REPLACES header in the REFER-TO header. FIFO accepts the REFER and the call is set up, however the REPLACES header is dropped between the REFER and the INVITE. 1) Has anyone been able to get a REFER with REPLACES to work with SER? a. Am I doing something completely friggin' stupid?? 2) Is this a SER bug. or a FIFO bug. or neither? See Below: Initial Call: *************************You'll note that replace is supported*************************************** INVITE sip:1999206@maggie-may.home.brianandkelly.ws:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bK5d9b3bcbFDFC6408 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060> CSeq: 1 INVITE Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130777573 1130777573 IN IP4 192.168.0.161 s=Polycom IP Phone c=IN IP4 192.168.0.161 t=0 0 m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 ACK sip:206@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKeb569c391EDD1A0 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060>;tag=as1624b103 Route: <sip:192.168.0.24;ftag=2A63079D-99C40A74;lr=on> CSeq: 1 ACK Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Max-Forwards: 70 Content-Length: 0 Attempted REFER: ***************************************Note that that Replaces info drops in the INVITE*********************************************************** REFER sip:kellydev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.24;branch=z9hG4bK3915.d2aefa37.0 To: sip:briandev@maggie-may.home.brianandkelly.ws From: sip:kellydev@maggie-may.home.brianandkelly.ws;tag=1f850894c564e190ae8e24 413352beae-4b42 CSeq: 10 REFER Call-ID: 2bbbc774-0(a)192.168.0.24 Content-Length: 0 User-Agent: Sip EXpress router(0.9.4 (i386/linux)) Refer-To: sip:briandev@maggie-may.home.brianandkelly.ws?Replaces=74e15821-cc9958e7 -2e639266(a)192.168.0.161;to-tag=as1624b103;from-tag=2A63079D-99C40A74 INVITE sip:briandev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK880b0a28917F93C5 From: "Kelly Roy - DEV" <sip:kellydev@maggie-may.home.brianandkelly.ws:5060>;tag=519BC0C2-649892 E1 To: <sip:briandev@maggie-may.home.brianandkelly.ws> CSeq: 1 INVITE Call-ID: 801c0ab6-44e87514-ba253cb(a)192.168.0.160 Contact: <sip:kellydev@192.168.0.160:5060;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130775178 1130775178 IN IP4 192.168.0.160 s=Polycom IP Phone c=IN IP4 192.168.0.160 t=0 0 m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685) Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685)

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[Serusers] REFER with Replaces header via FIFO - Invite drops Replaces
by Brian Roy 31 Oct '05

31 Oct '05

Short and sweet: I am attempting to REFER and existing call to another UA using a REFER with an imbedded REPLACES header in the REFER-TO header. FIFO accepts the REFER and the call is set up, however the REPLACES header is dropped between the REFER and the INVITE. 1) Has anyone been able to get a REFER with REPLACES to work with SER? a. Am I doing something completely friggin' stupid?? 2) Is this a SER bug. or a FIFO bug. or neither? See Below: Initial Call: *************************You'll note that replace is supported*************************************** INVITE sip:1999206@maggie-may.home.brianandkelly.ws:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bK5d9b3bcbFDFC6408 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060> CSeq: 1 INVITE Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130777573 1130777573 IN IP4 192.168.0.161 s=Polycom IP Phone c=IN IP4 192.168.0.161 t=0 0 m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 ACK sip:206@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKeb569c391EDD1A0 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060>;tag=as1624b103 Route: <sip:192.168.0.24;ftag=2A63079D-99C40A74;lr=on> CSeq: 1 ACK Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Max-Forwards: 70 Content-Length: 0 Attempted REFER: ***************************************Note that that Replaces info drops in the INVITE*********************************************************** REFER sip:kellydev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.24;branch=z9hG4bK3915.d2aefa37.0 To: sip:briandev@maggie-may.home.brianandkelly.ws From: sip:kellydev@maggie-may.home.brianandkelly.ws;tag=1f850894c564e190ae8e24 413352beae-4b42 CSeq: 10 REFER Call-ID: 2bbbc774-0(a)192.168.0.24 Content-Length: 0 User-Agent: Sip EXpress router(0.9.4 (i386/linux)) Refer-To: sip:briandev@maggie-may.home.brianandkelly.ws?Replaces=74e15821-cc9958e7 -2e639266(a)192.168.0.161;to-tag=as1624b103;from-tag=2A63079D-99C40A74 INVITE sip:briandev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK880b0a28917F93C5 From: "Kelly Roy - DEV" <sip:kellydev@maggie-may.home.brianandkelly.ws:5060>;tag=519BC0C2-649892 E1 To: <sip:briandev@maggie-may.home.brianandkelly.ws> CSeq: 1 INVITE Call-ID: 801c0ab6-44e87514-ba253cb(a)192.168.0.160 Contact: <sip:kellydev@192.168.0.160:5060;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130775178 1130775178 IN IP4 192.168.0.160 s=Polycom IP Phone c=IN IP4 192.168.0.160 t=0 0 m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685)

1 0

[Serusers] Windows Messenger 5.1 On-line
by gorand＠dvvti.com 31 Oct '05

31 Oct '05

I am testing Windows Messenger 5.l and it works fine except when I connect other users that are on the SER box, it does not show them as on-line. It displays them as off-line. I am able to make sip calls through the SER Box. Everything works fine in that aspect. I want to use the Windows Messenger as a communication aspect to be able to send instant messages to the users. Please let me know if this possible with this Windows Messenger? Thanks.

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[Users] How to remove fields from SQL accounting?
by Federico Giannici 31 Oct '05

31 Oct '05

I need to customize the data logged with the SQL accounting. The "db_extra" parameter let me ADD new fields but it seems that there is no way to REMOVE fields from the sql command. Infact, the "log_fmt" seems to affect only the syslog accounting. Is there a way to prevent a field to be included into the SQL INSERT command? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] Handling of REFER messages
by Miroslaw Nawrot 31 Oct '05

31 Oct '05

Hi, I am trying to do a peculiar task and I want to find out if SER can do it. The issue is with call transfers which generate the REFER message. I am connecting a Nortel voip switch to Asterisk via sip. I need to have all call transfers generated by the Nortel switch to not reach Asterisk as all the call transfers are done to dids/extensions not found on the Asterisk PBX. Instead I would like to intercept / re-route them back to my Nortel switch which can handle it. Currently I have a workaround using an Audiocodes gateway with loopback cables. The Audiocodes when sees the transfer request sends it back to the Nortel switch. As this is a wasteful use of expensive hardware I would like to find another method to do this. Can someone let me know if this is possible with SER? And if it is which functions do I use? Any help is greatly appreciated. Mirek The Future of Voice Globalive Communications Corp. T: 416-640-1088 F: 416-640-1089 60 Adelaide St. E, 6th Floor Toronto, ON, M5C 3E4 www.globalive.com

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[Serusers] Error with acc.so and db funtions
by Antonio Membrives 31 Oct '05

31 Oct '05

Hi again i have this lines in ser.cfg modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsp") for store missing calls in serweb, but when I launch the ser services received 3 errors and see this in /var/log/messages Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_flag> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (73,19-20): Can't set module parameter Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_missed_flag> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (74,19-20): Can't set module parameter Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_url> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (76,20-21): Can't set module parameter I have fedora 3+ser 0.9.3 , any idea of this. Thanks. Antonio Membrives

1 0

[Serusers] Error with msilo, dont work
by Antonio Membrives 31 Oct '05

31 Oct '05

Hi to all, I try to use MSILO in Fedora 3 + Ser 0.9.3 and received the next message in /var/log/messages /var/log/tail messages Oct 31 09:54:15 dns2 ser: WARNING: fix_socket_list: could not rev. resolve x.x.x.x Oct 31 09:54:15 dns2 ser[8823]: Maxfwd module- initializing Oct 31 09:54:15 dns2 ser[8823]: bind_dbmod: Module mysql//ser does not export db_use_table function Oct 31 09:54:15 dns2 ser[8823]: init_mod(): Error while initializing module msilo And ser not work, if I coment the line of msilo db_url work again but not work msilo. Any idea? Antonio Membrives. #ser.cfg debug=3 fork=yes log_stderror=no listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE alias=dominio.es alias=x.x.x.x port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) 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("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registar@telintcable.es") modparam("msilo", "db_url", "mysql//ser:heslo@localhost/msilo") #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 2) #modparam("acc", "log_missed_flag", 3) #modparam("acc", "log_fmt", "fimos") #modparam("acc.", "db_url","mysql://ser:heslo@localhost/ser") #modparam("acc", "db_flag", 2) #modparam("acc", "db_missed_flag", 3) modparam("tm","fr_inv_timer",30) modparam("tm","fr_timer",3) /var/log/tail messages Oct 31 09:54:15 dns2 ser: WARNING: fix_socket_list: could not rev. resolve x.x.x.x Oct 31 09:54:15 dns2 ser[8823]: Maxfwd module- initializing Oct 31 09:54:15 dns2 ser[8823]: bind_dbmod: Module mysql//ser does not export db_use_table function Oct 31 09:54:15 dns2 ser[8823]: init_mod(): Error while initializing module msilo

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[Serusers] where to find serweb tool
by dave cra 31 Oct '05

31 Oct '05

Hi all, I've tried to get the serweb tool off the iptel website and it can not find it. Is there another place to obtain this? I'm running windows 2000. Cheers Dave --------------------------------- Yahoo! FareChase - Search multiple travel sites in one click.

2 1

[Users] pa module
by Schweizer Laurent 31 Oct '05

31 Oct '05

Hello, I try to setup the pa module and I have some errors. The pa always send a NOTIFY with a status set to CLOSED. I have also see in some olds mails that it is possible to add an xcap server, what is doing this server? Is a free xcap server available? Thanks Laurent My config: modparam("pa", "default_expires", 1800) modparam("pa", "timer_interval", 10) modparam("pa", "use_db", 0) # def=0 disable db support modparam("pa", "use_place_table", 1) # def=0 disable use of place table modparam("pa", "use_bsearch", 0) # def=0 disable use of binary search for location modparam("pa", "use_location_package", 0) # def=0 disable use of location package #modparam("pa", "pa_domain", "bigU.edu") # def name of xcap server modparam("pa", "db_url", "mysql://..........") modparam("pa", "presentity_table", "presentity") modparam("pa", "presentity_contact_table", "presentity_contact") modparam("pa", "watcherinfo_table", "watcherinfo") modparam("pa", "place_table", "place") modparam("pa", "new_watcher_pending", 0) # def=0 default state ofwatchers 0=active modparam("pa", "pidf_priority", 1) # def=1 default priority of contacts in pidf documents

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[Serusers] fee-for-service SER server - clarified
by Michael Kendall 31 Oct '05

31 Oct '05

Greger, Thanks so much for your reply to my question, and thanks also to Olivier, Bravo, Iqbal, and Phil, who each sent me some great SER server recommendations or requested clarification to my confusing question. You are right that I did not provide enough info in my question. I do not need any of the example services that you mentioned in a, b, c, d. I don't want to control or customize any SER server, I only want to be a normal customer of an existing SER server. I do not need any access to regular telephones. What I want to do is have a call center with 10 tutors, with each tutor connected to a different remote student, The 10 teachers and 10 students would each be using the Windows (SIP) Messenger softphone and would find and call each other using the SER server. The SER server would need to allow media-proxy service so that we can traverse NATs and firewalls. I have selected Windows (SIP) Messenger as my softphone because it is cost-free (quite important if I have 300 students) and because it has a whiteboard (required for the tutoring). If someone can suggest a better softphone which has a whiteboard I would be very happy, especially if the softphone supports STUN. If the suggested softphone is free, that would be perfect. I can't really afford to use a softphone which charges me a fee for each named user, but if I need to pay a reasonanable price for each SIMULTANEOUS user, that would be ok. Most of the SER server websites that I have visited seem to recommend the Xten softphone, but since it has no whiteboard, I cannot use it. Windows (SIP) Messenger does not appear to support STUN, so media-proxy is the only way I know to traverse NATs. (Please correct me if I am wrong about Messenger not supporting STUN.) I know that the newest versions of Windows (SIP) Messenger do not allow connecting to 3rd party servers, so I plan to use the older versions that do allow it. Some of the SER server websites ( like fonosip ) claim (after a search) that they support Windows (SIP) Messenger, but they do not offer any configuration instructions. The fonosip site also does not say whether they will offer media-proxy services, which I need for NAT traversal. It is strange that these SER service provider websites provide very little info about whether they support media-proxy or how to configure a softphone for STUN. I know that media-proxy places a heavy load on the SER server, and that some SER servers are therefore reluctant to consume their expensive bandwidth by offering media-proxy services. But since there must be many other customers who are willing to pay for media-proxy service, there should be a group of SER servers who are offering this service for a fee. I just need to find one. Can someone please suggest a SER server which might satisfy the Messenger and media-proxy requirements? And can someone recommend a better softphone with whiteboard? thanks, Michael Greger wrote: Dear Michael, I know that many on this list operate own services they sell in the market. Others sell software packages and services, both to corporations (ex. Asterisk + ser in a combo setup) and to service providers (some sort of white-label service). There are also hosted services for enterprises that are not based on SER. I would suspect that lack of answer may have something to do with lack of information. You are not saying what you need. Here are some examples: a. A hosted SER server where you can log in and do changes to ser.cfg (for fun or commercial) b. A hosted corporate PBX-type solution c. A hosted SIP service ? la university setups with ENUM lookup and authentication against your own user database d. A white-label service that you will brand as your own and resell in a given market And BTW, you are not saying anything about your requirements (just a vanilla SER server?!) nor the size of the subscriber base you want to support etc etc. If you are looking for b, you are probably better off looking in your local market for a telephony provider who can give you IP access. g-)

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[Serusers] need a help!!!
by deviator 31 Oct '05

31 Oct '05

Hello! I'm using openser 0.9.5 with rtpproxy, i'm found a good config for this on onsip.org site. All works fine with public ips ua's, but i have a problem with nat :( server have a public ip, ua's have a private ips, symmetric nat. I have an acknowledgement error (ACK Timeout) and calls failure after 30-32 sec. What should i change or add in openser.cfg (nat-rtpproxy.5.0.cfg version) pls help!!!

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[Serusers] When to use MediaProxy
by tcchan 31 Oct '05

31 Oct '05

Hi All, I have SER work together with MediaProxy and two UAs; UA1 is NATED whereas UA2 has public IP. I have something like this to process the INVITE if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; t_on_reply("1"); and on the reply_route, I have if (client_nat_test("1")) { setflag(4); }; if (isflagset(4) && status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { fix_contact(); use_media_proxy(); } }; Now, if UA1 (nated) calls UA2, it is OK ( mediaproxy is properly setup and both side can hear each other. The client_nat_test() is met and flag4 is set which caused both the invite and the on_reply to use_media_proxy). If UA2(public) calls UA1 (nated), problem arises, UA2 can hear UA1 but not the reverse. The client_nat_test("3") is not met and thus flag4 not set. On the reply_route, although client_nat_test("1") is met, flag4 set and use_media_proxy() is executed, it return error: use_media_proxy(): empty response from mediaproxy ERROR: on_reply processing failed To temporary work around this problem, I forcefully set flag4 on and thus all calls route through the mediaproxy ( which is expensive in bandwidth usage). Is there any better way to solve this? Any suggestion? Has anybody already looks into this and has a solution? Thanks for any help available. Regards, TC Chan

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[Serusers] Inserting SER into my voice network
by Ray Van Dolson 31 Oct '05

31 Oct '05

First off, here is my voice network layout currently: http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png We're using Asterisk for voicemail, call routing (for long-distance, LNP, etc) and SER/rtpproxy at the other end which handles NAT onto private networks where customer's exist. This setup works fairly well for the most part, except that Asterisk does not have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy) for their jitter buffer on both ends of our voice links here. To me, that would mean shoving a SER/rtpproxy combo between Asterisk and our provider network. Possibly on the same server. I could easily throw up a SER installation, but I'm trying to figure out if there's any way to leave Asterisk in the SIP path but remove it from the media path (have RTP just go straight from SER/rtpproxy to my provider's RTP proxy). Have any of you set up a scenario somewhat like this? Any recommendations? Asterisk CVS-HEAD does let you apply a patch and get some very alpha jitter buffer support. But CVS HEAD doesn't work reliably for me at all currently so I'm sticking with the latest stable release of 1.0.x series. I'm running SER 0.9.3 FYI on the SER proxies I have set up currently. Thanks. -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

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[Serusers] Please advise...
by Stanly Levithan 30 Oct '05

30 Oct '05

Dear SER-users, We are looking to establish an educational business, namely a virtual learning center,using a conference service providing capabilities with the following calls flow: --Caller dial a local number -- Media gateway/media server (MG/MS)accepts the call and automatically mutes it so all the callers become passive listeners if number is registered in the Data Base; if the number is not registered the call is discarded --MG/MS plays Welcome message --MG/MS asks for a virtual conference room number --Caller enters virtual conference room number --Caller is entered into virtual conference room associated with the virtual conference room number. All callers entered virtual conference room number are now in associated virtual conference room --MG/MS accepts any callers decision to switch to a different virtual conference room by entering appropriate virtual conference room number without making another call (without leaving the system) --MG/MS permits the caller to terminate the call at will --MG/MS of our dream has to be able to process up to 50, 000 callers simultaneously --MG/MS provides billing. Kindly let us know if SER is capable of supporting such a calls flow. Please let us know if you need any additional information in order to advise us on a most efficient configuration. Regards, Stan *** This email was scanned by eSafe Content Inspection Server. *** __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

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[Serusers] Please advise...
by Stanly Levithan 30 Oct '05

30 Oct '05

Dear SER-users, We are looking to establish an educational business, namely a virtual learning center, using a conference service providing capabilities. Kindly take a look at our business requirements described in the attachment and let us know if SER is capable of supporting such a calls flow. Please let us know if you need any additional information in order to advise us on a most efficient configuration. Regards, Stan *** This email was scanned by eSafe Content Inspection Server. *** __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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[Serusers] SER installation procedure for Redhat linux 9.0: Please note this
by Dhiman D Chowdhury 30 Oct '05

30 Oct '05

Dear friends, If you are installing SER for the first time in REDHAT LINUX 9.0, please do the followings: 1. download ser-0.9.3_linux_i386_tar.gz 2. save it to a user directory 3. login as root and go to / directory 4.from "/" as root run the following command: tar -zxvf /home/user/ser-0.9.3_linux_i386_tar.gz [replace the /home/user directory with the directory name where you have initially downloaded the ser file. Please note ser-0.9.4_linux_i386.tar.gz will not work if you have downloaded it. For ser 0.9.4, you need to make binary by yourself. 5. Now run /usr/local/sbin/ser 6. Your SER server is running. 7. At this point you can download xten pro sofphone client and configure it to connect to your serv server. 8. Just to check out whether or not xten is registering with the server use admin as user and heslo as password. Now you are ready to connect mySQL server and follow the procedure for that. Thanks Dhiman Deb Chowdhury

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[Users] Making OpenSER give up being man-in-the-middle?
by Juan Carlos Castro y Castro 30 Oct '05

30 Oct '05

Hello, I'm using OpenSER to make load balancing among a bunch of Asterisk boxes. I whipped up a very simple script that works: # module stuff goes before here 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; }; route(1); break; } route[1] { exec_dset("/usr/local/bin/selectserver.sh"); if (!t_relay()) { sl_reply_error(); }; } selectserver.sh alters the server in the destination URI to choose from a dynamic pool of servers that's always changing according to a funky proprietary logic of ours. This works just peachy. The problem is OpenSER stays as man-in-the-middle all the time. I'd like it to do a REFER to a random server of our choosing and stay out of the loop from there on. How could I do that?

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[Serusers] Ser v0.9.4 error message " /lib/tls/libc.so.6: version `GLIBC_2.3.4' not found (required by ser)
by Dhiman D Chowdhury 30 Oct '05

30 Oct '05

Hi, I am using redhat linux 9.0 and installed ser v0.9.4. While I am trying to start ser, I get the following message: /lib/tls/libc.so.6: version `GLIBC_2.3.4' not found (required by ser) I will appricate your kind help in this regard. Thanks Dhiman Chowdhury

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[Users] usrloc-cl with openser
by Daryl Sanders 30 Oct '05

30 Oct '05

Greetings Everyone, Has anyone tried the Andreas' cacheless usrloc module with openser. Will it run on openser? I am thinking of testing the new openser release and I like to have this functionality if possible. Thanks! - Daryl

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[Serusers] Information on how to submit code to experimental
by Greger V. Teigre 30 Oct '05

30 Oct '05

To those of you who have developed code you are evaluating whether to commit to cvs, here is information found in the new HACKING file found in the experimental cvs directory. If you have any immediate questions or comments, I'm all ears! g-) HACKING SER =========== This information is for people who have or would like to develop a module or code for SER and would like to submit the code to the SER CVS tree. Please first read the README file in the experimental directory for information about the purpose of the directory and what type of code you can submit. If you have decided to go ahead, the information below is what you need to get started! Administrator ============= The administrator of the experimental directories is Greger V. Teigre. He can be contacted at greger(a)onsip.org. If you have any questions after reading this or would like to submit code, please send an email! Registration ============ As a new SER developer you need to register an account at http://bugs.sip-router.org/ and at http://developer.berlios.de/account/register.php. Use the same username for both. Requesting Access ================= In order to get write access to the experimental CVS you need to send information about the module/code to the administrator (see above), as well as your full name, and the username and email address used for the above registration. Information about parameters and function calls, the version of SER the module has been developed for, and one or more use cased for the code will be of help (unless it's obvious...). You should also include a 3-5 lines description of the code/module that you would like to include in the experimental README. You will then be advised on how to make sure your code is a use for others and if others are working on/requested submission of similar code. The purpose of this check is to make submitted code as easily accessible as possible and avoid overlapping coding efforts. Once approved, you check out using: cvs co experimental. You will get all the experimental modules, your module should have an empty directory where you have write CVS access. Development Requirements ------------------------ You should read this general info on the SER CVS: http://www.iptel.org/ser/cvs/ And to understand how SER is organized, you should read the first part of: http://www.iptel.org/~janakj/ser_cvs.xhtml We strive to keep the requirements as few as possible (we don't like overhead more than you do), but there are some rules you must follow: - You are responsible for maintaining the module for any branches you have submitted the module in. This includes keeping it updated as HEAD and the branches are updated - Bug and issue tracking MUST be kept on http://bugs.sip-router.org/ - If a module is not being maintained, it will be moved to the "attic" and eventually deleted - All experimental modules should be checked in with paths as if they live in the main modules directory - All experimental modules should have an updated README - All experimental modules should have #warning directives that will be shown when making, as well as a LOG(L_ALERT, "WARNING! ..."); in the initialization function of the module. This warning should also be added to the README file. Recommended warning template: "This module is experimental and may crash SER or create unexpected results. You use the module at your own risk. Please submit bugs at http://bugs.sip-router.org/" - IF YOUR MODULE REQUIRES PATCHING OF CORE OR OTHER MODULES: If the module requires a patch to the core tree, the README file and compile warning should explicitly warn about this and describe how to go about patching the tree (reference to install.sh below and README). Patch(es) should be included in a separate sub-directory called patches/. The README should have instructions on which changes will be made, and the maintainer should keep an updated install.sh that will apply the patches in the patches/ directory (assuming that the module is located in the ser modules directory after checkout by the user). This script may include patching code in root/other modules, as well as making a new directory in root and copying files from the modules/modulename/patches/ directory to the root. IF the contributed code is NOT a module, but a core extension/patch, the script should make the proper tree modifications, should NOT include a Makefile (i.e. a module makefile), and should leave its own directory intact after patching. - If and when the module has proven its usefulness to a broad range of users and its stability (and the developer his/her ability to support and maintain the module), it may be promoted to a standard module status. This is a decision made by the core developers. An application can be sent to serdev(a)lists.iptel.org if and when the maintainer finds the module applicable to such status

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[Serusers] Information on the experimental directory
by Greger V. Teigre 30 Oct '05

30 Oct '05

FYI here is the newly updated README from the experimental directory. The first section explains the purpose of the directory. g-) The Purpose of The Experimental Directory ========================================= The experimental directory has been created to lower the threshold for adding new code to SER, while keeping the main code tree robust. Experimental exists both in CVS head (the main development branch), as well as in release branches (ex. rel_0_9_0). In the CVS head, experimental can be used for patches to the main code tree, as well as for modules that extend SER's functionality. Any code will be excepted, provided that the code give some generic value and one or more developers are willing to maintain the code. Code that proves itself popular and stable will be moved to the main code tree. In a release branch, experimental can be used to add new functionality to an already released version. Code will never be allowed into the main code tree (because the code has been freezed), but people can add experimental features to the stable release. This way limited functionality can be added without having to use the CVS head, which will have lots of (possibly untested) functionality added. Often modules that have been developed for CVS head will be "backported", i.e. made to work with a previous, stable version. Also, some developers may use a stable release and have developed a module for that version. This module can be submitted to experimental of a released version to facilitate the introduction of the new code into SER. (NOTE that ALL code submitted to a stable version MUST be ported to also work with CVS head. Allowing a module only into the experimental directory of a stable release is just time-limited until porting has been done.) What You Will Find in Experimental ================================== The experimental directory may contain three types of code: 1. *Simple modules* that can be dropped into main code tree 2. *Complex modules* that require patching of the main tree 3. *Code patches* (no module) that modify the core to accomplish new functionality Submitting Bug Reports ====================== For experimental code to enter the main code tree, it needs to be used, tested, and bugs/feature requests most be reported. If you use code from the experimental directory,remember that the code maintainer will love to hear about your experiences! Contact the code maintainer directly (see information below). If you have found a bug or have a feature request, please use: http://bugs.sip-router.org/ ============================= Description of Modules: ============================= path module, simple module ----------- COMMENT:The path module has been ported into the main code tree by Andreas Granig. This module will thus be removed as this effort has been finished. Maintainer: Fermin Galan Marquez <fermin.galan(a)agora-2000.com> This module implements the Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts, as described in RFC 3327 tls, code patches ---------- Maintainer: Cesc Santasusana <cesc.santa(a)gmail.com> TLS is an optional part of the core, not a module. TLS, as defined in SIP RFC, is a mandatory feature for proxies and can be used to secure the SIP signalling on a hop-by-hop basis (not end-to-end). TLS works on top of TCP (DTLS, or TLS over UDP is already defined by IETF and may become available in the future). usrloc-cl module ----------- COMMENT: This module is currently only found in rel_0_9_0 branch. Work is in progress to port the module to CVS head. Maintainer: Andreas Granig <agranig(a)linguin.org> This is a cacheless implementation of the usrloc API and replaces the original usrloc module. It provides access to domain tables (like location and aliases) to other modules. The module exports no functions that could be used directly from scripts. The typical use case for this module over the original usrloc module is that one wants to replicate the tables on the database layer without the need of SIP replication. This allows better scalability at the cost of performance. osp module ---------- COMMENT: The maintainer is currently working on packaging the code for submission to the cvs. Maintainer: Dmitry Isakbayev <isakdim(a)gmail.com> This module adds support for the Open Settlement Protocol to SER. The Open Settlement Protocol (OSP) standard is an operations and billing support system protocol for IP network applications such as VoIP, video, short message services (SMS) and content brokering. OSP is an open standard defined by ETSI - the European Telecommunications Standards Institute. OSP has been widely deployed by VoIP carriers to enforce secure access control for peer to peer inter-domain VoIP routing and Call Detail Record (CDR) collection. For more information see www.etsi.org and search for ETSI TS 101 321. lcr module ---------- COMMENT: The LCR module of head has a slightly different feature set that have been difficult to backport to 0.9.x. Maintainer: Juha Heinanen <jh(a)tutpro.com> Backported Least Cost Routing module from HEAD.

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[Serusers] MySQL support faild - Ser can'r connect to Database - Why?
by Dirk Schäfer 29 Oct '05

29 Oct '05

Hello Friends, I have installed Ser with MySQl support and Serweb.Wenn I start the Ser in normal mode - all things are ok and the Ser don't make any failure. Whenn I turned the fork mode on , and start the Ser a second time, it show me a connection error with mysql. Ser can't connect to Database ,so that i can not register any Users with Serweb or so . But serctl work's fine. The database name and password was ser and heslo. I'm using Ser Verison 08-09 but ist still the same failure and i doesn't now, how i can make it better. I'm using the standard ser.cfg file with the normaly changes of IP-Adresse and db_mode. I hope anyone can help me with this problem. I only want to setup a Ser with Mysql Support and Serweb. Thanks and greetings, Dirk

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[Serusers] how to uninstall ser 0.9.3
by julien bos 29 Oct '05

29 Oct '05

Hi all, Can you tell me how to uninstall ser0.9.3? I want to install ser 0.9.4. i try to install ser 0.9.4, after that ser -V reponds version 0.9.3? Thank you so much for your instruction

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[Serusers] ser-rtpproxy bridge mode and T38
by Atif Rasheed 29 Oct '05

29 Oct '05

Hi everyone, I have raised this issue in dev list but got no response, hoping some one here will be able to answer this. I have read in RtpProxy that RtpProxy waits for both parties to send atleast one udp packet and then it fills up both ip:port structures associated with that call and then it starts relaying. Call Setup is like this: --------Media-----> A-Party ----->MG-------->SSW------>SER/RtpProxy------->B-Party now only B-Party is sending UDP Packets and Ser/RtpProxy sends them to SSW (signalling entity) instead of MediaGateway. As soon as RtpProxy receives one udp packet from A-Party then it sends those RTP packets to MG i.e. it starts relaying following the right path. how I can solve this ??? Adding to this, now after RTP relay starts successfully. B-Party (if Dialed number is a FAX) reinvites with T38 and the IP:Port info in SDP remains same. Now somehow RtpProxy is not recognizing the T38 packets becuase we are in the same session and RtpProxy start sending those T38 packets to SSW (signalling entity) instead of sending to MG. Please remember we are in the same session, and RtpProxy shud send these T38 packets to MG, as it was sending RTP packets before ReInvite. is it a bug in RTPProxy, should I use MediaProxy instead ??? Please comment, your help will be highly appriciated. I am using CVS version of SER and RTPProxy. Thank you Atif Rasheed

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[Serusers] pthread create failed with code 11
by Nicky 29 Oct '05

29 Oct '05

Hi Support, I have experienced this error, please advice what can be the caused. Thanks. ERROR: start (AmThread.cpp:85): pthread create failed with code 11 Regards, Nicky

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[Users] package submission
by Daniel-Constantin Mierla 28 Oct '05

28 Oct '05

Hello to everybody that volunteered to build packages, you can start creating the packages for the distributions you subscribed. Please take the tarball with the sources from: http://openser.org/pub/openser/1.0.0/src/ Inside, in the 'packaging' directory you will find the specs for several distributions. It is very likely that you have to change them a bit (see my previous post http://openser.org/pipermail/devel/2005-October/000872.html) and discover some missing items in the specs. For tls packages you have to set TLS=1 in the environment. To submit them, would be good if you have a site from where we can downloaded via http or ftp, otherwise let me know and I will give you further details. If you have some questions, send an email. Cheers, Daniel

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[Users] Insert name into from header of SIP messages
by Brandon Price 28 Oct '05

28 Oct '05

We are storing cnam (callerid name) values in a mysql database for use with openser. I have openser pulling the values from the database using avpops but I need to know how to insert the value into the from so that it will be passed to my sip user agents. Related code posed below modparam ("avpops","db_scheme","cnam_scheme:username_col=phonenumber;table=cnam_c ache;value_col=cnam;value_type=string") modparam("avpops", "avp_aliases", "cnam=s:cnam") avp_db_load("$from/cnam_scheme","cnam/$cnam_scheme"); #sample query -- select cnam from cnam_cache where phonenumber='13143212222' this should now store that value in $cnam. now how do I put that value for example "John Smith" in the from portion of sip messaging so that it displays on the users sip phone, or phone attached to an ata?

2 1

[Serusers] How to search the serusers archives
by Paul Antinori (pantinor) 28 Oct '05

28 Oct '05

I am looking for the search feature on the serusers archives, is this available? Also, is the SER 0.9.4 red hat RPM available for download as I cannot find it on iptel.org or the onsip website. I see the tar binaries itself, but not the RPM. Thank you for help, Paul

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[Serusers] Branch structure
by Ing. Francisco Talavera 28 Oct '05

28 Oct '05

Hi, I have a question. Can I put two URI in the same Invite message?.... My Idea is this I have a module (MY MODULE), and an external program (MY_PROGRAM) those are connected by Unix socket When SER receives an Invite, Mymodule send this to Myprogram, and Myprogram rewrite de uri of the invite and send to Mymodule.. then I want to implement Forwarding. That is A(pedro) --------------------call ---------------------------> B(alice house) If B dont answer, i must forward to C(alice office).. and so on It could arrive doing 2 Uris in the same invite or I have a branch structure in the msg message? Myprogam do Accounting, routing and others Some Ideas.. Thank in advance, Ing. Francisco Talavera Dpto. Investigación y Desarrollo Conexión S.A.

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[Users] TLS changes
by Klaus Darilion 28 Oct '05

28 Oct '05

Hi! I've changed the default value of tls_disable to 1 Thus allows using the standard openser.cfg with TLS enabled openser. klaus

1 0

[Users] debian packaging question
by Klaus Darilion 28 Oct '05

28 Oct '05

Hi! On my box there is installed: libmysqlclient10 3.23.56-3 LGPL-licensed libmysqlclient10-dev 3.23.56-3 LGPL-licensed libmysqlclient12 4.0.24-10 mysql database libmysqlclient14 4.1.11a-4sarge1 mysql database and: /usr/lib/libmysqlclient.so -> libmysqlclient.so.10 Thus, I suggest the mysql module is linked with libmysqlclient10. I further guess, others have to use the same libmysqlclient version? Is there a recommende version of libmysqlclient to use ? regards klaus

2 1

[Serusers] bridging SER peering other Sip Servers
by Joao Pereira 28 Oct '05

28 Oct '05

Hello to all, I have my SER in bridging mode (with two IPs in two non routable networks), with rtpproxy and nathelper: rtpproxy -l 10.0.0.135/193.136.2.205 And the clients in the private and public networks can speak each other with no problems. But in the calls from the private network to external SERs (for example: user(a)iptel.org), the speech only passes one way (from my private IPclient to the user(a)iptel.org) Can someone tell me what could be the problem? These are my routing lines: if (lookup("interno")) { if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAII")) t_on_reply("1"); } if (dst_ip == 193.136.2.205){ if (force_rtp_proxy("FAEI")) t_on_reply("1"); } } else if (lookup("externo")) { if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); } if (dst_ip == 193.136.2.205){ if (force_rtp_proxy("FAEE")) t_on_reply("1"); } } else { sl_send_reply("403", "Utilizador nao Encontrado"); break; }; (....) Thats my external routing section: if( !uri=~"@fccn.pt"){ if (!method=="REGISTER") # internal IPs must use RTP proxy to call user(a)iptel.org if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); route(1); break; } record_route(); t_on_reply("3"); t_relay(); break; } (....) onreply_route[1] { if (!(status=~"183" || status=~"200")) break; force_rtp_proxy("FA"); }

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[Serusers] how can i avoid multiple logins?
by Ryan Pagquil 28 Oct '05

28 Oct '05

Hi, How can I avoid multiple logins using the same username? I want to secure the users so that only the authentic user can only log on? Is there any modules or commands to make this? For example user1 is logged I don't want other people to log also using user1 since user1 is currently logged on. Thanks in advance, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Users] steps to build rpm packages
by Daniel-Constantin Mierla 28 Oct '05

28 Oct '05

Hello, I try to summarize basic steps for generating the RPMs (exemplified for RedHat). The RPM experts can update/correct these guidelines. a) an archive with the sources will be available for download (openser-1.0.0_src.tar.gz) once everything is prepared b) download it in /usr/src/redhat/SOURCES c) copy the spec file that is designed for your distribution in same folder (for RedHats is better to try first openser.spec.fedora) as openser.spec d) edit openser.spec as follows: ** Release: must be set to a string that identifies the distro (e.g., for fedora core 3.0, it should be "Release: fc3.0") ** update the packager name and address to yours ** comment the packages that you cannot generate (e.g., jabber, radius) - there are several sections for each package, which can be easily identified by the name of the package -- you have to comment the whole section e) copy the init file (either openser.init or openser.init.SuSE, from sip-server/packaging/rpm) in /usr/src/redhat/SOURCES f) cd /usr/src/redhat/SOURCES g) rpmbuild -ba openser.spec For SuSE there might be some small differences, but basically is the same process. These guidelines are for those to have root access. If not, then you have to follow first http://dag.wieers.com/howto/bits/rpm-build-user.php (more help via google searching after "building rpm as user"). If everything goes fine, you should get at the end something like (e.g., I built openser base and mysql package): Wrote: /usr/src/redhat/SRPMS/openser-1.0.0-rhel3.0.src.rpm Wrote: /usr/src/redhat/RPMS/i386/openser-1.0.0-rhel3.0.i386.rpm Wrote: /usr/src/redhat/RPMS/i386/openser-mysql-1.0.0-rhel3.0.i386.rpm These are the files we need. If possible, please make them available on a ftp or http site and send me the address to take them. I will notify you back onces I got them, to remove them from your location, if you wish. Otherwise, I will accept attachments to email (at this address), but let me know first that you are going to send them, to be prepared :-) Cheers, Daniel

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[Serusers] How does user agents send log-off request?
by Ryan Pagquil 28 Oct '05

28 Oct '05

Hi, How does User agents sends log-off signal to SER? I know that for the user agent to log in it needs to send a REGISTER request, what request should the user agent need to send SER to be able to log off and be removed from the location table? Could you please explain how the user agent does this... Thanks in advance, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Serusers] question on Adding aliases
by Eric Haskins 28 Oct '05

28 Oct '05

We are trying to get inbound DID's working on our SER box. Outbound is working perfectly as we are doing pre-paid. I ran this command [root@voip1 ~]# serctl alias add 6049094251 sip:5002@216.x.xxx.xx domain unknown: use usernames with domain or set default domain in SIP_DOMAIN [root@voip1 ~]# I get the same result if I put our domain in place of the IP address. We had someone else set this box up for us and I am having problems reaching them. 5002 is is the username of my test ATA logged into the system I can make outgoing calls to the PSTN What am I doing wrong?? I have been reading Archives allday trying to grasp this. The DID provider is directing that DID at the IP of this SER box Eric Haskins Lead Software Developer eric(a)v1010.com

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RE: [Serusers] end the session if no media
by Hakan YASTI 28 Oct '05

28 Oct '05

Yes, I know about it. I can see the active sessions from the session.py. But do not know how to end the seesion, if there is no rtp session and the bye has not come. My session con troller does not check the rtp session because the media is proxied at SER. As I know, may be I am wrong, the ser is managing the media proxy. Ser says to media proxy to start when a call is proxied, and end when the call is finished. Do you suggest a SBC that can do this function..? Thanks, Best Regards, ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Thursday, October 27, 2005 9:20 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media session.py will show if the session is active. Also when a call is put on hold, there will be no rtp stream. I don't know if your SBC handles that. Or is your question something else? g-) ----- Original Message ----- From: Hakan YASTI <mailto:hakan.yasti@teletek.net> To: Greger V. Teigre <mailto:greger@teigre.com> ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Wednesday, October 26, 2005 09:18 AM Subject: RE: [Serusers] end the session if no media Hi Teigre, Thanks for your reply and kind suggestions. In fact, I am using a session controller for sip calls. And it alerts me via email when a sessiıon is reached for about 60 minutes. At that time, I check it manually and kil the session if the session is abondened. By sending BYE messages to both legs. From the PSTN side I get the reply what I assume ( 200 OK ) but from the sip side I get No transaction exist. Because the session was ended before and session controller couldn’t get the BYE message for some reasons. ( Network failure or something else ). What I am looking for that, from the media proxy, I see the active rtp session when I use media proxy for the calls. And I assume that, when the call has finished and couldn’t send or get BYE message, there will be no rtp stream going on. May be I am wrong for this. If rtp session is still open when the a side hunp up and couldn’t send bye , there is nothing to do except killing the session with sending bye messages to both sides, because there is no way to understand the call is active or not. Best Regards, Hakan ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, October 26, 2005 10:00 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media Hakan and Ritesh, It seems to me that you are both asking about the same thing. This topic has been discussed many times on serusers. Search google (site:mail.iptel.org) for "BYE stateful sipsak", I'm pretty sure you will get some info. Also, this FAQ could help: http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 g-) ----- Original Message ----- From: Hakan YASTI <mailto:hyasti@teletek.net> To: serusers(a)lists.iptel.org Sent: Thursday, October 13, 2005 09:13 AM Subject: [Serusers] end the session if no media Hi, I wonder, is there any way, to end the session when the media has ended. Sometimes, we have problem at BYE message. For some reasons, the bye is not coming and may be it is coming wrong. At that time, the media is ended but the call is seen as active. I am using ser and using media proxy. Any suggestion for this issue ? Best Regards, Hakan YASTI NGN Engineer Teletek Telekomünikasyon Hizmetleri A.Þ. Ayazma Dere Cad. Akþit Plaza No:12 Fulya, Beþiktaþ ISTANBUL - TURKEY Tel : +90 212 310 22 34 Fax : +90 212 227 8700 GSM : +90 533 669 06 84 E-mail: hakan.yasti(a)teletek.net <mailto:hakan.yasti@teletek.net> http://www.teletek.net ________________________________ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir. Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir.

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[Serusers] Beer on Wednesday 26th
by Atle Samuelsen 28 Oct '05

28 Oct '05

Hi Guys And Girls, Fellow Serusers, Greger and Me where thinking of trying to join the norwegian troops at Beer Palace on Aker Brygge at 5pm, Upcoming Wednesday. Anybody Intrested in Joining for a Beer or 5? Best Regards Atle

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[Serusers] Removal of email
by voip learner 28 Oct '05

28 Oct '05

Dear sir, Please remove me from the list of the emails. Iam getting hundredsof emails and the mail folder is getting bulk. Kindly do it ASAP. Regards __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Re: Open source billing solution for SER (VoIP)
by Eric Haskins 27 Oct '05

27 Oct '05

>Hello experts, > > I was trying to set up a VoIP system with SER as my signalling >proxy and was able to get lot of useful info from this mailing list. >Thank you. > I would like to know if there exists an open source billing >solution which can fit seemlessly with SER and can handle realtime >rating ( in a prepaid scenario). My initial study on this gave me the >following link with a list of open source billing solutions. I would >also like to know some reivews on these from experts like you. Since I >have very less time to bring my project to life. > >http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems > > >Thanks a lot in advance. > >Regards, >Jithu We have a working system we built that allows for rates to be preset in a db. It calculates start and stop using acc table. I dont know if we will release as gpl but if you want to see the backend functionality contact me directly. It is already setup to work in prepaid eric(a)v1010.com Eric

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[Users] TCP-alias and NAT - one more try to get an answer
by Alexander Ph. Lintenhofer 27 Oct '05

27 Oct '05

Hi All, I just wanted to ask you once again about the TCP-alias riddle. I found out, that there is a problem with the combination of fix_nated_contact(), force_tcp_alias() and NAT: Imagine following situation: Alice behind NAT: socket 172.16.0.6:2421 Nat-Box translates this to 192.168.0.13:6007 The Outbound-Proxy of Alice is 192.168.0.1 Bob is registered with 192.168.1.1 1.) Her INVITE: ==================================================================== INVITE sip:bob@192.168.1.1:2331;transport=tcp;line=wxqurd1s SIP/2.0 [...] Via: SIP/2.0/TCP 172.16.0.6:2421;received=192.168.0.13; branch=z9hG4bK-wm9jcstcboys;rport=6007 From: "Alice" <sip:alice@atlanta.com>;tag=sufzmxi0us To: "Bob" <sip:bob@biloxi.com>;tag=c9550czwtn [...] Contact: <sip:alice@172.16.0.6:2421;transport=tcp;line=fyyuh6tl> [...] ==================================================================== 2.) fix_nated_contact() doesn't work with TCP (look at nathelper.c). force_tcp_alias() now creates following tuple as TCP-alias: 192.168.0.13:6007 to 192.168.0.13:2421 Reason: The TCP-alias is not built solely from the Via-header as suggested in the draft. The portnumber is taken from the Via-header and the IP-address is taken from the source of the incoming datagram. I read it in the sourcecode and assured it by contacting Andrei! 3.) So as a result of the notfixed Contact-header of Alice's INVITE the BYE of Bob is addressed to 172.16.0.6:2421. But no TCP-alias exists for this socket :-( 4.) I made following test by rewriting the Contact-header.... ==================================================================== if (method=="INVITE") { replace("Contact: <sip:alice@172.16.0.6:2421;transport=tcp;", "Contact: <sip:alice@192.168.0.13:2421;transport=tcp;"); } ==================================================================== ...with success. Now TCP-alias works as you can see on my Ethereal-trace below! Compare the destination port of the packet to the destination port of the RURI! ==================================================================== [...] Transmission Control Protocol, Src Port: 5060 (5060), Dst Port: 6007 (6007), ... Session Initiation Protocol Request-Line: BYE sip:alice@192.168.0.13:2421;transport=tcp;line=fyyuh6tl SIP/2.0 Message Header [...] From: "Bob" <sip:bob@biloxi.com>;tag=kcsveifugd To: "Alice" <sip:alice@atlanta.com>;tag=ricaq5cy15 Contact: <sip:bob@192.168.1.1:2331;transport=tcp;line=wxqurd1s> [...] ==================================================================== Another solution: Comment the lines in nathelper.c which force the return in case of TCP or TLS. Now all works well! But why?????????????? regards, Philipp

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[Serusers] LDAP authentication (without Radius)
by Felipe Louback 27 Oct '05

27 Oct '05

I'm working in a VoIP solution here in my university and we have a wide LDAP base and I am thinking about using it for I found on the internet Ldap/Ldaps modules for SER (http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldaps), but I am afraid that I could only use them for lookups and not direct authentication. I guess I could use radius for authentication but I dont know if installing a radius server "just" for this would be worth the effort. Does anyone know if there is a way to authenticate users in a LDAP server without using Radius? I mean, could I use SER directly with LDAP without using RADIUS? Regards, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

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[Users] openser 1.0.0 packages generation
by Daniel-Constantin Mierla 27 Oct '05

27 Oct '05

Hello everybody, I would like to ask all of you that have access to different OS distributions to help with package generation. Please reply (personally if you prefer better this way) with OS name, version and architecture. We are looking mainly for common used distributions: Debian - testing (the other two are accounted already) Redhat Suse Open/NetBSD (FreeBSD accounted) SUN/Solaris Gentoo ... We will help as much as possible with guidelines. We would require tomorrow an early compiling on all available systems to fix eventual warnings. Cheers, Daniel

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[Serusers] SER consultant wanted
by Kevin 27 Oct '05

27 Oct '05

I am looking for a consultant to help with a SER development project. - Kevin

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[Serusers] are voicemail and msilo module the same?
by Meidiana Ten 27 Oct '05

27 Oct '05

Hi there, i'm new here and need help to fix my problem. I have final project that is configuring msilo module in SER 0.9.1. But i'm new in Linux and don't have any idea bout it. i want to ask if: 1. voicemail and msilo module the same? 2. how can i make the very first step? umm..begin from the installation of modules that i need to configure msilo module i'm blank...please someone help me.. thanx.. Yours, Meidiana email add : gotcha_em(a)yahoo.com Indonesia

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[Serusers] IAS and SER
by Andrej Sirk 27 Oct '05

27 Oct '05

Hello, is there somebody who made an working authentication with SER and Microsoft's IAS.

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Re: RE : [Serusers] OpenSER infrastructure dimensioning
by Daniel-Constantin Mierla 27 Oct '05

27 Oct '05

The mailing list has no archive, it is created specially in such way. See: http://openser.org/pipermail/users/2005-September/000941.html Daniel On 10/27/05 17:11, Olivier Taylor wrote: > The Business Archives > > No messages have been posted to this list yet, so the archives are currently > empty. You can get more information about this list. > > Olivier > > -----Message d'origine----- > De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la > part de Daniel-Constantin Mierla > Envoyé : jeudi 27 octobre 2005 16:03 > À : Vanlandschoote Michel > Cc : serusers(a)lists.iptel.org > Objet : Re: [Serusers] OpenSER infrastructure dimensioning > > > Hello, > > On 10/27/05 16:13, Vanlandschoote Michel wrote: > >> Hi, >> >> we are considering OpenSER for offering free SIP-based voice, video >> and IM services and I am trying to get an idea of the project cost. >> >> The components we need are: >> - SIP stateful proxy + registrar server >> - location & presence server >> - simple2jabber >> - media proxy for symmetrical NAT traversal >> - (perhaps I forgot something here?) >> We'll rely on external gateways for PSTN access so that is not to be >> taken in consideration here. >> >> Now can somebody give me an indication on the server park >> dimensioning? In other words, how many SIP servers, media proxies, ... >> would I need per 10000 subscribers? >> > a computer as you described below is more than necessary. The bottle > neck could be the bandwidth if you proxy the media stream through the server > >> >> The server platform could be e.g.: >> - operating system: Linux >> - hardware: HP ProLiant DL380 Generation 3 (single CPU with 1GB RAM) >> but of course any equivalent would do as well. >> >> I also have a second question: which organisations in Europe offer >> commercial support for OpenSER? >> > The openser project is the result of many contributions, its public > activity and development is guided by business independence. For such > requirements and questions, the project provides a mailing list > business(a)openser.org, where the contributors and independent consultants > are ready to take your request: > http://openser.org/cgi-bin/mailman/listinfo/business > > > Daniel > > > >> >> Hope this forum is the right place for this kind of questions... I >> really appreciate your input, >> >> Michel >> >> ************************************************************* >> >> Dit e-mail bericht inclusief eventuele ingesloten bestanden kan >> informatie bevatten die vertrouwelijk is en/of beschermd door >> intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd >> voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit >> bericht (waaronder de volledige of gedeeltelijke reproductie of >> verspreiding onder elke vorm) door andere personen dan de >> geadresseerde(n) is verboden. Indien u dit bericht per vergissing >> heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit >> bericht te verwijderen. >> >> This e-mail and any attachment thereto may contain information which >> is confidential and/or protected by intellectual property rights and >> are intended for the sole use of the addressees. Any use of the >> information contained herein (including but not limited to total or >> partial reproduction or distribution in any form) by other persons >> than the addressees is prohibited. If you have received this e-mail in >> error, please notify the sender and delete its contents. >> >> ************************************************************* >> >> ---------------------------------------------------------------------- >> -- >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Serusers] variables with avp_check
by Joao Pereira 27 Oct '05

27 Oct '05

Hello, I want to use variables in ser.cfg and I m trying to use avpops for it # first by giving the variable "var" the value "internal" # string mysar = internal; avp_write("internal", "s:myvar"); # and checking this variable later: # if (myvar = internal) if (avp_check("s:myvar", "eq/internal/i")){ But this never enters in this if, can someone tell me the way to do it? Or where is some documentation about avpops (iptel.org/ser doesnt have). Thanks Joao Pereira

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[Serusers] OpenSER infrastructure dimensioning
by Vanlandschoote Michel 27 Oct '05

27 Oct '05

Hi, we are considering OpenSER for offering free SIP-based voice, video and IM services and I am trying to get an idea of the project cost. The components we need are: - SIP stateful proxy + registrar server - location & presence server - simple2jabber - media proxy for symmetrical NAT traversal - (perhaps I forgot something here?) We'll rely on external gateways for PSTN access so that is not to be taken in consideration here. Now can somebody give me an indication on the server park dimensioning? In other words, how many SIP servers, media proxies, ... would I need per 10000 subscribers? The server platform could be e.g.: - operating system: Linux - hardware: HP ProLiant DL380 Generation 3 (single CPU with 1GB RAM) but of course any equivalent would do as well. I also have a second question: which organisations in Europe offer commercial support for OpenSER? Hope this forum is the right place for this kind of questions... I really appreciate your input, Michel ************************************************************* Dit e-mail bericht inclusief eventuele ingesloten bestanden kan informatie bevatten die vertrouwelijk is en/of beschermd door intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit bericht (waaronder de volledige of gedeeltelijke reproductie of verspreiding onder elke vorm) door andere personen dan de geadresseerde(n) is verboden. Indien u dit bericht per vergissing heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit bericht te verwijderen. This e-mail and any attachment thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the sole use of the addressees. Any use of the information contained herein (including but not limited to total or partial reproduction or distribution in any form) by other persons than the addressees is prohibited. If you have received this e-mail in error, please notify the sender and delete its contents. *************************************************************

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[Serusers] Curious interop bug.
by David Gilbert 27 Oct '05

27 Oct '05

For those of you who want to follow along, the packet dump in question is at http://strike.dclg.ca/~dgilbert/22-3-filtered ... it's a pcap format file. I'm using a 0.9.4 copy of ser, compiled by hand on debian stable. Only the core portion of ser is being used, and I have simple config file (attached). So, roughly, a call comes in level3, they make it SIP and forward it to my SER on port 5070. SER turns around and forwarded it to asterisk on port 5060 of the same machine and then asterisk forwards it on to a linksys ATA. In short, the problem is that level3 abuses the RFC by sending from a port and expecting you to reply on another port (specified in the Contact: header). This goes almost right except for level3's equally literal read of the Contact header that SER sends. Packet Sip Command Contact ====== =========== ======= level3:38999 ser:5070 INVITE sip:+1416...@<level3 ip>:5060 ser:5070 level3:38999 100 Trying sip:+1866...@<ser ip> (ser invites asterisk, not in dump) asterisk:5060 ata:5060 INVITE sip:+1416...@<asterisk ip> ata:5060 asterisk:5060 100 Trying None ata:5060 asterisk:5060 180 Ringing None (asterisk relays 180 to ser, not in dump) ser:5070 level3:38999 180 Ringing sip:+1866...@<ser ip> ata:5060 asterisk:5060 200 OK sip:604...@<ata ip>:5060 asterisk:5060 ata:5060 ACK sip:+1416...@<asterisk ip> (asterisk relays 200 OK to ser, not in dump) ser:5070 level3:5060 200 OK sip:+1866...@<ser ip> level3:5060 ser:5060 !?! ACK sip:+1866...@<level3 ip>:5060 Now... the problem is that each time ser repsonds, the contact line should be appended with ":5070" and it is not. Therefore, being the literal people they are, level3 repsonds to port 5060 with their ACK. This means that the ACK goes directly to asterisk rather than to SER and it means that when the call ends, the BYE notification from asterisk goes directly to the wrong port at level3. Dave. -- ============================================================================ |David Gilbert, Independent Contractor. | Two things can only be | |Mail: dave(a)daveg.ca | equal if and only if they | |http://daveg.ca | are precisely opposite. | =========================================================GLO================

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re:[Users] SIP clients and DNS SRV
by Zen Kato 27 Oct '05

27 Oct '05

Hi, If we set 'Use DNS SRV: Yes' and 5 minutes REGISTRATION interval on BT101(1.0.7.11) and GXP2000(1.0.1.9),these UACs can successfully Re-REGISTER and INVITE other UACs on time period B, which comes after ser1 stops, every UACs changed their target from first DNS SRV(sip1) to second DNS SRV(sip2). But during time period A if caller is not Re-REGISTERed on sip2, when he dials some number, there is 'no tone(white tone?)' this means the server is down. If the caller's phone is Re-REGISTERed on sip2 and callee's is not yet Re-REGISTERed on sip2 in the time period A, the caller's phone displays "404 not found". So, this is first 5 minutes out of service probabilty time. After the operator of servers room knew that sip1 went down and sip2 is take care of sip calls, he tries to repair sip1 and restarts 'ser(openser)' on sip1. But all INVITEs are handled on sip2 now. BT101s and GXP2000s never automatically go back to sip1. So, the operator must stop sip2 untill all REGISTERs move from sip2 to sip1. This is the second 5 minutes out of service probability time. So, (5+5)minutes out of service probability time exist. If we set 60 minutes(default) for REGISTRATION interval, and all Re-REGISTERs happen on 50 minutes after sip1 down, (50+50)minutes out of service probabilty becomes high. stop ser1 stop ser2 | | V V ....|<-- 5 min-->|<--sip2-->|<-- 5min -->|<--sip1--.....-> |service stop|service |service stop|service | |continue | |continue time|<----A----->|<---B---->|<----C----->|<----D---.....-> Do we need another backup method like Linux-HA if UACs(ONE outgoing line only devices) don't support DNS SRV? Regards, Zen ------------ quote from Daniel's question ----------- Daniel-Constantin Mierla daniel at voice-system.ro Wed Oct 26 16:46:32 CEST 2005 I do not get the question regarding the out of service interval when the client does not support srv. How you computed the 5+5 interval? ----------------------------------------------------

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[Users] rport in via header
by ranveer kunal 27 Oct '05

27 Oct '05

hi all, Is there a way to configure openser so that it sends rport in via header ? Actually th packet from server is reverse natted through another system using any random port, i want it the reply to come on the same port . thanks .... Regards, Ranveer. -- Memories : They bring Diamonds and Rust

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[Users] Detecting runaway calls
by Daryl Sanders 27 Oct '05

27 Oct '05

Other than Session-Timers or B2BUA is there any way to detect a runaway call? I use third party PSTN providers so, I do not control the PSTN gateway. I have read everything I can find on the lists but haven't had much luck. I know I can pull info from the acc table and fake a BYE to disconnect the call, but is there any way to first determine if the call is still actually in progress or if it a runaway. I have heard rumor of some providers that automatically drop calls after an hour or so. This is obviously not a good solution. As always, your help is most appreciated! - Daryl

3 2

[Serusers] Avoiding caller ID spoofing....
by sip 27 Oct '05

27 Oct '05

I think I asked this question before, but I honestly can't remember (been one of those weeks). How can I avoid someone spoofing caller ID by just putting in fake info into their Display Name field in their UA client? I'd like to be able to replace the display name with one of my choosing (preferably based on some rules about the caller's ID... i.e. if the caller is a user on our system, replace his/her display name with the last name, first name from the DB subscriber info. If the caller is from a different system, remove the display name completely and just pass the sip info so that at least a valid identification is seen). This involves the dreaded modification of the From: header (although it shouldn't break RFC because it doesn't actually involve modifying the URI. Is there a reasonable way to do this or does this call for some more module hacking (maybe taking bits from the UAC module and rewriting it to accept an avp or some such) ? N.

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[Serusers] RE: Segmentation fault - Core dump
by deepak.chandrasekaran＠wipro.com 27 Oct '05

27 Oct '05

GDB says like this: Program received signal SIGSEGV, Segmentation fault. 0x00f104aa in mod_init () at authrad_mod.c:123 123 attrs[A_SERVICE_TYPE]->n = "Service-Type"; (gdb) backtrace #0 0x00f104aa in mod_init () at authrad_mod.c:123 #1 0x080772bb in init_mod (m=0x810c2c8) at sr_module.c:474 #2 0x0805e378 in main (argc=2, argv=0xbff7efc4) at main.c:1553 Any ideas? Thanks! Regards, Deepak ________________________________ From: Deepak Chandrasekaran (WT01 - Voice & Next Generation Networks) Sent: Thursday, October 27, 2005 10:24 AM To: 'serusers(a)lists.iptel.org' Cc: 'serdev(a)lists.iptel.org' Subject: Segmentation fault - Core dump Hi, Is there any problem with Fedora Core 4 and ser-0.9.3 with radius authentication? I am using ser-0.9.3, radiusclient-ng 0.5.1 with freeradius-1.0.4 and it is working fine on RedHat 9. I am using the same config with Fedora Core 4. Modules are compiled fine and .so files are successfully generated. But whenever I start SER, it gives me like: 0(0) DEBUG: init_mod: auth_radius 0(0) auth_radius - Initializing Segmentation fault (core dumped) Please let me know if am going wrong somewhere. This is of urgent need and so kindly reply ASAP... Thanks in advance... Regards, Deepak Confidentiality Notice 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 confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

1 0

RE: [Serusers] end the session if no media
by Hakan YASTI 27 Oct '05

27 Oct '05

Hi Teigre, Thanks for your reply and kind suggestions. In fact, I am using a session controller for sip calls. And it alerts me via email when a sessiıon is reached for about 60 minutes. At that time, I check it manually and kil the session if the session is abondened. By sending BYE messages to both legs. From the PSTN side I get the reply what I assume ( 200 OK ) but from the sip side I get No transaction exist. Because the session was ended before and session controller couldn’t get the BYE message for some reasons. ( Network failure or something else ). What I am looking for that, from the media proxy, I see the active rtp session when I use media proxy for the calls. And I assume that, when the call has finished and couldn’t send or get BYE message, there will be no rtp stream going on. May be I am wrong for this. If rtp session is still open when the a side hunp up and couldn’t send bye , there is nothing to do except killing the session with sending bye messages to both sides, because there is no way to understand the call is active or not. Best Regards, Hakan ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, October 26, 2005 10:00 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media Hakan and Ritesh, It seems to me that you are both asking about the same thing. This topic has been discussed many times on serusers. Search google (site:mail.iptel.org) for "BYE stateful sipsak", I'm pretty sure you will get some info. Also, this FAQ could help: http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 g-) ----- Original Message ----- From: Hakan YASTI <mailto:hyasti@teletek.net> To: serusers(a)lists.iptel.org Sent: Thursday, October 13, 2005 09:13 AM Subject: [Serusers] end the session if no media Hi, I wonder, is there any way, to end the session when the media has ended. Sometimes, we have problem at BYE message. For some reasons, the bye is not coming and may be it is coming wrong. At that time, the media is ended but the call is seen as active. I am using ser and using media proxy. Any suggestion for this issue ? Best Regards, Hakan YASTI NGN Engineer Teletek Telekomünikasyon Hizmetleri A.Þ. Ayazma Dere Cad. Akþit Plaza No:12 Fulya, Beþiktaþ ISTANBUL - TURKEY Tel : +90 212 310 22 34 Fax : +90 212 227 8700 GSM : +90 533 669 06 84 E-mail: hakan.yasti(a)teletek.net <mailto:hakan.yasti@teletek.net> http://www.teletek.net ________________________________ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir.

2 1

[Serusers] end the session if no media
by Hakan YASTI 27 Oct '05

27 Oct '05

Hi, I wonder, is there any way, to end the session when the media has ended. Sometimes, we have problem at BYE message. For some reasons, the bye is not coming and may be it is coming wrong. At that time, the media is ended but the call is seen as active. I am using ser and using media proxy. Any suggestion for this issue ? Best Regards, Hakan YASTI NGN Engineer Teletek Telekomünikasyon Hizmetleri A.Þ. Ayazma Dere Cad. Akþit Plaza No:12 Fulya, Beþiktaþ ISTANBUL - TURKEY Tel : +90 212 310 22 34 Fax : +90 212 227 8700 GSM : +90 533 669 06 84 E-mail: hakan.yasti(a)teletek.net <mailto:hakan.yasti@teletek.net> http://www.teletek.net

3 3

[Serusers] Segmentation fault - Core dump
by deepak.chandrasekaran＠wipro.com 27 Oct '05

27 Oct '05

Hi, Is there any problem with Fedora Core 4 and ser-0.9.3 with radius authentication? I am using ser-0.9.3, radiusclient-ng 0.5.1 with freeradius-1.0.4 and it is working fine on RedHat 9. I am using the same config with Fedora Core 4. Modules are compiled fine and .so files are successfully generated. But whenever I start SER, it gives me like: 0(0) DEBUG: init_mod: auth_radius 0(0) auth_radius - Initializing Segmentation fault (core dumped) Please let me know if am going wrong somewhere. This is of urgent need and so kindly reply ASAP... Thanks in advance... Regards, Deepak Confidentiality Notice 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 confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

1 0

[Users] Read through database : openser
by ranveer kunal 27 Oct '05

27 Oct '05

Hello friends, I need a common database as I am trying to loadbalance 2-3 opensers. How can I make sure that the changes committed by one server, is reflected to other. ie. say user1 registers via ser1 and entry is added to the database. now when i quesry server2 for options of user1 i should get 200 ok. i mean is there a mechanism to read thru the database. like cache less mode. Regards, Ranveer.

1 0

[Serusers] REGISTER attempt when db is down
by yserge 26 Oct '05

26 Oct '05

With recently changed behavior of MySQL the reconnection scheme of SER ( and I am running SER 0.9.4 and MySQL 5.0.13 ) would no longer work.. It's OK, that can be fixed with slight correction of MySQL configuration or with minor change in SER code.. What I noticed though is that if the database does go down, the REGISTER request does not get any response, while I would expect SER to still send something like 500 Internal Server Error.. I wonder if it's a SER bug or a routing script issue.. Thanks, Sergei

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Re: [Serusers] AVPOPS syntax help
by Pol Novell Balcells 26 Oct '05

26 Oct '05

Ok! I will try to acomplish it by creating a new table in ser database. But this suxs a lot because serctl script only fills in aliases database, so I will have to do it by hand!! Any trouble will be reported again. p.d: Moltes gràcies, tant sem nota? El 26/10/2005, a las 12:35, Samuel Osorio Calvo escribió: > I haven't played with AVP much so I might be wrong, but I think you > should replace the $ from the username after the alias: > > avp_db_load($alias/$username,i:alias")---> > avp_db_load($alias/username,i:alias") > > ^ > And about the pushto....I have no idea if it accepts two headers, > so somebody with more experience will tell you if it is possible to > push the alias into the ruri and to in the same avp_pushto. Just a > guess, try to do it in two different commands. > > Hope it helps, > > Samuel. > > P.D. Espero q algú et pugui ajudar...sort!! > > Unclassified > > >>>> Pol <kuroki(a)gmail.com> 10/19/05 09:27PM >>> >>>> >>>> > Hi SerUsers! > I'm completly crazy tryin' to do the following: > I just want to load from DB user alias and rewrite it into "$to > $ruri", because these calls shoud go directly to asterisk voicemail > and they are numerical not alphanumeric. > i tried to do the following: > > if(avp_db_load($alias/$username,i:alias") > avp_pushto("$ruri/$to","alias") > just to load from alias table username into an integer called alias > and then pushed to new $ruri. > > I'm new in ser and avps module it's really usefull but a little bit > "unreadable", plz this should work for tomorrow :$ any help plz!? > be as much especific as can, because i don't know if i have done the > proper modparams. > these are: > > modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") > modparam("avpops", "avp_table", "usr_preferences") > #modparam("avpops", "use_domain", "1") > modparam("avpops", "uuid_column", "uuid") > modparam("avpops", "username_column", "username") > modparam("avpops", "domain_column", "domain") > modparam("avpops", "attribute_column", "attribute") > modparam("avpops", "value_column", "value") > modparam("avpops", "type_column", "type") > > THX > > > On 10/13/05, Iqbal <iqbal(a)gigo.co.uk> wrote: > > >> >> >> Hi >> >> I think you have the source syntax written below in the avp_db_load >> statements >> >> so the source will have $from or $to or $ruri optionally followed >> by the >> words "username" or domain >> >> eg >> >> avp_db_load($from/username) of avp_db_load($to/domain) ---note >> these are >> not completed syntax >> >> OR instead of the sip_uri which is $from,$to,$ruri you can have >> avp_alias >> >> which is exactly what it says, its an alias, >> >> Str_value this is the number directly related to the predefined >> alias in >> the line above >> >> The name can be a string S: >> interger i: >> >> or you can leave out the S: and just use the name >> >> or of course an alias >> >> I know thats probably not clear...in fact I just read it myself, >> the best >> thing with avpops , is to look at the db, and create a few, and >> then see >> how to pop them out and play with 'em >> >> Iqbal >> >> >> On 10/12/2005, "Lenir" <lenirsantiago(a)yahoo.com> wrote: >> >> >> >>> Hello list, >>> >>> I'm reading the documentation for AVPOPS, and Im a little >>> confused in this >>> section: >>> >>> -------------------------------------------------------------------- >>> -------- >>> ----- >>> #avp_db_load(source,name) >>> #Loads from DB into memory the AVPs corresponding to the given >>> source. >>> >>> Meaning of the parameters is as follows: >>> >>> >>> source - what info is used for identifying the AVPs. Parameter >>> syntax: >>> >>> >>> source = (sip_uri)['/'('username'|'domain')]) | (avp_alias) | >>> str_value >>> >>> sip_uri = '$from' | '$to' | '$ruri' >>> >>> >>> name - which AVPs will be loaded from DB into memory. Parameter >>> syntax is: >>> >>> >>> name = avp_spec['/'(table_name|'$'db_scheme)] >>> >>> avp_spec = ''|'s:'|'i:'|avp_name|avp_alias >>> >>> >>> Example 1-12. avp_db_load usage >>> >>> .... >>> avp_db_load("$from","i:678"); >>> avp_db_load("$ruri/domain","i:/domain_preferences"); >>> avp_db_load("$uuid","s:404fwd/fwd_table"); >>> avp_db_load("$ruri","i:123/$some_scheme"); >>> .... >>> >>> -------------------------------------------------------------------- >>> -------- >>> ----- >>> >>> My question are: >>> >>> Can someone explain the source syntax? >>> Can someone explain the name syntax? >>> what are i:, s:, avp_name and avp_alias? >>> >>> Thanks! >>> >>> Lenir >>> >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> _____________________________________________________________________ >> Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es >> >> >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _____________________________________________________________________ Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es

1 0

[Serusers] SER Server and Cisco Call Manager 3.3
by gorand＠dvvti.com 26 Oct '05

26 Oct '05

Okay you Cisco Call Manager and SER GURU's. Two questions in this posting the first one is what I call the harder of the two. I work for a company which has Cisco Call manager 3.3 and 2 gateway routers running h323. The system works well when it works. Yet as we all know Cisco Call Manager does not support SIP. Only 4.0 supports sip trunking. Yet my company has sales people who want to use the phones to connect into the company and use their system in a remote user fashion. What I want to do is setup a small SER server on the DMZ configure it for remote users to call internal extensions which are configured by the Cisco Call Manager and I want to my remote users to be able to call out to PSTN via our routers using h323. Is this difficult to do with SER. If possible in your response give me details, and configuration examples. Currently we are using a 5 year old VPN and that is killing our VPN appliance and would rather have it external for our sales force to connect directly from home via a sip device. That was question 1. Second question, I have SER installed on a FEDORA Core 4 and one of the forum users was nice enough to forward me RPM's of Ser. I installed all the RPM along with SERWEB. I am unable to access the url via http://<your-host>/ser/admin Everything installed properly I have looked at my httpd and have done multple /etc/rc.d/init.d/httpd start Everything seems like it starting up. I just can't get the SerWeb working properly. Thanks in Advance Goran Donev

2 1

[Users] SIP clients and DNS SRV
by Helge Waastad 26 Oct '05

26 Oct '05

Hi, this is not a technical question regarding OpenSER, but more of req for information. I'm using Cisco phones and gateways and have implemented DNS SRV redundancy. However, I have never seen a SIP phone that really works with DNS SRV redundancy, except from Cisco. (which actually works as a charm) Does anyone know other phones that can use dns srv for redundancy? At least Cisco tries the next DNS SRV entry if it receives a port unreachable from the highest priority proxy... -- mvh/best regards Helge Waastad System Engineer Smartnet

5 5

[Serusers] Is there a way to do read through database in ser ?
by ranveer kunal 26 Oct '05

26 Oct '05

Hi, I am using 2 ser servers and they both share a common database, so they write the location information to that database. but problem is they don'read the information from database, is there a way to force them to lookup the database, so that changes made by one are reflected to other or atleast to refresh cache from database. Regards, Ranveer.

2 2

[Serusers] help on linking ser to mysql
by Dhiman D Chowdhury 26 Oct '05

26 Oct '05

Hi, I will appreciate your kind help in the following issue: 1. I could not find ser-mysql package for download. I need the package for REDHAT linux 9.0. -- Can anybody please help? 2. I have Mysql server that automatically installed with REDHAT linux but when I tried: mysql> connect ser -- it fails with a message, "error 1049 unknown database ser". Please help. Thanks Pappu

1 0

[Serusers] AVPOPS syntax help
by Lenir 26 Oct '05

26 Oct '05

Hello list, I'm reading the documentation for AVPOPS, and Im a little confused in this section: ---------------------------------------------------------------------------- ----- #avp_db_load(source,name) #Loads from DB into memory the AVPs corresponding to the given source. Meaning of the parameters is as follows: source - what info is used for identifying the AVPs. Parameter syntax: source = (sip_uri)['/'('username'|'domain')]) | (avp_alias) | str_value sip_uri = '$from' | '$to' | '$ruri' name - which AVPs will be loaded from DB into memory. Parameter syntax is: name = avp_spec['/'(table_name|'$'db_scheme)] avp_spec = ''|'s:'|'i:'|avp_name|avp_alias Example 1-12. avp_db_load usage ... avp_db_load("$from","i:678"); avp_db_load("$ruri/domain","i:/domain_preferences"); avp_db_load("$uuid","s:404fwd/fwd_table"); avp_db_load("$ruri","i:123/$some_scheme"); ... ---------------------------------------------------------------------------- ----- My question are: Can someone explain the source syntax? Can someone explain the name syntax? what are i:, s:, avp_name and avp_alias? Thanks! Lenir

4 3

[Serusers] mailing list
by Iqbal 26 Oct '05

26 Oct '05

Hi has there been extremly low traffic on the list over the last few days, or is it my ISP Iqbal

2 1

[Users] openSER on interix
by Public Dump 26 Oct '05

26 Oct '05

Was anynbody able to compile/run openSER on the Windows Interix Environment (part of Microsoft Services for UNIX) ?

2 1

[Serusers] Automatic purge of users
by Damien Sandras 26 Oct '05

26 Oct '05

Hello, I would like to know if SER stores somewhere the last "registration" date for a given user. I have looked at all tables and couldn't find such a field. So I would like to know if there is a recommended way to automatically delete all users who haven't logged in since more than x months? Thank you, -- _ Damien Sandras (o- GnomeMeeting: http://www.gnomemeeting.org/ //\ FOSDEM 2006 : http://www.fosdem.org v_/_ SIP Phone : sip:dsandras@gnomemeeting.net sip:600000@gnomemeeting.net

2 1

FW: [Users] SIP clients and DNS SRV
by Helge Waastad 26 Oct '05

26 Oct '05

Forgot CC on this one..... -----Original Message----- From: Helge Waastad Sent: 26. oktober 2005 13:56 To: 'Zen Kato' Subject: RE: [Users] SIP clients and DNS SRV Hi, You know, even if DNS SRV works fine, you will have problems with NAT'ed clients behind firewalls. Ther reason is that the session is opened between the server handeling the registration and the UAC. If this on crashes, the second one will never be able to send INVITE to the UAC since there is no open session through the firewall. Yes, you will see service unavaliability up to the registration interval, depending on the UAC functionality. The only solution for this is a (outbound) proxy cluster using Linux-HA (or vrrp) and a IP address take-over. (Or a Border Controller, but these are not for free..) Cisco 7960 works fine (also snom) I've seen that Sipura in 3.1.5 release also supports dns srv for redundancy/failover, but I have'nt tried it. -----Original Message----- From: Zen Kato [mailto:zenkato@pis.bekkoame.ne.jp] Sent: 26. oktober 2005 13:45 To: Helge Waastad Cc: klaus.mailinglists(a)pernau.at; users(a)openser.org Subject: Re: [Users] SIP clients and DNS SRV Hi, I posted on [serusers] regarding DNS SRV,t_replicate,backup server and GS phones. My test resut of BT101(1.0.7.11-newest) and GXP2000(1.0.1.9) was no good. My question is eventhough we set up DNS SRV on server side, if sip clients does not suppot DNS SRV, do we have a risk of (5+5)minutes out of service time when we set up 5 minutes REGISTER intervals? If UA have multiple REGISTERs outgoing lines such as GXP2000, we could set up Line1 for sip1&sip2(DNS SRV) Line2 for sip3 server. In this case if line1 does not work, the user can use Line2. Xlite can register 3 sip servers, but only default sip proxy can use as outgoing line, so the Xlite users have (5+5)minutes out of service risk. Does Cisco 7960 work perfectly on DNS SRV? Regards, Zen Helge Waastad wrote : > Hi, > I've started to add some more information to voip-info. > I'm still waiting for response from Grandstream and others on their > support (or future support) for DNS SRV. > > br hw > > On Wed, 2005-10-19 at 17:32 +0200, Klaus Darilion wrote: > > Hi Helge! > > > > According to my tests, also SNOM can do SRV based failover. > > > > I've published my test results at: > > http://www.voip-info.org/wiki/view/SRV+implementations > > > > Please add your test results to this. > > > > regards > > klaus > > > > Helge Waastad wrote: > > > Hi, > > > this is not a technical question regarding OpenSER, but more of > > > req for information. > > > > > > I'm using Cisco phones and gateways and have implemented DNS SRV > > > redundancy. > > > However, I have never seen a SIP phone that really works with DNS > > > SRV redundancy, except from Cisco. (which actually works as a > > > charm) > > > > > > Does anyone know other phones that can use dns srv for redundancy? > > > > > > At least Cisco tries the next DNS SRV entry if it receives a port > > > unreachable from the highest priority proxy... > > > > > > > > > -- > mvh/best regards > Helge Waastad > System Engineer > Smartnet > (+47)67830017 > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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Re: [Serusers] AVPOPS syntax help
by Samuel Osorio Calvo 26 Oct '05

26 Oct '05

I haven't played with AVP much so I might be wrong, but I think you should replace the $ from the username after the alias: avp_db_load($alias/$username,i:alias")---> avp_db_load($alias/username,i:alias") ^ And about the pushto....I have no idea if it accepts two headers, so somebody with more experience will tell you if it is possible to push the alias into the ruri and to in the same avp_pushto. Just a guess, try to do it in two different commands. Hope it helps, Samuel. P.D. Espero q algú et pugui ajudar...sort!! Unclassified >>> Pol <kuroki(a)gmail.com> 10/19/05 09:27PM >>> Hi SerUsers! I'm completly crazy tryin' to do the following: I just want to load from DB user alias and rewrite it into "$to $ruri", because these calls shoud go directly to asterisk voicemail and they are numerical not alphanumeric. i tried to do the following: if(avp_db_load($alias/$username,i:alias") avp_pushto("$ruri/$to","alias") just to load from alias table username into an integer called alias and then pushed to new $ruri. I'm new in ser and avps module it's really usefull but a little bit "unreadable", plz this should work for tomorrow :$ any help plz!? be as much especific as can, because i don't know if i have done the proper modparams. these are: modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") #modparam("avpops", "use_domain", "1") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") THX On 10/13/05, Iqbal <iqbal(a)gigo.co.uk> wrote: > > > Hi > > I think you have the source syntax written below in the avp_db_load > statements > > so the source will have $from or $to or $ruri optionally followed by the > words "username" or domain > > eg > > avp_db_load($from/username) of avp_db_load($to/domain) ---note these are > not completed syntax > > OR instead of the sip_uri which is $from,$to,$ruri you can have avp_alias > > which is exactly what it says, its an alias, > > Str_value this is the number directly related to the predefined alias in > the line above > > The name can be a string S: > interger i: > > or you can leave out the S: and just use the name > > or of course an alias > > I know thats probably not clear...in fact I just read it myself, the best > thing with avpops , is to look at the db, and create a few, and then see > how to pop them out and play with 'em > > Iqbal > > > On 10/12/2005, "Lenir" <lenirsantiago(a)yahoo.com> wrote: > > >Hello list, > > > >I'm reading the documentation for AVPOPS, and Im a little confused in this > >section: > > > >---------------------------------------------------------------------------- > >----- > >#avp_db_load(source,name) > >#Loads from DB into memory the AVPs corresponding to the given source. > > > >Meaning of the parameters is as follows: > > > > > >source - what info is used for identifying the AVPs. Parameter syntax: > > > > > >source = (sip_uri)['/'('username'|'domain')]) | (avp_alias) | str_value > > > >sip_uri = '$from' | '$to' | '$ruri' > > > > > >name - which AVPs will be loaded from DB into memory. Parameter syntax is: > > > > > >name = avp_spec['/'(table_name|'$'db_scheme)] > > > >avp_spec = ''|'s:'|'i:'|avp_name|avp_alias > > > > > >Example 1-12. avp_db_load usage > > > >.... > >avp_db_load("$from","i:678"); > >avp_db_load("$ruri/domain","i:/domain_preferences"); > >avp_db_load("$uuid","s:404fwd/fwd_table"); > >avp_db_load("$ruri","i:123/$some_scheme"); > >.... > > > >---------------------------------------------------------------------------- > >----- > > > >My question are: > > > >Can someone explain the source syntax? > >Can someone explain the name syntax? > >what are i:, s:, avp_name and avp_alias? > > > >Thanks! > > > >Lenir > > > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _____________________________________________________________________ > Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] EmbeddedObj.o sipstack make error - Cannot compile b2bua
by administrator 26 Oct '05

26 Oct '05

Hello everybody, I am trying to install b2bus from cvs and I get the following error: make[1]: Leaving directory `/root/vocal/sip/sipstack' make[1]: Entering directory `/root/vocal/sip/sipstack' g++ -Wall -fPIC -Wno-deprecated -D_REENTRANT -DUSE_PTHREADS -g -I../../build -I../../build/../sdp/sdp2 -I../../build/../util -I../../build/../util/threads -I../../build/../util/logging -I../../build/../util/crypto -I../../build/../util/statistics -I../../build/../util/snmp -I../../build/../util/signals -I../../build/../util/behavior -I../../build/../util/io -I../../build/../util/services -I../../build/../util/transport -I../../build/../util/config -I../../build/../util/dnssrv -I../../build/../util/deprecated -I../../build/../util/adt -I../../build/../contrib/libxml2.Linux.i686 -I../../build/../contrib/libxml2.Linux.i686/include/libxml -I../../build/../contrib/libxml2.Linux.i686/include -DVOCAL_USE_DEPRECATED -DVOCAL_USING_PENTIUM -DOLD_PROV -c -o obj.debug.Linux.i686/EmbeddedObj.o EmbeddedObj.cxx BaseUrl.hxx:146: error: specialization of βtemplate<class _Key> struct __gnu_cxx::hashβ in different namespace /usr/lib/gcc/i386-redhat-linux/4.0.1/../../../../include/c++/4.0.1/ext/hash_ fun.h:71: error: from definition of βtemplate<class _Key> struct __gnu_cxx::hashβ EmbeddedObj.cxx: In member function βData Vocal::EmbeddedObj::doReverseEscape(const std::string&)β: EmbeddedObj.cxx:213: warning: comparison between signed and unsigned integer expressions make[1]: *** [obj.debug.Linux.i686/EmbeddedObj.o] Error 1 make[1]: Leaving directory `/root/vocal/sip/sipstack' make: *** [sip] Error 2 I user SER(0.9.4) and freeRadius (latest) on Fedora Core 4. Please Help ASAP. Thank you in advance Yiannis Marios

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[Serusers] EmbeddedObj Sipstack make error
by Yiannis Martis 26 Oct '05

26 Oct '05

Hello everybody, I am trying to install b2bus from cvs and I get the following error: make[1]: Leaving directory `/root/vocal/sip/sipstack' make[1]: Entering directory `/root/vocal/sip/sipstack' g++ -Wall -fPIC -Wno-deprecated -D_REENTRANT -DUSE_PTHREADS -g -I../../build -I../../build/../sdp/sdp2 -I../../build/../util -I../../build/../util/threads -I../../build/../util/logging -I../../build/../util/crypto -I../../build/../util/statistics -I../../build/../util/snmp -I../../build/../util/signals -I../../build/../util/behavior -I../../build/../util/io -I../../build/../util/services -I../../build/../util/transport -I../../build/../util/config -I../../build/../util/dnssrv -I../../build/../util/deprecated -I../../build/../util/adt -I../../build/../contrib/libxml2.Linux.i686 -I../../build/../contrib/libxml2.Linux.i686/include/libxml -I../../build/../contrib/libxml2.Linux.i686/include -DVOCAL_USE_DEPRECATED -DVOCAL_USING_PENTIUM -DOLD_PROV -c -o obj.debug.Linux.i686/EmbeddedObj.o EmbeddedObj.cxx BaseUrl.hxx:146: error: specialization of βtemplate<class _Key> struct __gnu_cxx::hashβ in different namespace /usr/lib/gcc/i386-redhat-linux/4.0.1/../../../../include/c++/4.0.1/ext/hash_fun.h:71: error: from definition of βtemplate<class _Key> struct __gnu_cxx::hashβ EmbeddedObj.cxx: In member function βData Vocal::EmbeddedObj::doReverseEscape(const std::string&)β: EmbeddedObj.cxx:213: warning: comparison between signed and unsigned integer expressions make[1]: *** [obj.debug.Linux.i686/EmbeddedObj.o] Error 1 make[1]: Leaving directory `/root/vocal/sip/sipstack' make: *** [sip] Error 2 I user SER(0.9.4) and freeRadius (latest) on Fedora Core 4. Please Help ASAP. Thank you in advance Yiannis Marios _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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[Serusers] Error while compiling kphone package version 4.2
by rajesh.rajguru＠wipro.com 26 Oct '05

26 Oct '05

Hi, We are trying to make package of kphone from http://www.wirlab.net/kphone/ <http://www.wirlab.net/kphone/> We have downloaded kphone package version "KPhone 4.2" from the above mentioned site After referring to INSTALL instructions, we have executed following steps 1) go to kphone dir and do ./configure. This was executed properly 2) do make . When we are doing make we got following error ---------- make[1]: Entering directory `/home/wipro/SER_Test/kphone/kphone' c++ -I/usr/lib/qt3-gcc2.96/include -Wall -O3 -I. -I../gsm -I../ilbc -I../dissipa te2 -DHAVE_CONFIG_H -DSHARE_DIR=\"/usr/local/share/apps/kphone\" -DPO_DIR=\"/usr /local/share/kphone/translations//\" -c -o kphone.o kphone.cpp kphone.cpp: In member function `void KPhone::createTrayIcon()': kphone.cpp:272: no method `QPixmap::fromMimeSource' kphone.cpp:274: no method `QPixmap::fromMimeSource' kphone.cpp: In member function `void KPhone::updateTrayIcon()': kphone.cpp:297: no method `QPixmap::fromMimeSource' kphone.cpp:299: no method `QPixmap::fromMimeSource' make[1]: *** [kphone.o] Error 1 make[1]: Leaving directory `/home/wipro/SER_Test/kphone/kphone' make: *** [all] Error 2 ----------- In fact we also got this error for older versions of kphone package like KPhone 4.1.1, KPhone 4.1.0 We tried commenting call to fromMimeSource in file kphone.cpp, and then doing make, but it gives some unresolved symbols at the end Please let us know if you have also faced same types of errors and if yes then any pointers about this will be of great help Thanks in advance for your help Regards, Rajesh Confidentiality Notice 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 confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

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[Users] Beer on Wednesday 26th
by Helge Waastad 26 Oct '05

26 Oct '05

Hi, If I can make it, I'll stop by. The question is, what do SER/OpenSER people look like ;-) BTW, There will be a seminar, first of all regarding Asterisk but also discussing SER, on the 1th of November I guess it would be possible to have some discussions after this aswell. http://www.asterisk.no/asterisk/endelig_norges_foerste_asterisk_seminar_ meld_deg_paa_naa hw

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[Serusers] FW: Call logs
by Colin Jordan 26 Oct '05

26 Oct '05

I can not find where the call logs are in the ser database. What can I do to enable call information logging (time stamps, destination, exc) ? DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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[Serusers] fee-for-service SER server
by Michael Kendall 26 Oct '05

26 Oct '05

Dear Wizards, Many of you seem to be running public SER servers with hundreds of users. I would like to become a paying customer of such a SER server, but cannot find any such services advertised on the internet. Can someone please recommend a running, fee-for-service SER server? A few weeks ago, I asked this same question, and got zero responses. And I mean no responses of any kind. Is this the wrong forum to ask such a question? Even messages pointing out why I am a total idiot would have been preferable to total silence. If there are reasons that it is not practical for anyone to operate a fee-for-service SER server, I would love to see some discussion of what makes it so impractical. Please feel free to speculate on why you think such services have not been created. thanks, Michael

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[Serusers] latest rtpproxy + ser-0.8.14 + nathelper!!
by s0914508 26 Oct '05

26 Oct '05

Hi! I have some problems with rtp proxy. And the syslog tells that my rtpproxy can work (enable!!) ---------------------------------------- UA1: 10.10.54.131 (caller), behind NAT UA2: 218.171.144.86 (callee), public Internet SIP UA: Windows Messenger 5.1 SIP Server: ser-v0.8.14 Load module: nathelper RTP Proxy: latest rtpproxy (not portaone) NAT: CiscoSymmetric NAT ----------------------------------------- [Problem] Callee can hear caller's voice, but caller can't hear callee's. The the packet analysis is like this http://sip.csie.ncyu.edu.tw/~coolboy/calleereceiveinvite.JPG the line ( c= IN IP4 140.130.201.15 ) is NAT Server's IP address, not rtp proxy's IP address(140.130.175.2). I use force_rtp_proxy("140.130.175.2" ); to change c=IP4 with IP address(140.130.175.2), but it seems doesn't work. ----------------------------------------- Does anyone know how to solve my problem and the reason? It's very important to me. Thanks a lot !! ----------------------------------------- [Question] 1. Does c=IP4 get from rtpproxy if I use rtpproxy.?? And how they operate?? 2. I want to compile the latest nathelper module with FreeBSD system. I've read the 'Makefile', but it says "don't run with this Makefile, you should use 'master.Makefile' ". Does anyone can tell me how to do?? --== Mail Via Ncyu Mail ==--

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[Serusers] How to modify the URI.
by gc 26 Oct '05

26 Oct '05

I am new to SER. I'd like to modify some of the sip headers like 'uri' and 'from' header to remove +1 prefix from the phone number. What I want to do is to change +15042339988(a)3.2.4.3 to 5042339988(a)3.2.4.3 Can somebody show me how to do it in ser.cfg? gc

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[Serusers] Ser 0.94 Segfaults when using auth_db module
by Hilco de Lathouder 26 Oct '05

26 Oct '05

2 1

RE: [Serusers] auto configuration
by Ricardo Martinez 26 Oct '05

26 Oct '05

Hi Nhadie. I'm using the autoprovisioning system with PAP2 and Sipura. I don't know if the other phones have this feature. I also know that SJPhone can have this feaure too. Regards, Ricardo Martinez.- > -----Mensaje original----- > De: Nhadie [mailto:nhadie@cbcpworld.com] > Enviado el: Jueves, 08 de Septiembre de 2005 10:43 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] auto configuration > > > Hi List! > > Would just like to ask how can I automatically download a > configuration for > a SIP phone. I mean , let's say I have a client that bought > the phone, once > he plugged it in on the internet it will download a > configuration file, that > has it's username, password, sip proxy etc. I use > Grandstream, PAP2, Sipura, > x-lite x-pro and eyebeam. Is that possible? > > Thank You > > Regards, > Nhadie > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] SER cannot connect to MySQL Server
by romulus 26 Oct '05

26 Oct '05

HI, I'm using Debian Linux(while I have no idea how to check its version), and running SER on my PC. I installed MySQL 4.1.14 and SER 0.9.4, chmoded the /tmp directory to 777, and the MySQL server runs well. But when I start SER, it does not run and leaves some error message in syslog like this: -------------------------------------------------------------------------------------------- Oct 10 14:31:03 Serverhost /usr/sbin/ser[6525]: Maxfwd module- initializing Oct 10 14:31:03 Serverhost /usr/sbin/ser[6525]: new_connection(): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Oct 10 14:31:03 Serverhost /usr/sbin/ser[6525]: db_init(): Could not create a connection Oct 10 14:31:03 Serverhost /usr/sbin/ser[6525]: ERROR: auth_db_ver: unable to open database connection Oct 10 14:31:03 Serverhost /usr/sbin/ser[6525]: auth_db:str_fixup(): Error while querying table version Oct 10 14:31:03 Serverhost /usr/sbin/ser[6525]: ERROR: fix_expr : fix_actions error -------------------------------------------------------------------------------------------- The file /var/run/mysqld/mysqld.sock doest not exist, but the MySQL still runs well. I also don't know if it is correct. Does anyone know how to resolve this? Thank you! ser.cfg, "gulab.twbbs.org" is my domain. -------------------------------------------------------------------------------------------- # # $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=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (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 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/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=~"gulab.twbbs.org") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("gulab.twbbs.org", "subscriber")) { www_challenge("gulab.twbbs.org", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; 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(); }; } -------------------------------------------------------------------------------------------- my.cnf -------------------------------------------------------------------------------------------- # # The MySQL database server configuration file. # # You can copy this to one of: # - "/etc/mysql/my.cnf" to set global options, # - "/var/lib/mysql/my.cnf" to set server-specific options or # - "~/.my.cnf" to set user-specific options. # # One can use all long options that the program supports. # Run program with --help to get a list of available options and with # --print-defaults to see which it would actually understand and use. # # For explanations see # http://dev.mysql.com/doc/mysql/en/server-system-variables.html # This will be passed to all mysql clients # It has been reported that passwords should be enclosed with ticks/quotes # escpecially if they contain "#" chars... # Remember to edit /etc/mysql/debian.cnf when changing the socket location. [client] port = 3306 socket = /var/run/mysqld/mysqld.sock # Here is entries for some specific programs # The following values assume you have at least 32M ram # This was formally known as [safe_mysqld]. Both versions are currently parsed. [mysqld_safe] socket = /var/run/mysqld/mysqld.sock nice = 0 [mysqld] # # * Basic Settings # user = mysql pid-file = /var/run/mysqld/mysqld.pid socket = /var/run/mysqld/mysqld.sock port = 3306 basedir = /usr datadir = /var/lib/mysql tmpdir = /tmp language = /usr/share/mysql/english skip-external-locking # # For compatibility to other Debian packages that still use # libmysqlclient10 and libmysqlclient12. old_passwords = 1 # # Instead of skip-networking the default is now to listen only on # localhost which is more compatible and is not less secure. bind-address = 127.0.0.1 # # * Fine Tuning # key_buffer = 16M max_allowed_packet = 16M thread_stack = 128K # # * Query Cache Configuration # query_cache_limit = 1048576 query_cache_size = 16777216 query_cache_type = 1 # # * Logging and Replication # # Both location gets rotated by the cronjob. # Be aware that this log type is a performance killer. #log = /var/log/mysql.log #log = /var/log/mysql/mysql.log # # Error logging goes to syslog. This is a Debian improvement :) # # Here you can see queries with especially long duration #log-slow-queries = /var/log/mysql/mysql-slow.log # # The following can be used as easy to replay backup logs or for replication. #server-id = 1 log-bin = /var/log/mysql/mysql-bin.log # See /etc/mysql/debian-log-rotate.conf for the number of files kept. max_binlog_size = 104857600 #binlog-do-db = include_database_name #binlog-ignore-db = include_database_name # # * BerkeleyDB # # According to an MySQL employee the use of BerkeleyDB is now discouraged # and support for it will probably cease in the next versions. skip-bdb # # * InnoDB # # InnoDB is enabled by default with a 10MB datafile in /var/lib/mysql/. # Read the manual for more InnoDB related options. There are many! # # * Security Features # # Read the manual, too, if you want chroot! # chroot = /var/lib/mysql/ # # If you want to enable SSL support (recommended) read the manual or my # HOWTO in /usr/share/doc/mysql-server/SSL-MINI-HOWTO.txt.gz # ssl-ca=/etc/mysql/cacert.pem # ssl-cert=/etc/mysql/server-cert.pem # ssl-key=/etc/mysql/server-key.pem [mysqldump] quick quote-names max_allowed_packet = 16M [mysql] #no-auto-rehash # faster start of mysql but no tab completition [isamchk] key_buffer = 16M # # * NDB Cluster # # See /usr/share/doc/mysql-server-*/README.Debian for more information. # # The following configuration is read by the ndbd storage daemons, # not from the ndb_mgmd management daemon. # # [MYSQL_CLUSTER] # ndb-connectstring=127.0.0.1 --------------------------------------------------------------------------------------------

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[Serusers] DNS SRV,t_replicate,backup server and GS phones
by Zen Kato 26 Oct '05

26 Oct '05

Hi, I have been testing 'DNS SRV', backup server and t_replicate with two 'ser'servers, BT101(firmware:1.0.7.11), and two GXP2000s(1.0.1.9). The t_replicate works fine between sip1(main) and sip2(backup) servers. Then, I stopped 'ser' on sip1 server, three UAs sent REGISTERs not to sip1 but to sip2. I tested make calls between three UAs, sip2 took care of sip calls, worked fine. After that, I restarted 'ser' on sip1. But the three UAs still send REGISTER to sip2, eventhough sip1 came back and became alive. I set up 5 minutes intervals for these theree UAs for REGISTER. So, I stopped 'ser' on sip2 for 10-15 minutes, three UAs sent REGISTERs to sip1 again and tryed to 't_replicate' to sip2, these were failed as 'destination unreachable'. Then I restarted 'ser' on sip2 again. After REGISTERed on sip1, t_replicate send REGISTERs from sip1 to sip2. Now, everything works fine. Is this server side opration normal? Or these UAs are still bad boys? Regards, Zen

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[Users] Release 1.0.0 scheduled for Friday, October 28
by Daniel-Constantin Mierla 25 Oct '05

25 Oct '05

Hello everybody, the new stable release is scheduled for next Friday, October 28. It will be versioned *1.0.0* instead of 0.10.0 since mandatory requirements of RFC are in place (the last major one, TLS is now in the trunk) and a lot of new features have been introduced. The release was postponed several time to allow more time for testing. There is no known bug at this moment that is not going to be fixed before release (the last one, reported by Klaus causing memory leak for CANCEL transactions in failure routes has been localized and expect to have it fixed tomorrow on CVS). Until Friday, remaining docs should be updated and relevant documents (changelog, new features, deprecated stuff) have to be written. We have to ask people willing to generate packages for different architecture to give some help since we do not have access to all OS distributions. We can provide guidelines for those that need them. If you are aware of some things that are missing for this release, please report them to devel(a)openser.org Cheers, Daniel

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[Users] OpenSER and Asterisk
by Barry Flanagan 25 Oct '05

25 Oct '05

Hi, I want to use openser as proxy/registrar with Asterisk. Users will exist both in openser's subscriber database and in asterisk's sip friends. I have this kind of working, but have a few questions. Currently openser authenticates users fine, and will route to * for pstn and asterisk will accept the call if I have openser set up in * sip.conf as a proxy. I would rather, however, that the user gets authenticated as themselves on asterisk, rather than the call happening as the proxy user. without openser set up in asterisk, I get continual 407 Proxy Authentication required. Additionally, it appears that the call is still routing through openser, rather than being directly between the client and asterisk. Questions: 1. How can I set it up so that when a call gets sent to * that authentication takes place directly between asterisk and the client 2. How should I be sending the call from openser so that the rest of the call is directly between asterisk and the client? Any working configs would be welcome - these seem to be hard to come by. Thanks. -- -Barry Flanagan

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[Users] Error
by Ronald Wiplinger 25 Oct '05

25 Oct '05

I have in my name server the lines: trace-intl.net: _sip._udp.trace-intl.net. IN SRV 20 0 5062 dns.trace-int.net. trace-intl.net-internal: _sip._udp IN SRV 20 0 5062 voip but on the same machine another domain runs with the same lines for port 5060 My intention was to use trace-intl.net for Openser on 5062, while Asterisk rmains on 5060. It seems, that this lines convert all tries to connect at 5062 to connect to 5060. Could that be? I have now disabled the lines ( _sip._udp ....) for Asterisk, but still receive the error: Oct 25 08:17:36 viop2 /usr/local/sbin/openser[9142]: Maxfwd module- initializing Oct 25 08:17:36 viop2 /usr/local/sbin/openser[9142]: ERROR: udp_init: bind(5, 0x622a04, 16) on 127.0.0.1: Address already in use How can I solve that? bye Ronald Wiplinger

2 1

[Serusers] SER and multiple gateways
by Stefan Prelle 25 Oct '05

25 Oct '05

Hi, I like to use SER for distribute calls on multiple PSTN gateways. In a first step I just have two gateways and need SER to do some load balancing. AFAIK previous SER versions came with a load balancing module. What are the current options to do this (beside some kind of suffix routing)? Regards, Stefan

2 1

[Serusers] IAS (Microsoft Internet Authentication Service)
by Andrej Sirk 25 Oct '05

25 Oct '05

Hello, can somebody suggest me how to configure those two IAS and SER. I need to authenticate from SER to IAS. I am using radiusclient-ng.

1 0

[Users] Children=
by Glenn Dalgliesh 25 Oct '05

25 Oct '05

Wondering what the school of thought it about the number the children that should be specified given a statefull configuration with approx 300 users with expiration of 60 sec and nat_ping of 30sec and modparam("usrloc|domain", "db_mode", 1) with postgres and any other related tweaks. Thanks

2 1

[Serusers] HELP:could NOT account BYE/CANCEL messages with RADIUS
by shikun.wang 25 Oct '05

25 Oct '05

Hi,all I've tried to use RADIUS(freeradius-0.9.1) to account with SER(ser-0.9.3).But I can't account BYE or CANCEL messages,while I can account INVITE and ACK. In ser.cfg I write as follows: ... modparam("acc", "log_level", 4) modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) ... if (method=="INVITE") { log(1, "INVITE\n"); if (!radius_proxy_authorize("")) { proxy_challenge("", "0"); setflag(3); break; }; setflag(2); }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n");\ setflag(2); }; ... What's wrong with it?Is there anything I miss to configure? Please give me any hint and thanks a lot.

1 0

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