- sr-users - kamailio.org

Re: [Serusers] One way audio
by Greger V. Teigre 10 Jan '07

10 Jan '07

You need to look at when NAT is detected (nat_uac_test) for INVITE and for 200 OK in onreply route for the same call. Make sure NAT is detected for both, if not, find out why it is not detected. g-) Gareth Owen wrote: > > Thanks Greger, I've already worked through the getting started doc and > think I've got a good understanding but am still getting the one-way > audio. Can you suggest anything to help me understand how I can track > the problem? Thanks gareth. > > > > ------------------------------------------------------------------------ > > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* 04 January 2007 06:5 > *To:* gareth.owen(a)bevoice.net > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] One way audio > > > > Try gettingstarted doc. http://iptel.org/ser/doc/gettingstarted > g-) > > Gareth Owen wrote: > > Can someone help? > > > > I am fairly new to SER and Mediaproxy and can't find a detailed > explanation of how nathelper etc work, so I'm struggling! > > > > My config is:- > > > > SER 0.9.6 > > Mediaproxy 1.7.2 > > Fedora Core 4 86x64 > > > > SER and Mediaproxy run on the same server which acts as a gateway to a > remote Sonus PSTN gateway over a VPN. > > > > I also use Asterisk on my local network. The SER server acts as a > router between Asterisk servers and the PSTN via the Sonus. This works > fine but my problem is when I attempt to use X-Lite 3.0 UA behind NAT > on a remote network. Asterisk is not used in this config but proves > that two way audio is fine using SER and the Sonus gateway. > > > > Problem: X-Lite behind NAT on remote network to Sonus PSTN gateway via > SER and Mediaproxy, I get only one ay audio. The audio is fine from > X-Lite to PSTN but not in the reverse direction. > > > > Thanks for your help > > ** ** > > ** ** > > ** ** > > ** ** > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Can I see your PSTN with Accounting?
by Bart Fisher 10 Jan '07

10 Jan '07

Noob here, sorry :) I've been working with SER for about a month and I have a working config using what's provided by iptel.org 'getting started' gw-pstn.cfg - But here's were I'm stuck. I'd like to add accounting, but I haven't found a good example based on gw-pstn.cfg where accounting has been added. Google turns up old updated information. I'm hoping someone can post or email their gw-pstn.cfg with acc enabled: Note: Basically the gw-pstn.cfg script provides a good example of using SER with nathelper and mediaproxy as well as rewriting to a PSTN Gateway Can you help me? TIA Bart bart(a)icpage.com

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[Serusers] Question
by Bill 10 Jan '07

10 Jan '07

I am setting up a SER/Mediaproxy/MySQL box. I have it setup and running except for NAT. When I put up a call this is what my session look like. The 127 are the fakeconversations.py. The other session is my live traffic but the other end has all ????????????????. Any ideas. I am using the basic SER/Mediaproxy config below. Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------- ---------------------------------------- 127.0.0.1:10000 - 10.0.1.100:60004 - 127.0.0.1:10002 active 0'10" Dynamic(127) Audio 94.38k/188.58k/94.38k 10.0.2.1:15040 - 10.0.1.100:60000 - ?.?.?.?:? inactive 4'41" Dynamic(107) Audio 1.58M/0/0 Total traffic: 120.55kbps/74.61kbps/149.22kbps (in1/in2/out) Session count: 2 Proxy version: 1.8.0 #####################SER.CFG [root@voipproxy1 mediaproxy]# cat /usr/local/etc/ser/ser.cfg # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=7 fork=yes log_stderror=no listen=10.0.1.100 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 sock_mode=0666 dns=no rev_dns=no fifo="/tmp/ser_fifo" 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" 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_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) 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("10.0.1.100:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ######################MEDIAPROXY.INI [Dispatcher] defaultProxy = /var/run/mediaproxy.sock [MediaProxy] ;start = yes ;socket = /var/run/mediaproxy.sock ;group = openser ;listen = None ;allow = None proxyIP = 10.0.1.100 portRange = 60000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 [Accounting] ; one of none, radius or database accounting = database [Database] user = ser password = heslo host = localhost database = ser table = acc [Radius] secret = secret server = localhost authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/local/mediaproxy/dictionary retries = 2 timeout = 3 Bill

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[Serusers] ser_ctl
by Jean Sigue 10 Jan '07

10 Jan '07

Hi, After compiling the pre-release 0.10 , I still have the serctl not ser_ctl. How do i fix it Jean R. Sigue Senior Network Engineer CCNP, CSS1, NSA OMX Global Services USA Direct: 1 908-673-6004 Mobile: (917-257-9742) Fax: 1 908-673-6057 Visiting address: 2 Connell Drive Berkeley Heights, NJ 07922 <http://www.omxgroup.com/>www.omxgroup.com No. 1 in marketplace solutions ******************************************************************************** This e-mail and the information it contains may be privileged and/or confidential. It is for the intended addressee(s) only. The unauthorised use, disclosure or copying of this e-mail, or any information it contains, is prohibited. If you are not an intended recipient, please contact the sender and delete the material from your computer. ********************************************************************************

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[Serusers] Error with MTU size in SER with MIPv6
by Jose Angel Calvo 10 Jan '07

10 Jan '07

Thank you for your help and sorry for my English. I have a problem with mipv6 and a server SIP, SER (SIP Express Router). The mipv6 path is "mipv6-2.0.2" from http://www.mobile-ipv6.org/ ,in kernel 2.6.16.24-Ubuntu. The problem occurs when a packet is too big (it must be fragmented), and go to the node what makes function of P-CSCF (Proxy in SIP) and Correspondent Node. If the P-CSCF is not a CN, all is OK. But if the node is CN, then i have the problem. The SER logs are: ############################################################## ................ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: SER: new INVITE Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: parse_headers: flags=-1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: ERROR: warning_builder: buffer size exceeded Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING: warning skipped -- too big Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING:vqm_resize: resize(0) called Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: reply sent out. buf=0x8106270: SIP/2.0 1..., shmem=0xb5e8b330: SIP/2.0 1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: _reply_light: finished Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: mk_proxy: doing DNS lookup... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: DBG: tcp_main_loop: dead child 2 (shutting down?) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: child process 8051 exited by a signal 11 Nov 20 18:28:25 pcscfD kernel: [4296472.851000] <0>skb_under_panic: text:c02d2aad len:321 put:14 head:d62cfc00 data:d62cfbea tail:d62cfd2b end:d62cfd80 dev:eth0 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: core was not generated Nov 20 18:28:25 pcscfD kernel: [4297162.639000] ------------[ cut here ]------------ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: INFO: terminating due to SIGCHLD Nov 20 18:28:25 pcscfD kernel: [4297162.639000] kernel BUG at net/core/skbuff.c:112! Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8057]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8058]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8060]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8061]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8062]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8063]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8064]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8065]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8066]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8067]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8050]: INFO: signal 15 received Nov 20 18:28:25 pcscfD kernel: [4297162.639000] invalid opcode: 0000 [#3] Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: Memory status (pkg): ....................... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2055 fragments no.: 1, unused: 0 ^I^I bucket size: 1048576 - 2097152 (first 1572864) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2059 fragments no.: 1, unused: 0 ^I^I bucket size: 16777216 - 33554432 (first 31934160) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: TOTAL: 26 free fragments = 33537656 free bytes Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: ----------------------------- Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: shm_mem_destroy Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: destroying the shared memory lock Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: terminating due to SIGCHLD ############################################################## The problem is in the MTU size, because if the packet is not fragmented, there is not a problem. But the INVITE packet needs to fragment, what it means, i always have the same problem. My SER version is version: ser 0.9.6 (i386/linux) and it's a vanilla version. The packets is sent over UDP. My questions are: - Have anybody the same o similar problem? - The MTU size is a problem in SER? - Have anybody sent fragmented packets with SER? Thanks for your help. __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.es

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[Serusers] SER LCR Config
by Arun Kumar 10 Jan '07

10 Jan '07

Hi, I'm trying to configure LCR with ser here is my config file. This config is sending the call to my first gateway but when My first gateway is down its not sending the calls to other gateways. Plese help me to make this working. ------------------------ debug=3 fork=no log_stderror=yes listen=192.168.2.210 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:pass@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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/lcr.so" modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " mysql://ser:pass@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_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 120) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("lcr","db_url", "mysql://ser:pass@localhost/ser") modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","grp_id_column","grp_id") modparam("lcr","lcr_table","lcr") modparam("lcr","prefix_column","prefix") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","contact_avp","1401") modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") modparam("lcr","fr_inv_timer",90) modparam("lcr","fr_inv_timer_next",60) # ------ Account and Radius modules-------------- modparam("acc", "db_url", "mysql://ser:pass@localhost/ser") modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc", "db_flag", 1) modparam("acc","radius_flag",1) modparam("acc","service_type",15) #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") 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")) { # setflag(1); # INSERT YOUR IP ADDRESS HERE record_route_preset("192.168.2.210:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden Cant cross 1st bridge"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { sl_send_reply("404", "I am trusted host, "); route(4); route(1); } else { sl_send_reply("403", "Forbidden, I am here"); }; break; }; if (method=="ACK") { setflag(1); route(1); break; } if (method=="CANCEL") { setflag(1); route(1); break; } else if (method=="INVITE") { setflag(1); # sl_send_reply("410", "I am sending your call, "); route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { setflag(1); end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; route(4); if(!load_gws()){ sl_send_reply("500", "Internal server error, unable to load gateways"); break; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); break; }; if (!is_uri_host_local()) { route(4); route(1); break; }; # route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ------------------------------- thanks arun

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[Users] uri_radius and radius_does_user_exist() not working
by Toni Heinonen 09 Jan '07

09 Jan '07

Hi, I've been through the OpenSER + RADIUS configuration tutorial many, many times, and it works like a charm, except for the uri_radius module which I can't get to detect user existence with the radius_does_user_exist() function. I'm trying to decide whether to send a 404 or a 480, as such: if (!lookup("location")) { if(radius_does_uri_exist()) { sl_send_reply("480", "User offline"); } else { sl_send_reply("404", "User not found"); }; }; It worked identically with the uri_db module's does_user_exist() function. Now the RADIUS server doesn't seem to understand the Call-Check request. I have a dump here: http://tonih.iki.fi/temp/uri_radius.cap Can anyone guess why OpenSER always gives a 404 even for users that exist, but that are simply offline? The users are currently hand-configured into freeradius's text configuration file users.conf. PS. What have you found to be the best way to authenticate users from a domain? FreeRADIUS using Kerberos 5, LDAP or relaying to a Microsoft RADIUS (IAS) server? -- http://tonih.iki.fi/ ~ http://blogit.helsinki.fi/toni.heinonen/ "The progress of a dynamic civilization depends on the special people who make play out of work. In their all-absorbing passion, they create the variations that, through trial and error, become the sources of progress. They make the discoveries that drive the infinite series." - Virginia Postrel

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[Users] sip register
by sinclair fils 09 Jan '07

09 Jan '07

i am developping one sip dialer using Microsoft RTC in vb6 i want to register in my opnser server put no request of registration are send to my server any help __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail

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[Users] NEW FEATURE: error_route - automatic error handling
by Daniel-Constantin Mierla 09 Jan '07

09 Jan '07

Hello, today was introduced in development version a framework for automatic error handling. Based on past experiences and comments on the mailing lists, error handling was a bit deficient from the configuration file. Log files could get full of error messages when SIP messages were incorrect, the processing continued even it should have stopped. With the new framework, the administrator can write a error_route block and there decide what to do when an error is encountered. Right now, error_route is executed for parsing errors in sip requests - one can send back now the appropriate reply. I use this email to start a discussion about how to extend it, there could be system errors (e.g., memory, database access, ...) or errors in replies (here we can use same route block and check the type of sip message not to force reply to reply - a new more flexible and easier way to access pseudo-variables is underway). In the error_route are available 4 new pseudo-variables: - $err.class - the class of error (now is ‘1’ for parsing errors) - $err.level - severity level for the error - $err.info - text describing the error - $err.rcode - recommended reply code - $err.rreason - recommended reply reason phrase With the new sl_reply() from sl module you can reply with the appropriate code and reason from error_route[] - here is an example: error_route { xlog("--- error route class=$(err.class) level=$(err.level) info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n"); xlog("--- error from [$si:$sp]\n+++++\n$mb\n++++\n"); sl_reply("$err.rcode", "$err.rreason"); exit; } In addition to sl_send_reply(), append_to_reply() and xlog functions can be used in error_route. You have to use 'exit' if you do not want to continue the execution when an error occurred. Error level is meant to show if the processing can continue without any issue, or if the error allows to send reply or not. Please note that malformed SIP messages can still go through your proxy in the case the processing didn't discover it (OpenSER has an incremental parser, it parses only what it needs). If no error_route is defined, the functionality is same as before. Looking forward to feedback about this new automatic error handling. Cheers, Daniel

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[Serusers] serweb 0.10.0 - error: Allowed memory size of 8388608 bytes exhausted
by "tzieleniewski" 09 Jan '07

09 Jan '07

Hi After instaling serweb 0.10.0 for ser 0.10.99 when I try to log into user aplication I get the following error: Fatal error: Allowed memory size of 8388608 bytes exhausted (tried to allocate 9699 bytes) in /home/ser/serweb/serweb-0.10.0/smarty/Smarty_Compiler.class.php on line 196 I configured serweb with the multidomain support according to README in modules/multidomain/. What can I do with that? And what information do You need to investigate the probelem? Bests Tomasz

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[Users] Re: [Devel] OpenSER not sending Ack on Request Cancel
by Klaus Darilion 09 Jan '07

09 Jan '07

Rajesh Kalagarla wrote: > Hi Klaus, > > As per rfc 3261, if the proxy is stateful the CANCEL request will be a > hop-by-hop request. in that case Relay is not the correct option right? It is the correct function to handle CANCEL. t_relay will pass the CANCEL to openser's transaction modul. The transaction module usualy will detect that this CANCEL is for an ongoing transaction and takes care of the hop-by-hop CANCEL. This should also take care of the hop-by-hop ACK to the 487 response. regards klaus > > Thanks & Regards, > Rajesh > > ----- Original Message ----- > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > To: "Rajesh Kalagarla" <RajeshK(a)idnltd.com> > Cc: <devel(a)openser.org> > Sent: Tuesday, January 09, 2007 10:38 AM > Subject: Re: [Devel] OpenSER not sending Ack on Request Cancel > > >> Hi! >> >> This indeed looks strange. Are you using stateful forwarding (t_relay)? >> >> It loks like your are using lookup() also for CANCEL - try to avoid that >> and just t_relay: >> >> if (is_method("CANCEL")) { >> t_relay(); >> } >> >> regards >> klaus >> >> Rajesh Kalagarla wrote: >>> Hi Klaus, >>> >>> here is the log. >>> >>> Thanks & Regards, >>> Rajesh >>> ----- Original Message ----- >>> From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> >>> To: "Rajesh Kalagarla" <RajeshK(a)idnltd.com> >>> Cc: <devel(a)openser.org> >>> Sent: Monday, January 08, 2007 1:00 PM >>> Subject: Re: [Devel] OpenSER not sending Ack on Request Cancel >>> >>> >>>> Rajesh Kalagarla wrote: >>>>> Hello All, >>>>> >>>>> OpenSER is not responding with a Ack for the call that was terminated >>> with CANCEL because of no response from remote user. because of this SIP >>> client is keep retransmitting the 487 Response for the Invite request. >>>>> is it a bug or something need to be enabled? >>>> Please send a "ngrep -W byline port 5060" dump. >>>> >>>> regards >>>> klaus >>>> >>>> >>>>> Thanks, >>>>> Rajesh >>>>> >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>>> _______________________________________________ >>>>> Devel mailing list >>>>> Devel(a)openser.org >>>>> http://openser.org/cgi-bin/mailman/listinfo/devel >>>> -- >>>> Klaus Darilion >>>> nic.at >>>> >>>> >> >> -- >> Klaus Darilion >> nic.at >> >> > > -- Klaus Darilion nic.at

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RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 09 Jan '07

09 Jan '07

Behind a PIX, it works Behind a SonicWall without the SIP_aware option, it doesn't work Behind a Watchguard, it doesn't work too (and there is no SIP_aware option) In fact, I have a centrex IP and many clients in the country with, many different configurations and I have to be sure that all works. It's why I have no enable the SIP_aware option in the sonicwall. Thanks for your help, Thomas -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Thomas Deillon Envoyé : mercredi, 3. janvier 2007 17:06 À : daniel(a)voice-system.ro Cc : users(a)openser.org Objet : RE: [Users] SER as proxy in front of Asterisk Hi, In fact, now, every phone does not response. The options didn't go thought the FW, so, the phone cannot answer. I don't understand why the traffic don't go thought it !! I have test this configuration with two firewall: a sonicwall and a soho. Cheers, Thomas -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Envoyé : mercredi, 3. janvier 2007 14:52 À : Thomas Deillon Cc : users(a)openser.org Objet : Re: [Users] SER as proxy in front of Asterisk Hello, what is the type of NAT/Firewall you have in front of Thomson? As I can see from the diagram, the phone does not reply to OPTIONS request, and I guess, the nat/firewall needs outward traffic to keep the pinhole open. Check why thomson ignores the options requests. Cheers, Daniel On 01/03/07 15:17, Thomas Deillon wrote: > > Hi, > > I have made more tries and now, all my phones don't work ... > > The firewall opens a connection to REGISTER phone but after a while > the OPTION message from the Asterisk will be DROP by the FW ... > > You can find a diagram of the exchange here: > http://deillon.eu/tmp/exchange_sip.png > > A list of all exchange: http://deillon.eu/tmp/ip_sip.txt > > And all sip message: http://deillon.eu/tmp/message.txt > > Thanks a lot for your help, > > Thomas > > ------------------------------------------------------------------------ > > *De :* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *De la part de* Thomas Deillon > *Envoyé :* mardi, 2. janvier 2007 11:44 > *À :* users(a)openser.org > *Objet :* [Users] SER as proxy in front of Asterisk > > Hi all, > > I wish you first a happy new year !! > One again, I ask you some help. Thanks a lot for your patience and > your answers that really helped me. > > So, I want to put a SER in front of one Asterisk for the moment (more > after). > All Servers have public IP address and so, I don't care about the > RTP/SDP messages. > All phones are behind NAT somewhere in Switerland :) > > > I have the configuration below in openser: > > > modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") > #modparam("dispatcher", "force_dst", 1) > modparam("dispatcher", "flags", 2) > > modparam("usrloc", "db_mode", 0) > #modparam("rr", "enable_full_lr", 1) > > route{ > > > xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > # if (search("User-Agent:.*Thomson.*")) { > # }; > > > if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { > if(method=="OPTIONS") { > }; > avp_pushto("$ru","$tu"); > forward(); > > }else { > fix_nated_contact(); > force_rport(); > if(method=="REGISTER"){ > ds_select_dst("4", "0"); > t_relay(); > } > else { > ds_select_dst("0", "4"); > forward(); > } > } > } > > > And in my dispatcher.list I have: > > 0 sip:212.xxx.xxx.153:5060 > 4 sip:212.xxx.xxx.153:5060 > > > > With this configuration, a snom phone, a X-lite phone and a Cisco > phone seams to work .... > But with the Thomson ST2030, it doesn't work. > In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION > message to the FW IP adress and the right port where the thomson is > and then, after a while, the FW close the connection. The thomson > phone is so "UNREACHABLE" on asterisk status and nobody can call it. > I'm not sure that my configuration is ok in ser but I think that is a > problem with the Thomson2030. Do you had the same kind of problem or > do you understand the problem here ? > > > Thanks for your help, > > Thomas Deillon > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Failover with messages BYE
by Jose Gil Navarrete 09 Jan '07

09 Jan '07

Hi list. I try to use the module dispatcher like failover, when the call is established and after the proxy is down then resend the BYE message at the next proxy using ds_select_dst("2", "4") and forward the message, but I obtain the message 503 of the failure_route[2], Any idea?? Thanks in advance In my route seccion I made the next: if (loose_route()) { if (search("Path")) { route(1); }; if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); exit; }; route(5); }; route[5] { t_on_failure("2"); t_relay(); exit; } failure_route[2] { #500, 503, 504 if ( t_check_status("408") | t_check_status("500") | t_check_status("503") | t_check_status("504") ) { if (ds_next_domain()) { #if (ds_next_dst()) { # On Failure: Choose next Proxy t_on_failure("2"); # Relay to Proxy t_relay(); } else { t_reply("503", "Service not available"); return; } }; } __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/

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[Serusers] Work with a new column in SER.subscriber
by inge 09 Jan '07

09 Jan '07

Hi all, I add a new column named "client" in subscriber table. Next, I want to do a test wich seems like that in SQL : "SELECT client FROM subscriber WHERE username = "actual calling number" => equivalent of avp_db_load I think ? Then, I think I must do a test on the previous values with an if and avp_check I think. But my problem is to do this both requests with SER syntax in ser.cfg. Someone can help me ? Thanks

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[Serusers] ser_ctl: (111, 'Connection refused'): error
by "tzieleniewski" 09 Jan '07

09 Jan '07

Hi ! I've set up the ser 0.10.99, basic functionality works - ser registers users and establishes the media sessions correctly. Now I tried to use the new serctl tool: version 0.3.17. When I try to invoke for instance #ser_ctl ps while ser is running I get the following: ser_ctl: (111, 'Connection refused'): error my ser.cfg file part: port=5060 ... modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") modparam("ctl", "mode", 0666) and my serctl.conf and serctl/config.py files lines: .. SER_URI = 'http://localhost:5060/' ... Please point me what do I need to do to make it work. Best Tomasz

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Re: Re: Re: [Serusers] ser_ctl: (111, 'Connection refused'): error
by Michal Matyska 08 Jan '07

08 Jan '07

I did copy&paste from one of my testing scripts I have on disk .... as it contained the create_via (which is not neccesary now) it was designed for the older xml-rpc way of handling. You can safely use the code from the CVS ser.cfg example, it's kept up to date. Michal On Mon, 2007-01-08 at 19:55 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > I have something like this > if ((method=="POST" || method=="GET") & src_ip==127.0.0.1) { > if (msg:len >= 8192) { > sl_send_reply("513", "Request to big"); > drop; > } > dispatch_rpc(); > drop; > } > taken from the latest cvs etc/ser.cfg file. > Is this rewriteuri(...) required? Does it imply that sip:xmlrpc.. is somehow reserved? > > > > The ctl module serves the sercmd (utils/sercmd). > > > > If you want to use ser_ctl then you have to support it in the script > > file like this: > > > > if (method=="POST" || method=="GET") { > > # create_via(); shouldn't be required with the latest CVS source > > rewriteuri("sip:xmlrpc@127.0.0.1:5060"); > > dispatch_rpc(); > > drop; > > } > > > > Michal > > > > On Mon, 2007-01-08 at 17:26 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > Results in the attached files > > > > > > tomasz > > > > > > > Plese, try 'netstat -tupan' and 'ser_ctl -Gg ps' and post results. -- hallik > > > > > > > > On Mon, Jan 08, 2007 at 04:41:59PM +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > > > Hi ! > > > > > > > > > > I've set up the ser 0.10.99, basic functionality works - ser registers users > > > > > and establishes the media sessions correctly. Now I tried to use the new serctl > > > > > tool: version 0.3.17. When I try to invoke for instance #ser_ctl ps while ser is > > > > > running I get the following: > > > > > ser_ctl: (111, 'Connection refused'): error > > > > > > > > > > my ser.cfg file part: > > > > > port=5060 > > > > > ... > > > > > modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") > > > > > modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") > > > > > modparam("ctl", "mode", 0666) > > > > > > > > > > and my serctl.conf and serctl/config.py files lines: > > > > > .. > > > > > SER_URI = 'http://localhost:5060/' > > > > > ... > > > > > > > > > > Please point me what do I need to do to make it work. > > > > > Best > > > > > Tomasz > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > 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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Re: Re: Re: [Serusers] ser_ctl: (111, 'Connection refused'): error
by Michal Matyska 08 Jan '07

08 Jan '07

serctl (note the missing underscore) was the only tool for ser < 0.9.x ser 0.10 introduced xmlrpc interface (basically the dispatch_rpc in the script calls the function) and ser_ctl family of tools was introduced (ser_rpc just for the XML-RPC call, ser_user, ser_domain, ..... specialized (the name suggests what they do), ser_ctl as the shell around all of them. They are written in python. Due to the fact of XML parsing, etc. the binary protocol was re-introduced. You need to load the ctl module, set the protocol, address, port) and use sercmd as cli. All methods which are callable via XML-RPC are visible via sercmd, it is just protocol between ser and controller. So if you have configured both was you can use ser_rpc and sercmd interchangeable. For the methods available use: sercmd system.listMethods Michal P.S. Please keep always the list in To/Cc to follow the thread. Somebody else might be interested as well. On Mon, 2007-01-08 at 20:01 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > Just to make sure:) > So sercmd is a tool used to send command directly so through ser unix socket and rpc calls are entrily handled by xmlrpc module after dispatch_rpc() method?? > > > > The ctl module serves the sercmd (utils/sercmd). > > > > If you want to use ser_ctl then you have to support it in the script > > file like this: > > > > if (method=="POST" || method=="GET") { > > # create_via(); shouldn't be required with the latest CVS source > > rewriteuri("sip:xmlrpc@127.0.0.1:5060"); > > dispatch_rpc(); > > drop; > > } > > > > Michal > > > > On Mon, 2007-01-08 at 17:26 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > Results in the attached files > > > > > > tomasz > > > > > > > Plese, try 'netstat -tupan' and 'ser_ctl -Gg ps' and post results. -- hallik > > > > > > > > On Mon, Jan 08, 2007 at 04:41:59PM +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > > > Hi ! > > > > > > > > > > I've set up the ser 0.10.99, basic functionality works - ser registers users > > > > > and establishes the media sessions correctly. Now I tried to use the new serctl > > > > > tool: version 0.3.17. When I try to invoke for instance #ser_ctl ps while ser is > > > > > running I get the following: > > > > > ser_ctl: (111, 'Connection refused'): error > > > > > > > > > > my ser.cfg file part: > > > > > port=5060 > > > > > ... > > > > > modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") > > > > > modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") > > > > > modparam("ctl", "mode", 0666) > > > > > > > > > > and my serctl.conf and serctl/config.py files lines: > > > > > .. > > > > > SER_URI = 'http://localhost:5060/' > > > > > ... > > > > > > > > > > Please point me what do I need to do to make it work. > > > > > Best > > > > > Tomasz > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > 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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[Users] Caller-ID setup help
by Chahn Kim 08 Jan '07

08 Jan '07

Hi, a Happy New Year to all. Looking for help to get our caller-id working for outging PSTN calls. We have 3rd party PSTN gateway service and all our callee gets the gateway phone number as CID for all of our sip calls to PSTN. I had implemented so far sip-rpid, remote-party-id, through avp, append_rpid_hf() and confirmed rpid headers on all pstn outgoing frames. However, callee only see the SP's PSTN gateways number as incoming caller-id. Is there other way of implementing caller-id for sip calls before I bring the subject to PSTN service providor? Or is it most likely SP's policy setup? Thanks in advnace. John K

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[Serusers] Instant Messaging Chat/Presence Client Please
by Gould, Aaron 08 Jan '07

08 Jan '07

Does anyone know of a free SIP IM/Presence Chat client that allows for multiple users conferencing? Aaron

2 1

[Users] Chunking Messages
by Danilo Vilela Dantas 08 Jan '07

08 Jan '07

Hi Congratulations! Does Openser supports chunking messages? Is it support at MSRP Relay? Thanks! Danilo

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Re: [Serusers] Problemswith compiling ser 0.10.99
by Dmitry Isakbayev 08 Jan '07

08 Jan '07

Hi, The OSP toolkit project is being moved from sipfoundry to sourceforge. It should be available shortly. Meanwhile, the latest snapshot of the toolkit can be downloaded from http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit.htm Regards, Dmitry > Date: Mon, 08 Jan 2007 11:25:24 +0100 > From: "tzieleniewski" <tzieleniewski(a)o2.pl> > Subject: [Serusers] Problemswith compiling ser 0.10.99 > To: serusers(a)lists.iptel.org > Message-ID: <54c54dce.20e2e287.45a21c14.f0624(a)o2.pl> > Content-Type: text/plain; charset="utf-8" > > Hi All!! > I am trying to compile the lattest cvs version of ser but I get errors > which I cann't handle myself so i kindly ask for your help:) > What I want to achive basically is the ser instance with presence > support. > I am compiling ser in the following way: > make group_include="standard standard-dep stable" all > I read that osp requires osptoolkit i tried to get it from sipfoundry > but i couldn't find any source of that. > I also cann't fight problems with tls i didn't includethe experimental > modules(tls) but still I get some errors. > Below is the make output > > >

1 0

[Serusers] Sipp and SER using TCP protocol
by Zeeshan Sajid 08 Jan '07

08 Jan '07

Hey ppl, I am trying to find out the peak number of calls that ser supports using sipp running the default scenario. I am using SER-0.9.6 and sipp version1.8 for this purpose. Now my setup on a local lan is SIpp (UAC) -----------------------> SER --------------------> Sipp (UAS) 192.168.0.8 192.168.0.200 192.168.0.2 the commands that i have used are for Sipp (UAC) ulimit -n 99999 ./sipp -sn uac -rsa 192.168.0.200:5060 -i 192.168.0.8 -p 5061 192.168.0.2:5060 -t t1 -r 1000 -rp 10 for Sipp (SER) default settings for Sipp (UAS) ulimit -n 99999 ./sipp -sn uas -i 192.168.0.2 -p 5060 -rsa 192.168.0.200:5060 -t tn now the problem is that after a while the UAS crashes saying "Broken pipe on TCP connection, remote peer probably closed the socket." my understanding is that SER is unable to handle the messages burst to the Sipp UAS and so please tell me how can i solve this problem .... thanks in advance regards Zeeshan Sajid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] ser 0.10.99 problems with ser.cfg
by "tzieleniewski" 08 Jan '07

08 Jan '07

Hi!! Right now I am trying to migratemy ser.cfg file from ser0.9.7 to ser 0.10.99 when I try to check the ser.cfg file structure I get errors: 0(28672) parse error (252,16-17): unknown command, missing loadmodule? the errors correspond to the following line in the ser.cfg file: www_challenge("scscf.ims.touk.pl","0"); if (!check_to()) { proxy_challenge("scscf.ims.touk.pl","0"); } else if (!check_from()) { below is the list of my loaded modules: loadmodule "/home/ser/src/ser-0.10.x/modules/ctl/ctl.so" loadmodule "/home/ser/src/ser-0.10.x/modules/mysql/mysql.so" loadmodule "/home/ser/src/ser-0.10.x/modules/sl/sl.so" loadmodule "/home/ser/src/ser-0.10.x/modules/tm/tm.so" loadmodule "/home/ser/src/ser-0.10.x/modules/rr/rr.so" loadmodule "/home/ser/src/ser-0.10.x/modules/maxfwd/maxfwd.so" loadmodule "/home/ser/src/ser-0.10.x/modules/usrloc/usrloc.so" loadmodule "/home/ser/src/ser-0.10.x/modules/registrar/registrar.so" loadmodule "/home/ser/src/ser-0.10.x/modules/auth/auth.so" loadmodule "/home/ser/src/ser-0.10.x/modules/auth_db/auth_db.so" loadmodule "/home/ser/src/ser-0.10.x/modules/uri/uri.so" loadmodule "/home/ser/src/ser-0.10.x/modules/uri_db/uri_db.so" loadmodule "/home/ser/src/ser-0.10.x/modules/domain/domain.so" loadmodule "/home/ser/src/ser-0.10.x/modules/mediaproxy/mediaproxy.so" loadmodule "/home/ser/src/ser-0.10.x/modules/nathelper/nathelper.so" loadmodule "/home/ser/src/ser-0.10.x/modules/textops/textops.so" loadmodule "/home/ser/src/ser-0.10.x/modules/avpops/avpops.so" loadmodule "/home/ser/src/ser-0.10.x/modules/xlog/xlog.so" Are those functions not present in ser 0.10.99 any more? Bests Tomasz

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[Users] 2 Instances of OpenSER on the same server
by Gerardo Amaya 08 Jan '07

08 Jan '07

Hello All. I'm trying to install 2 Instances of openser on the same machine with the following interfaces: Instance #1 -----> 172.16.18.2 and 192.168.1.112 Instance #2 -----> 172.16.1.2 and 192.168.1.113 I made one installation on the default OpenSER folders and one in the /usr/loca/openser_1/ The configuration went quite smooth. The first instance of the server is running with no problem at all, but when I try to start the second instance I got an error that it was unable to resolve the IP addresses I add the following lines to the /etc/hosts file: ### OpenSER 1st interface configuration #### 192.168.1.112 openser1_internal 172.16.18.2 openser1_external ### OpenSER 1st Interface configuration #### ### OpenSER 2nd Interface configuration #### 192.168.1.113 openser1_internal 172.16.1.2 openser1_external ### OpenSER 2nd Interface configuration #### After that Now I have the following error: Jan 5 17:26:30 atel-router ./openser[17772]: ERROR: tcp_init: bind(7, 0x813e194, 16) on 192.168.1.112:5060 : Address already in use Any Help will be appreciated, of better is there a How-to on OpenSER multiple instances please let me know. Thanks in advance Gerardo Amaya Here is the configuration file for each instance: OpenSER.cfg Instance #1 # # $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=5 # 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) listen=192.168.1.112 listen=172.16.18.2 port=5060 mhomed=1 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/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) # -- nathelper and mediaproxy params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # -- End of nathelper and mediaproxy params -- # -- auth_db database params for domain.so module to work properly -- modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://root:codevoz@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # -- End of database params -- # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method == "REGISTER" || ! search("^Record-Route:")) { fix_nated_contact(); # Rewrite contact with source IP of signalling #if (method == "INVITE") { # fix_nated_sdp("1"); # Add direction=active to SDP #}; force_rport(); # Add rport parameter to topmost Via } if (method == "INVITE") { if (dst_ip == 192.168.1.112){ if (force_rtp_proxy("FAII")) t_on_reply("1"); } else if (dst_ip == 172.16.18.2){ if (force_rtp_proxy("FAEI")) t_on_reply("1"); } else if (dst_ip == 192.168.1.112){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); } else if (dst_ip == 172.16.18.2){ if (force_rtp_proxy("FAEE")) t_on_reply("1"); } }; # 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 # This is the Call Tear Down Section if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; route(1); } route[1] { t_on_reply("1"); #force_rtp_proxy(); t_relay("udp:192.168.1.2:5060"); } onreply_route[1] { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } OpenSER.cfg Instance #2 # # $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=5 # 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) listen=udp:192.168.1.113:5060 listen=udp:172.16.1.2:5060 #port=5060 mhomed=1 children=4 fifo="/tmp/ser_fifo2" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/openser_1/lib/openser/modules/sl.so" loadmodule "/usr/local/openser_1/lib/openser/modules/tm.so" loadmodule "/usr/local/openser_1/lib/openser/modules/rr.so" loadmodule "/usr/local/openser_1/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/openser_1/lib/openser/modules/usrloc.so" loadmodule "/usr/local/openser_1/lib/openser/modules/registrar.so" loadmodule "/usr/local/openser_1/lib/openser/modules/textops.so" loadmodule "/usr/local/openser_1/lib/openser/modules/uri.so"loadmodule "/usr/local/openser_1/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/openser_1/lib/openser/modules/auth.so" loadmodule "/usr/local/openser_1/lib/openser/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) # -- nathelper and mediaproxy params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy_1.sock") # -- End of nathelper and mediaproxy params -- # -- auth_db database params for domain.so module to work properly -- modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://root:codevoz@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # -- End of database params -- # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method == "REGISTER" || ! search("^Record-Route:")) { fix_nated_contact(); # Rewrite contact with source IP of signalling #if (method == "INVITE") { # fix_nated_sdp("1"); # Add direction=active to SDP #}; force_rport(); # Add rport parameter to topmost Via } if (method == "INVITE") { if (dst_ip == 192.168.1.113){ if (force_rtp_proxy("FAII")) t_on_reply("1"); } else if (dst_ip == 172.16.1.2){ if (force_rtp_proxy("FAEI")) t_on_reply("1"); } else if (dst_ip == 192.168.1.113){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); t_on_reply("1"); } else if (dst_ip == 192.168.1.113){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); } else if (dst_ip == 172.16.1.2){ if (force_rtp_proxy("FAEE")) t_on_reply("1"); } }; # 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 # This is the Call Tear Down Section if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; route(1); } route[1] { t_on_reply("1"); #force_rtp_proxy(); t_relay("udp:192.168.1.2:5060"); } onreply_route[1] { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } loadmodule "/usr/local/openser_1/lib/openser/modules/uri_db.so" loadmodule "/usr/local/openser_1/lib/openser/modules/domain.so" loadmodule "/usr/local/openser_1/lib/openser/modules/mediaproxy.so" ifconfig output eth0 Link encap:Ethernet HWaddr 00:0E:2E:0A:61:68 inet addr:172.16.18.2 Bcast:172.16.18.255 Mask:255.255.255.0 inet6 addr: fe80::20e:2eff:fe0a:6168/64 Scope:Link UP BROADCAST MULTICAST MTU:1500 Metric:1 RX packets:216465 errors:0 dropped:0 overruns:0 frame:0 TX packets:323925 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:18190685 (17.3 MiB) TX bytes:118579358 (113.0 MiB) Interrupt:185 Base address:0xaf00 eth1 Link encap:Ethernet HWaddr 00:0E:2E:06:EC:17 inet addr:172.16.1.2 Bcast:172.16.1.255 Mask:255.255.255.0 UP BROADCAST MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) Interrupt:209 Base address:0xce00 eth2 Link encap:Ethernet HWaddr 00:0C:76:BE:E5:87 inet addr:192.168.1.112 Bcast:192.168.1.255 Mask:255.255.255.0 inet6 addr: fe80::20c:76ff:febe:e587/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1155321 errors:0 dropped:0 overruns:0 frame:0 TX packets:664343 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:187622098 (178.9 MiB) TX bytes:125573054 (119.7 MiB) Interrupt:217 Base address:0xed00 eth2:0 Link encap:Ethernet HWaddr 00:0C:76:BE:E5:87 inet addr:192.168.1.113 Bcast:192.168.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 Interrupt:217 Base address:0xed00

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[Users] Surefire Failure
by Bill Neely 08 Jan '07

08 Jan '07

Does anybody have a way to force an unconditional branch to a failure route. Or, if not that, a command that will surely fail and force a branch to the failure route. Bill -- 702-874-3833 1-866-553-3833

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[Users] problem forwarding ACKs when using IPv6
by Emil Ivov 08 Jan '07

08 Jan '07

Hello all, We are currently performing a number of interoperability tests between OpenSER and SIP Communicator over IPv6. The good news is that we have successfully established IPv6 calls between two SIP Communicator instances registered on our local OpenSER installation (version 1.1.0-7 on Debian testing). There is however something weird we'd appreciate any ideas on what might be causing it: When running OpenSER with a default configuration, any ACKs that the caller tries to proxy through OpenSER don't get forwarded to the callee. Changing syn_branch to "no" (we saw somewhere that this might be related) didn't have any effect. I am attaching a log snippet that seems to be corresponding to the ACK processing. In a desperate attempt to resolve the issue we added the following to our route{} clause: if (method == "ACK") { forward("[2001:660:4701:f009:213:ceff:fea7:a88c]:5060"); exit; } where [2001:660:4701:f009:213:ceff:fea7:a88c] is the IPv6 address of the callee. This worked but it is obviously not (even close to) a "viable" solution. So our question is, what would be the best way to make OpenSER forward ACKs that a caller is trying to send to a callee? Thanks Emil 2(11235) receive_msg: cleaning up 4(11237) SIP Request: 4(11237) method: <ACK> 4(11237) uri: <sip:emcho@ipv6.sip-communicator.org> 4(11237) version: <SIP/2.0> 4(11237) parse_headers: flags=2 4(11237) Found param type 232, <branch> = <z9hG4bKc01a084c923dbc4a5de11530d611fe64>; state=16 4(11237) end of header reached, state=5 4(11237) parse_headers: Via found, flags=2 4(11237) parse_headers: this is the first via 4(11237) After parse_msg... 4(11237) preparing to run routing scripts... 4(11237) DEBUG : sl_filter_ACK: to late to be a local ACK! 4(11237) parse_headers: flags=100 4(11237) get_hdr_field: cseq <CSeq>: <1> <ACK> 4(11237) DEBUG: add_param: tag=b9b9c012 4(11237) DEBUG:parse_to:end of header reached, state=29 4(11237) DBUG:parse_to: display={"emcho"}, ruri={sip:emcho@ipv6.sip-communicator.org} 4(11237) DEBUG: get_hdr_field: <To> [60]; uri=[sip:emcho@ipv6.sip-communicator.org] 4(11237) DEBUG: to body ["emcho" <sip:emcho@ipv6.sip-communicator.org>] 4(11237) DEBUG:maxfwd:is_maxfwd_present: value = 70 4(11237) DEBUG: add_param: tag=b7cb05bc 4(11237) DEBUG:parse_to:end of header reached, state=29 4(11237) DBUG:parse_to: display={"77"}, ruri={sip:77@ipv6.sip-communicator.org} 4(11237) parse_headers: flags=200 4(11237) is_preloaded: No 4(11237) grep_sock_info - checking if host==us: 25==25 && [ipv6.sip-communicator.org] == [ipv6.sip-communicator.org] 4(11237) grep_sock_info - checking if port 5060 matches port 5060 4(11237) after_strict: Next hop: 'sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on' is loose router 4(11237) parse_headers: flags=ffffffffffffffff 4(11237) DEBUG: get_hdr_body : content_length=0 4(11237) found end of header 4(11237) rewrite_uri: Rewriting Request-URI with 'sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on' 4(11237) after_strict: The last route URI: 'sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on' 4(11237) parse_headers: flags=ffffffffffffffff 4(11237) DEBUG: t_newtran: msg id=3 , global msg id=2 , T on entrance=0xffffffff 4(11237) parse_headers: flags=ffffffffffffffff 4(11237) parse_headers: flags=78 4(11237) t_lookup_request: start searching: hash=42196, isACK=1 4(11237) parse_headers: flags=38 4(11237) DEBUG: t_lookup_request: e2e proxy ACK found 4(11237) DEBUG:tm:REF_UNSAFE: after is 1 4(11237) DEBUG:tm:t_newtran: building branch for end2end ACK 4(11237) DEBUG:tm:UNREF_UNSAFE: after is 0 4(11237) DEBUG:tm:t_relay: forwarding ACK 4(11237) DEBUG: mk_proxy: doing DNS lookup... 4(11237) check_via_address(2001:660:4701:F009:215:FF:FE35:EE1F, 0.0.0.0, 0) 4(11237) Sending: ACK sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on SIP/2.0 Record-Route: <sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];lr=on;ftag=b7cb05bc> Via: SIP/2.0/UDP [2001:660:4701:1001:20E:7FFF:FEB5:AAA3];branch=z9hG4bK4d4a.9db19213.2 Via: SIP/2.0/UDP 0.0.0.0;received=2001:660:4701:F009:215:FF:FE35:EE1F;branch=z9hG4bKc01a084c923dbc4a5de11530d611fe64 CSeq: 1 ACK Call-ID: 39bdf32f765ff3a4b67d1eedc2d538c9(a)0.0.0.0 From: "77" <sip:77@ipv6.sip-communicator.org>;tag=b7cb05bc To: "emcho" <sip:emcho@ipv6.sip-communicator.org>;tag=b9b9c012 User-Agent: SIP Communicator 1.0 CVS-Fri_Jan_05_18-27-08_EET_2007 Max-Forwards: 69 Content-Length: 0 P-hint: rr-enforced . 4(11237) orig. len=490, new_len=679, proto=1 4(11237) DEBUG:destroy_avp_list: destroying list (nil)

2 1

[Users] RTPProxy and SER, Multiple Interface Server
by Gerardo Amaya 08 Jan '07

08 Jan '07

Hello All. After investigating quite a while about this issue I'm kind of desperate now, I will appreciate your help on this. I have a OpenSER Server with multiple interfaces, I'm able to handle all the SIP packets with the multi homed enabled. Now the part that I'm having real trouble with is the RTP audio streams. After struggling with MediaProxy I finally started using RTPProxy for this, since it has bridge mode available. Now the problem is that I have more than one network to connect audio from, so the internal and external configuration is not enough for me. So my questions are: 1.-Can I have RTPProxy Bridge mode working for all the interfaces in my server? 2.-Can I have more than one RTPProxy socket available for each of the multiple interfaces I'm connecting in order to have the audio working, so each route will use the proxy of their own network? 3.-If #2 is true, is this where the force_socket and force_rtp_proxy functions play a role in my configuration? Please help!! Thanks you so much in advance Gerardo Amaya

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[Users] Re: [Devel] openserctl user addition problem
by Klaus Darilion 08 Jan '07

08 Jan '07

Rajesh Kalagarla wrote: > Hi Klaus, > > Thanks for your response. > now i am trying to add user using openserctl only. > but when i am giving a command to add the user " command: openserctl add > 1122 earth rajeshk(a)idnltd.com" > > output is, ERROR 1045(28000): Access denied for user openser@localhost > (using PASSWORD: YES) > > the query inside the openserctl looks to be failing. bingo. you are right. you have to configure openserctl to use the proper username/password. take a look at /etc/openser/openserctlrc regards klaus > > Thanks, > Rajesh > > ----- Original Message ----- > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > To: "Rajesh Kalagarla" <RajeshK(a)idnltd.com> > Cc: <devel(a)openser.org> > Sent: Thursday, January 04, 2007 1:29 PM > Subject: Re: [Devel] openserctl user addition problem > > >> This has nothing todo with your openser.cfg. openserctl needs to know >> the mysql password for the openser mysql user. >> >> IIRC it must be configured somewhere in /etc/openser >> >> regards >> klaus >> >> >> Rajesh Kalagarla wrote: >>> Hello All, >>> >>> I am trying to setup the OpenSER on linux system and the distribution is > RHL. >>> How can we add multiple users credentials as part of configuration file? > i am not able to authorize more than one User Agent thru the config file. >>> when i am adding the second user info, it is simply overwriting the old > info and not authorizing the first user. >>> >>> modparam("auth_db","user_column" "1122") >>> modparam("auth_db", "password_column", "pluto") >>> >>> modparam("auth_db","user_column" "3344") >>> modparam("auth_db", "password_column", "earth") >>> >>> with this info, i am able to register only the second user.if i comment > the second user info, first user was able to register. please suggest how to > handle this, >>> when i am trying to add the user using the "openserctl" i am getting > permission denied error. >>> command: openserctl add 1122 earth rajeshk(a)idnltd.com >>> >>> output is, ERROR 1045(28000): Access denied for user openser@localhost > (using PASSWORD: YES) >>> >>> any help greatly appreciated, >>> >>> Thanks, >>> Rajesh >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Devel mailing list >>> Devel(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/devel >> >> -- >> Klaus Darilion >> nic.at >> >> > > -- Klaus Darilion nic.at

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[Serusers] ser 0.96 # mediaproxy # one-way audio SOLVED!!!
by Gareth Owen 08 Jan '07

08 Jan '07

Hi Davor I notice you had a problem with one way audio. I have the same problem but when the X-Lite client is behind firewall. Do you have any ideas or direction I could follow? Best regards Gareth Gareth Owen The information contained in this email message is intended only for the use of the individual(s) or entity to whom it is addressed, and may contain information that is confidential and may be legally privileged and exempt from disclosure under applicable laws. If the reader of this message is not the intended recipient or the employee or agent responsible to deliver it to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error please immediately notify the sender and destroy the original message from your system. The recipient should check this email and any attachments for the presence of viruses. BeVoice accepts no liability for any damage caused by any virus transmitted by this email

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[Serusers] Anyone like to finish my SER install for a fee?
by Bart Fisher 07 Jan '07

07 Jan '07

Hello all, I have SER 0.9.6 installed, along with Mediaproxy 1.8.0, Nathelper, SEMS 0.9.6, SerWeb 0.9.6 set up as a PSTN gateway on CentOS. However, there are some issues with NAT, Accounting (ACC) and the way I setup the web access. Also, I'd like to get to Voice mail and Conference working. Basically, I'd want you to take what's there and finish the install so everything works together. For an expert, all this is probably a few hours job. There is already SSH access in to the system. I started with a "LiveCD - VOIP" version of SER (http://www.wifi.com.ar/english/voip.html) with all the above (almost) but found it grossly out-dated and the Author unwilling to assist nor update to current releases. FYI: If you are are thinking of purchasing this software, save you money! The only thing nice is there is a prepaid billing script that might be converted to current releases. I just need to 'jump-start' this project to some usable level. I'm willing to pay for your time. If you are interested, please contact me at bart (at) icpage (dot) com. TIA Bart

1 0

[Serusers] SER 0.9.6 and Media-Server Asterisk 1.4
by Stefan Tichy 07 Jan '07

07 Jan '07

Hi all, SER 0.9.6 is used as registrar/proxy and asterisk 1.4 as media server (voicemail, ...) The situation when phone A calls phone B and B is busy is handled by a failure route revert_uri(); rewritehostport("10.11.12.13:5070"); append_branch(); t_relay; This works fine, if for example phone A is some Snom 190. If another asterisk server is used instead of phone A it works if "pedantic=no" is set in the sip.conf of asterisk. If it is "pedantic=yes", then asterisk compares the tags in the To header fields. There is one tag from phone B and a second from the asterisk media server. Asterisk ignores every response not having the tag used by phone B. IMHO asterisk is wrong, but since I can't controll which asterisk servers are used as clients - is there a way to may asterisk happy? Would it be possible to remove tags from 1XX responses? -- Stefan Tichy <seruser(a)pi4tel.de>

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[Serusers] how many calls can SER support?
by zahra bahar 06 Jan '07

06 Jan '07

Hi. do you know how many calls can ser support? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Start SER with xlite
by zahra bahar 06 Jan '07

06 Jan '07

Hi. I installed SER as a proxy server and Xlite as a softphone.but xlite can't make a call. it sends :408 or 404 .is it any trouble with ser?or xlite?how can I find out ? what changes in ser.cfg are needed for having a pc to pc call with xlite? in ser.cfg: should fork=no or yes? is it necessary to use mysql module? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] proxy and redirect servers
by zahra bahar 06 Jan '07

06 Jan '07

Hi. Could any one say me when we use proxy server and when it is necessary to use redirect server? thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] TM : BYE retransmissions
by Jean-François SMIGIELSKI 05 Jan '07

05 Jan '07

Hello list! I observed, on an openser-1.1.0, that stateful relayings of BYE requests absorb retransmissions from the sender but do not retransmit it to the next hop in the Route header. Please, is it normal or is ti a misconfiguration from myself ? Thanks a lot! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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Re: [Serusers] aliases
by Nils Ohlmeier 05 Jan '07

05 Jan '07

As this might be interesting for other users I cc'ed the list again. Just to be sure: you know that their is no alias table any more, right? Alias or not is just a matter of the flag value in the uri table. In general there is no need any more for re-writting a request which was targeted towards an alias. Request towards an alias are handled exactly the same way as "normal" requests, because both will get the same UID value. If you used the alias table for re-distributing the requests outside of your domain, which is actually more a forwarding function then an alias, then you could have this function as one (out of many) possible alternative(s) with AVPs like this: (Note: non-tested pseudo code) lookup_user("$t.uid", "ruri"); ... if ($t.uid) { load_attrs("$tu", "$t.uid"); if ($t.forwarding_target) { attr2uri("$t.forwarding_target"); t_relay(); } } Greetings Nils On Friday 05 January 2007 17:27, you wrote: > Thanks for the reply, Nils! I suppose I am a more of a newbie than I first > appeared as your answer is a little over my head. What I am trying to do > with the new version of SER is what used to be done (in previous versions) > with > > If (lookup("aliases")){ > Route(1); > > To redirect users based on a few entries in the aliases table. I won't be > using the SER server as a registrar or to handle dynamic user locations or > even authenticated accounts -- it will be open to the public without > authentication to access SIP accessible numbers and users. Mostly what I > want to do is just to route calls that SER receives (like john(a)iptn.org, > 5555555555(a)iptn.org etc.) to the domain/uri that used to be accessed in the > aliases table (openser worked this way too). > > Sorry for not explaining better and thanks again for your help! > > Eliott > > > -----Original Message----- > From: Nils Ohlmeier [mailto:nils@iptel.org] > Sent: Friday, January 05, 2007 8:32 AM > To: serusers(a)lists.iptel.org > Cc: Eliott Spencer > Subject: Re: [Serusers] aliases > > Hello, > > On Friday 05 January 2007 04:27, Eliott Spencer wrote: > > I am coming over from OpenSER because I ran some tests and found SER to > > be much faster. There are some things that are new to me that I imagine > > are easy to do. I have a working config with the 12/22/06 development > > snapshot > > > of 0.10 but I would like to lookup aliases to redirect inbound calls to > > another domain. I filled in the domain table and the uri table with a > > user > > > and I looked through the sample config but can't seem to get it to work. > > I > > > have also loaded the domain module, then the uri module, then the avp and > > avpops modules (among others). > > > > > > > > Any idea what I might need to put in my route? This doesn't seem to work > > (stolen from ser.cfg.sample): > > > > > > > > if (lookup_user("$t.uid", "@ruri")) { > > > > setflag(1); > > > > route(1); > > > > break; > > > > }; > > from looking at your code snipset I'm not sure what you really want to > achieve. > In general you should just do the lookup_user and then examine the AVPs > afterwards. I think that it is easier then looking at the result code of > the > > lookup_user function. > If you really want to distingiush if the call was targeted to an alias or > the > non-alias URI of an user then check for the AVP ruri_canonical. If it is > set > > (to the value 1) the URI contained the canonical (=non-alias) URI of the > user. > If you just want to know if the URI is a know one and belongs to a user, > just > check for the presence of the uid AVP after the lookup_user call. If it is > present the user is know to the system under the value of the uid AVP. > Otherwise the user/URI is not known to the system. > > Hope this helps > Nils

1 0

[Users] forking call to another user on same proxy
by Christian Schlatter 05 Jan '07

05 Jan '07

I try to fork a call to another user registered at the same proxy as the caller. I figured out that the easiest method would be to use the append_branch() function, something like route[0] { if (method=="REGISTER") { save("location"); exit; } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } # if someone calls test2@domain, fork call to test4@domain # (test2 and test4 are both registered at domain) if (avp_check("$rU", "eq/test2")) { append_branch("sip:test4@127.0.0.1"); } if (!t_relay()) { sl_reply_error(); }; exit; } I know that this could also be achieved by adding permanent location records, but I'd like to load the forking targets from a DB in the future. So my first question would be if using append_branch("sip:test4@127.0.0.1"); is the correct way to do parallel forking to a local user. I'm asking because this method does not work for me as expected. It seems that the two UAC state machines running on the same proxy host that are involved in a forking call do somehow not work well together. Forking the INVITE to both test2 and test4 works well, but when I pick up test2 the call gets terminated after a few seconds. Further investigations have revealed that canceling the second call leg does not work correctly. There is a message flow diagram available at http://www.unc.edu/~cschlatt/openser.html . For some reason, the proxy process sending from 127.0.0.1 is retransmitting the 487 "Request Terminated" although it receives ACKs from the upstream proxy. And this causes the upstream proxy to send a 487 back to the caller (time 15.360), which then terminates the call established between caller and test2. And this only happens if I fork to the same proxy, if I use two proxy hosts everything works fine (??). Any advice is highly appreciated, Christian

3 3

[Users] integrating with Asterisk and OpenSER for Voicemail
by raviprakash sunkara 05 Jan '07

05 Jan '07

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24) modparam("tm","wt_timer",1) #mrodparam("tm", "ruri_matching", 0) #modparam("tm", "via1_matching", 0) modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$avp(inv)"); setflag(2); }; .............. .... route(1); } route[1] { if(isflagset(2)) { t_on_failure("1"); }; } failure_route[1] { log("----------------------------------------- \n"); if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport("192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060"); resetflag(2); } ......................... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

3 2

[Users] openser+freeradius:(radius_is_user_in(): Failure)+(DEBUG:avpops:ops_c)
by buptzhaoyuan＠sina.com 05 Jan '07

05 Jan '07

Hi, Lists,happy new year! Probably a stupid question,but I can not solved it correctly.The ua can not register the openser normally.I think the 5 step runs well and the last is down.The debug shows two errors. The first error is " 0(5898) radius_is_user_in(): Failure" ;the other is " 0(5898) DEBUG:avpops:ops_check_avp: no src avp found".So please give me help and some suggestion. And the fellowing is my configure file and debug infomation.Thanks!! HTTP/SIP RADIUS +-----+ (1) +-----+ +-----+ | |==========>| | | | | | (2) | | | | | |<==========| | | | | | (3) | | | | | |==========>| | | | | A | | B | (4) | C | | | | |---------->| | | | | | (5) | | | | | |<----------| | | | (6) | | | | | |<==========| | | | +-----+ +-----+ +-----+ ====> HTTP/SIP ----> RADIUS I setup the system by using the document(http://www.openser.org/docs/openser-radius-1.0.x.html) and my openser is openser-1.1.0-tls, the file openser.cfg is the same as the document. # #$Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.168.3 alias="swifton.org" alias="192.168.168.3" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- [root@localhost openser]# cat openser.cfg # #$Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.168.3 alias="swifton.org" alias="192.168.168.3" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # check if user is suspended if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) { if (radius_is_user_in("From", "suspended")) { sl_send_reply("403", "Forbidden - suspended"); exit; }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (!uri==myself) { # check if user is allowed to do voip calls to other domains if(is_method("INVITE|MESSAGE")) { if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden VoIP"); exit; }; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("swifton.org")) { www_challenge("swifton.org", "0"); exit; }; # check the src ip address if(!avp_check("$avp(i:2)", "eq/$src_ip/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "pstn")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # set gateway address rewritehostport("10.10.10.10:5090"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("$avp(i:3)", "eq/i:1")) { # print time in an avp avp_printf("$avp(i:100)", "$Tf"); # extract day avp_subst("$avp(i:100)/$avp(i:101)", "/(.{3}) .+/*\1*/"); if(!avp_check("$avp(i:6)", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("$avp(i:100)/$avp(i:102)", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("$avp(i:4)") && avp_check("$avp(i:4)", "gt/$time")) || (is_avp_set("$avp(i:5)") && avp_check("$avp(i:5)", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; ...... route(1); } ...... exit; } # and the openser start as follows: 898) SIP Request: 898) method: <REGISTER> 898) uri: <sip:swifton.org> 898) version: <SIP/2.0> 898) parse_headers: flags=2 898) end of header reached, state=5 898) parse_headers: Via found, flags=2 898) parse_headers: this is the first via 898) After parse_msg... 898) preparing to run routing scripts... 898) parse_headers: flags=100 898) DEBUG:maxfwd:is_maxfwd_present: value = 70 898) parse_headers: flags=10 898) DEBUG: add_param: tag=f59c388b5de348778e76ebda993c2d8e 898) DEBUG: add_param: epid=b6757bf3fc 898) DEBUG:parse_to:end of header reached, state=29 898) DBUG:parse_to: display={}, ruri={sip:zhaoy@swifton.org} 898) radius_is_user_in(): Failure 898) parse_headers: flags=200 898) DEBUG:parse_to:end of header reached, state=10 ...... 0(5898) grep_sock_info - checking if host==us: 11==13 && [swifton.org] == [192.168.168.3] 0(5898) grep_sock_info - checking if port 5060 matches port 5060 0(5898) grep_sock_info - checking if host==us: 11==13 && [swifton.org] == [192.168.168.3] 0(5898) grep_sock_info - checking if port 5060 matches port 5060 0(5898) check_nonce(): comparing [459bb612ad569015987d4849d1ae603515f32e6b] and [459bb612ad569015987d4849d1ae603515f32e6b] 0(5898) DEBUG:auth_radius:radius_authorize_sterman: Success 0(5898) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(5898) DEBUG:auth_radius:extract_avp: string is <Sip-Group:voip> 0(5898) DEBUG:auth_radius:extract_avp: AVP name is <Sip-Group> 0(5898) DEBUG:auth_radius:extract_avp: AVP val is <voip> 0(5898) DEBUG:auth_radius:generate_avps: AVP 'Sip-Group'/0='voip'/0 has been added 0(5898) DEBUG:avpops:ops_check_avp: no src avp found 0(5898) parse_headers: flags=ffffffffffffffff 0(5898) check_via_address(192.168.168.30, 192.168.168.30, 0) 0(5898) DEBUG:destroy_avp_list: destroying list 0x422bc440 0(5898) receive_msg: cleaning up the freeradius debug infomation as follows: er-Name = "zhaoy(a)swifton.org" Digest-Attributes = 0x0a077a68616f79 Digest-Attributes = 0x010d73776966746f6e2e6f7267 Digest-Attributes = 0x022a34353962623631326164353639303135393837643438343964316165363033353135663332653662 Digest-Attributes = 0x04117369703a73776966746f6e2e6f7267 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "2ead4f5b9d78013a35f2eb7a792971a5" Service-Type = Sip-Session Sip-Uri-User = "zhaoy" NAS-Port = 5060 NAS-IP-Address = 192.168.168.3 Processing the authorize section of radiusd.confmodcall: entering group authorize for request 167 modcall[authorize]: module "preprocess" returns ok for request 167 modcall[authorize]: module "chap" returns noop for request 167 modcall[authorize]: module "mschap" returns noop for request 167 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "zhaoy" Digest-Realm = "swifton.org" Digest-Nonce = "459bb612ad569015987d4849d1ae603515f32e6b" Digest-URI = "sip:swifton.org" Digest-Method = "REGISTER"rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 167 ....... Processing the authenticate section of radiusd.confmodcall: entering group authenticate for request 167A1 = zhaoy:swifton.org:zhaoyA2 = REGISTER:sip:swifton.orgH(A1) = eff77a9105c5c0973492694f59c944f2H(A2) = d3add39dad2709bea55016ba79eb675aKD = eff77a9105c5c0973492694f59c944f2:459bb612ad569015987d4849d1ae603515f32e6b:d3add39dad2709bea55016ba79eb675aEXPECTED 2ead4f5b9d78013a35f2eb7a792971a5RECEIVED 2ead4f5b9d78013a35f2eb7a792971a5 modcall[authenticate]: module "digest" returns ok for request 167modcall: group authenticate returns ok for request 167radius_xlat: 'Authenticated'Login OK: [zhaoy(a)swifton.org] (from client openser port 5060)Sending Access-Accept of id 13 to 192.168.168.3:33101 SIP-AVP = "Sip-Group:voip" Reply-Message = "Authenticated" Sip-Group = "voip"Finished request 167 and my MSN:buptzhaoyuan@sina.com .I am nice to communicate with you! ------------------------------------------------------------------- 惠普商用台式机dc5750,高性能，低功耗！( http://ad4.sina.com.cn/sina/limeng3/mail_zhuiyu/2006/mail_zhuiyu_20061225.h… ) =================================================================== 注册新浪2G免费邮箱（ http://mail.sina.com.cn/chooseMode.html ）

2 1

[Users] Reg:Instance messages
by venkid 05 Jan '07

05 Jan '07

hai all, i am using X-lite as sip client .i have installed in window and am able to send instance messages and presence info is also fine with openser In redhat Linux i am trying to send instance messages using X-lite , but i am not able send and not able to receive any messages . is X-lite linux version is support for redhat linux Es 3.0 or any other sip client is there for redhat linux Es 3.0 Thanks and Regards venkatesh.d

2 1

[Serusers] aliases
by Eliott Spencer 05 Jan '07

05 Jan '07

I am coming over from OpenSER because I ran some tests and found SER to be much faster. There are some things that are new to me that I imagine are easy to do. I have a working config with the 12/22/06 development snapshot of 0.10 but I would like to lookup aliases to redirect inbound calls to another domain. I filled in the domain table and the uri table with a user and I looked through the sample config but can't seem to get it to work. I have also loaded the domain module, then the uri module, then the avp and avpops modules (among others). Any idea what I might need to put in my route? This doesn't seem to work (stolen from ser.cfg.sample): if (lookup_user("$t.uid", "@ruri")) { setflag(1); route(1); break; }; Thanks!

2 1

[Serusers] PA module segfault
by TERRY ROBERT 05 Jan '07

05 Jan '07

Hi, I'm getting a segfault error while running the latest presence server snapshot (ser-0.10.99-dev35-pa-4.2). The scenario is: UA SER SUBSCRIBE ----------> 200 <---------- NOTIFY <---------- 200 ----------> Everything is fine until the subscription expires, at which point the PA module segfaults and generates a core file. Any suggestions on this? (logfile and backtrace from gdb are attached).

2 1

Re: [Users] LCR db_only mode problem
by Jayesh Nambiar 05 Jan '07

05 Jan '07

Hi Juha, Thanks for your reply. First prominent doubt that I have is; when I am using LCR in db_only mode why does openser try to load the table while starting up? The scenario is as follows: I had 50 rows in LCR table and started openser and it started properly and also routed the calls properly according to the values in lcr and gw table. Now I just appended more 40,000 rows to this table and again the routing worked properly as per my lcr table. This behaviour was correct as I was using query only mode. I did not need any restart or any fifo reload command. Only when I restarted my openser, the openser again died as it was trying load the lcr table while starting. Could this be a bug in lcr module as I am not sure whats happening. I tried increasing MAX_NO_OF_LCRS but even that did not help. Somebody please help if poosible. Thanks, w/regards, jayesh ----- Original Message ---- From: Juha Heinanen <jh(a)tutpro.com> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Tuesday, 26 December, 2006 10:58:01 PM Subject: [Users] LCR db_only mode problem X-VM-v5-Data: ([nil nil nil nil nil nil nil nil nil] the error messages you get are coming from this statement: if (lcr_dbf.query(dbh, NULL, 0, NULL, lcr_cols, 0, 4, 0, &res) < 0) { LOG(L_ERR, "lcr_reload_gws(): Failed to query lcr data\n"); lcr_dbf.close(dbh); return -1; } i don't know which kind of memory db queries reserve. if shared memory, you can start openser with larger -m parameter value. if pkg memory, you need to edit PKG_MEM_POOL_SIZE in config.h and recompile openser. with lots of lcr entries and no caching, performance may be bad. if you use caching then you need to increase value of #define MAX_NO_OF_LCRS 256 in lcr_mod.c and recompile lcr module. increasing MAX_NO_OF_LCRS means that more shared memory will be used so you may also need to start openser with larger -m parameter value. -- juha Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php

3 3

[Serusers] Aastra 9133i
by Bill 04 Jan '07

04 Jan '07

Does anyone have a config for an Aastra 9133i IP phone to a SER box? I am trying to configure one. I can make calls to the Aastra but can't make calls from the Aastra. It registers properly. Thanks Bill

1 0

[Users] new CDRTool release 5.0.5
by Adrian Georgescu 04 Jan '07

04 Jan '07

Hello, I have been asked many times where is the best place to obtain support and ask questions for CDRTool. As CDTool is at this moment primarily designed and tested for OpenSER deployments I have decided that OpenSER users mailing is the best place to make announcements for new releases. There is of course no guaranteed support, as you may expect for a free product, support requests will however not go unheard. Unless there is no objection or complaint, I will keep this mailing list updated about CDRTool new releases from now on. Here it is: CDRTool 5.0.5 has been released. The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog: * Fixed determination of previous year in update_raddact_record_mediaproxy Consolidate SET statements to minimize binary log entries. Fixed bug in update_raddact_record_mediaproxy, the record values were reset after execution of update on the first table causing failure to update the previous table. setup/radius/OpenSER/radius_accounting.proc must be reloaded into the MySQL server: mysql -u root -p -h sipdb radius < setup/radius/OpenSER/ radius_accounting.proc * Use same CDR structure for MaxSessionTime and DebitBalance * Change logging for prepaid actions to fit one line * Normalize tries previous month if no record has been updated * Mention how to see prepaid account status * Updated rating docs * Moved E164 class to phplib/local.inc * Show accounts with quota that exceeded a certain treshold scripts/SER/quotaShowAccounts.php treshhold * Added documentation for the Quota system (doc/QuotaSystem.txt) * Use normalization lock per table using GET_LOCK() server function http://dev.mysql.com/doc/refman/5.0/en/miscellaneous-functions.html This provided faster web response when multiple clients access the interface * Fixed confirmation for delete operations in rating tables when global vars are turned off in php.ini Regards, Adrian

2 1

[Users] Strange OK Message after Register
by Martin Klisch 04 Jan '07

04 Jan '07

Hi, yesterday i've began to upgrade openser from 1.0.1 to 1.1.0. Now i got a strange situation when registering with two different clients with the same sip account. in the OK Message from the sip proxy i have a strange contact field: Contact: <sip:012345@xxx:5070>;q=0.9;expires=3600, <sip:012345@xxxx:63036>;expires=25 i think in the OK Message should only be one contact adress?

2 1

[Users] Forward to the "To" field
by Thomas Deillon 04 Jan '07

04 Jan '07

Hi all, Merry Xmas ! I'm just a small question if some people are still working :-) I try to make a load balancing solution with SER and Asterisk. I use the dispatcher module and I set on asterisks servers the default gw as SER. All requests go through the SER to the asterisks and on the SER I'm have a line like If source-ip = Asterisk source IP do ... And I want to do something like forward to the TO FIELD ... Do you know how can I do this ? Thanks a lot for your help, Thomas

3 2

[Users] Secure SIP messages
by Steffen Witt 04 Jan '07

04 Jan '07

Hello, I have a few questions about how to secure SIP messages... IPSec or TLS can be used to guarantee privacy, integrity and authenticity of transmitted data. But SIP is only payload/data for these protocols. Digest Authentication does only guarantee authenticity... S/MIME seems to be interesting but I don't know any implementation in a SIP client. My questions: - Is there any mechanism that prevents me from corrupting or faking SIP messages? - Is it possible to create a kind of binding between the certificates used for TLS/IPSec and the SIP accounts? Thanks for your answers... Best regards, Steffen

2 3

[Serusers] HELP!!! Table acc is full
by Nhadie Ramos 04 Jan '07

04 Jan '07

Hi All, How can i solve this problem? is this a mysql problem? anyone has encountered this error before? Regards, Ronald

1 0

[Users] European and Australian based SIP termination providers
by T.R. Missner 04 Jan '07

04 Jan '07

Hello, Does anyone have recommendations for a SIP based termination provider in Europe and/or Australia? I am looking specifically for a wholesale, carrier grade provider. I have found several which appear to be asterisk based and quite small. What I am looking for is more of a traditional carrier that is SIP enabled as opposed to a smallish mom and pop shop. Thanks in advance. T.R.

3 2

[Serusers] Ser, MediaProxy and Asterisk
by Gerardo Amaya 04 Jan '07

04 Jan '07

Hello all, I'm trying to use a SER server for the following: | | | | ISP #1 ISP #2 Private Link Private LINK NATed SIP SIP Clients 172.18.xxx.yyy 172.16.5.xx Clients | | | | | | | | | | | ---------------------------------------------------------------------------------------------------- | ---------------------------------------------------------------- SER with MEDIAPROXY ---------------------------------------------------------------- | | | ================ Asterisk (172.16.5.1) ================= | | PSTN I need asterisk to handle all the calls to the PSTN and event the SIP calls between them. I need to use the SER server as a tool to receive calls from any of the environments that I draw on top, and all communicating with Asterisk, even I also have a backup Asterisk box, that I can use in load balance with SER. I saw that there is a function "rewritehostport()" which I'm pretty sure it won't work since I will not keep the call in the SER server. I have made the NAT configuration of the Getting Started Guide, but I now that this is just part of what I need to build the complete solution. Can anyone please give me a hint in what is that I need to acomplish such task Thanks in advance Gerardo Amaya

2 1

[Serusers] strange problem ..... /tmp/mysql.sock and /var/lib/mysql/mysql.sock
by Tako Chang 04 Jan '07

04 Jan '07

Hi, all I install mysql server and SER in my FC6 when I run SER with mysql to do authenticate #ser -f auth-mysql.cfg (the SER getting started example) it prompt me that can't connect to local MySQL server through socket 'tmp/mysql.sock' so I modified the /etc/my.cnf' file to change socket=/tmp/mysql.sock but when i start SER and use serctl to add a user ERROR 2002 (HY000): Can't connect to local MySQL server through socke 'var/lib/mysql/mysql.sock' was prompted I don't know if I could open two sock at the same time or how can I let SER or serctr chage the path to fit another. Did someone can show me some clue or solution? thanks!!

2 1

[Serusers] One way audio
by Gareth Owen 04 Jan '07

04 Jan '07

Can someone help? I am fairly new to SER and Mediaproxy and can't find a detailed explanation of how nathelper etc work, so I'm struggling! My config is:- SER 0.9.6 Mediaproxy 1.7.2 Fedora Core 4 86x64 SER and Mediaproxy run on the same server which acts as a gateway to a remote Sonus PSTN gateway over a VPN. I also use Asterisk on my local network. The SER server acts as a router between Asterisk servers and the PSTN via the Sonus. This works fine but my problem is when I attempt to use X-Lite 3.0 UA behind NAT on a remote network. Asterisk is not used in this config but proves that two way audio is fine using SER and the Sonus gateway. Problem: X-Lite behind NAT on remote network to Sonus PSTN gateway via SER and Mediaproxy, I get only one ay audio. The audio is fine from X-Lite to PSTN but not in the reverse direction. Thanks for your help

2 1

[Users] Issues with calls using openser.
by Shane Burrell 04 Jan '07

04 Jan '07

I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the far end is left off-hook. This worked before but I did modify my script to work with mediaproxy. Below is the wireshark decode of the sip messagining. Any help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway. |Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 | |22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@152.53.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060 | |(5060) ------------------> (5060) | | |22.031 | 100 Giving a try | |SIP Status | |(5060) <------------------ (5060) | | |22.031 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |22.040 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |22.042 | | 180 Ringing |SIP Status | | |(5060) <------------------ (5060) | |22.042 | 180 Ringing | |SIP Status | |(5060) <------------------ (5060) | | |25.244 | | 200 OK SDP ( telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |25.245 | 200 OK SDP ( telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |25.269 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |25.269 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060 | |(5060) ------------------> (5060) | | |25.270 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | |25.291 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |25.291 | BYE | | |SIP Request | |(5060) ------------------> (5060) | | |25.293 | | BYE | |SIP Request | | |(5060) ------------------> (5060) | |25.326 | | 200 OK | |SIP Status | | |(5060) <------------------ (5060) | |25.327 | 200 OK | | |SIP Status | |(5060) <------------------ (5060) | | Shane

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[Users] How relay/retransmit subscribe, notify or publish messages...
by Marvin Ottley 03 Jan '07

03 Jan '07

Good day all, Currently I'm working on an interface/design where I have openser relaying messages to asterisk servers. The stumbling block I've run into is having openser relay SUBSCRIBE, NOTIFY or PUBLISH messages to Asterisk. So far, I've tried using openser's forward() and t_relay() functions without success; By the way, these work just fine with INVITE and REGISGER messages. Any pointers would be greatly appreciated Thanks, Marvin Ottley Email: marvino(a)nbsvoice.com <mailto:marvino@nbsvoice.com> This email may contain confidential and/or privileged information, and is intended solely for the addressee identified above. Any unauthorized use, disclosure or dissemination of the information contained in this email is strictly prohibited. If you have received this email in error, then please notify the sender immediately, and delete this email and all copies.

2 1

[Users] several presence question
by Klaus Darilion 03 Jan '07

03 Jan '07

Hi! I read the docs of the presence module and have some question. Maybe someone can enlighten me ;-) 1. If I understand correctly, the pua module acts as a "UAC". Thus, it can be used to send PUBLISH or SUBSCRIBE. I wonder what is the usecase for sending SUBSCRIBE? Usually, a client which sends a SUBSCRIBE receives subsequent NOTIFY messages. How can these NOTIFY messages be handled? 2. If an external application sends PUBLISH via the uac_mi interface, how can be distinguished between a "new" PUBLISH or an "update" PUBLISH? For example if the MI interface returns the E-Tag from the 200 Ok response this could be used in the next request to group identify PUBLISH reqeusts which belong together. 3. There is a bug in the pua module docs: the db_url(str) parameter is described twice. 4. The following module documentations lack of default values for the module parameters: pua, pua_usrloc, presence (maybe some example show the default value, but IMO the default value should be explicitely stated) 5. How can I update the presence status from a third party application? Is it possible to write directly into the tables of the presence modules? Or can I fake an update using the pua_mi to loop a PUBLISH to itself for handling it afterwards with handle_publish()? May this cause problems because it interferes the subsequent PUBLISH request from the client (etag)? regards Klaus

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RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 03 Jan '07

03 Jan '07

Hi, In fact, now, every phone does not response. The options didn't go thought the FW, so, the phone cannot answer. I don't understand why the traffic don't go thought it !! I have test this configuration with two firewall: a sonicwall and a soho. Cheers, Thomas -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Envoyé : mercredi, 3. janvier 2007 14:52 À : Thomas Deillon Cc : users(a)openser.org Objet : Re: [Users] SER as proxy in front of Asterisk Hello, what is the type of NAT/Firewall you have in front of Thomson? As I can see from the diagram, the phone does not reply to OPTIONS request, and I guess, the nat/firewall needs outward traffic to keep the pinhole open. Check why thomson ignores the options requests. Cheers, Daniel On 01/03/07 15:17, Thomas Deillon wrote: > > Hi, > > I have made more tries and now, all my phones don't work ... > > The firewall opens a connection to REGISTER phone but after a while > the OPTION message from the Asterisk will be DROP by the FW ... > > You can find a diagram of the exchange here: > http://deillon.eu/tmp/exchange_sip.png > > A list of all exchange: http://deillon.eu/tmp/ip_sip.txt > > And all sip message: http://deillon.eu/tmp/message.txt > > Thanks a lot for your help, > > Thomas > > ------------------------------------------------------------------------ > > *De :* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *De la part de* Thomas Deillon > *Envoyé :* mardi, 2. janvier 2007 11:44 > *À :* users(a)openser.org > *Objet :* [Users] SER as proxy in front of Asterisk > > Hi all, > > I wish you first a happy new year !! > One again, I ask you some help. Thanks a lot for your patience and > your answers that really helped me. > > So, I want to put a SER in front of one Asterisk for the moment (more > after). > All Servers have public IP address and so, I don't care about the > RTP/SDP messages. > All phones are behind NAT somewhere in Switerland :) > > > I have the configuration below in openser: > > > modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") > #modparam("dispatcher", "force_dst", 1) > modparam("dispatcher", "flags", 2) > > modparam("usrloc", "db_mode", 0) > #modparam("rr", "enable_full_lr", 1) > > route{ > > > xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > # if (search("User-Agent:.*Thomson.*")) { > # }; > > > if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { > if(method=="OPTIONS") { > }; > avp_pushto("$ru","$tu"); > forward(); > > }else { > fix_nated_contact(); > force_rport(); > if(method=="REGISTER"){ > ds_select_dst("4", "0"); > t_relay(); > } > else { > ds_select_dst("0", "4"); > forward(); > } > } > } > > > And in my dispatcher.list I have: > > 0 sip:212.xxx.xxx.153:5060 > 4 sip:212.xxx.xxx.153:5060 > > > > With this configuration, a snom phone, a X-lite phone and a Cisco > phone seams to work .... > But with the Thomson ST2030, it doesn't work. > In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION > message to the FW IP adress and the right port where the thomson is > and then, after a while, the FW close the connection. The thomson > phone is so "UNREACHABLE" on asterisk status and nobody can call it. > I'm not sure that my configuration is ok in ser but I think that is a > problem with the Thomson2030. Do you had the same kind of problem or > do you understand the problem here ? > > > Thanks for your help, > > Thomas Deillon > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 03 Jan '07

03 Jan '07

Hi, I have made more tries and now, all my phones don't work ... The firewall opens a connection to REGISTER phone but after a while the OPTION message from the Asterisk will be DROP by the FW ... You can find a diagram of the exchange here: http://deillon.eu/tmp/exchange_sip.png A list of all exchange: http://deillon.eu/tmp/ip_sip.txt And all sip message: http://deillon.eu/tmp/message.txt Thanks a lot for your help, Thomas ________________________________ De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Thomas Deillon Envoyé : mardi, 2. janvier 2007 11:44 À : users(a)openser.org Objet : [Users] SER as proxy in front of Asterisk Hi all, I wish you first a happy new year !! One again, I ask you some help. Thanks a lot for your patience and your answers that really helped me. So, I want to put a SER in front of one Asterisk for the moment (more after). All Servers have public IP address and so, I don't care about the RTP/SDP messages. All phones are behind NAT somewhere in Switerland :) I have the configuration below in openser: modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") #modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 2) modparam("usrloc", "db_mode", 0) #modparam("rr", "enable_full_lr", 1) route{ xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # if (search("User-Agent:.*Thomson.*")) { # }; if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { if(method=="OPTIONS") { }; avp_pushto("$ru","$tu"); forward(); }else { fix_nated_contact(); force_rport(); if(method=="REGISTER"){ ds_select_dst("4", "0"); t_relay(); } else { ds_select_dst("0", "4"); forward(); } } } And in my dispatcher.list I have: 0 sip:212.xxx.xxx.153:5060 4 sip:212.xxx.xxx.153:5060 With this configuration, a snom phone, a X-lite phone and a Cisco phone seams to work .... But with the Thomson ST2030, it doesn't work. In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION message to the FW IP adress and the right port where the thomson is and then, after a while, the FW close the connection. The thomson phone is so "UNREACHABLE" on asterisk status and nobody can call it. I'm not sure that my configuration is ok in ser but I think that is a problem with the Thomson2030. Do you had the same kind of problem or do you understand the problem here ? Thanks for your help, Thomas Deillon

2 1

[Users] 1.1.0 Compiling for Solaris 10
by Martin Klisch 03 Jan '07

03 Jan '07

Hi, i can not compile openser 1.1.0 for solaris 10. any idea why? gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.3"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -c cfg.tab.c -o cfg.tab.o /usr/ccs/bin/yaccpar:5: warning: ignoring #pragma ident cfg.y: In function `yyparse': cfg.y:859: error: `IPTOS_MINCOST' undeclared (first use in this function) cfg.y:859: error: (Each undeclared identifier is reported only once cfg.y:859: error: for each function it appears in.) /usr/ccs/bin/yaccpar:164: warning: label `yynewstate' defined but not used /usr/ccs/bin/yaccpar:374: warning: label `yyerrlab' defined but not used gmake: *** [cfg.tab.o] Error 1

1 1

[Users] Problem registering the UA with openSER(tls enabled)
by Ncheeku Baranov 03 Jan '07

03 Jan '07

Hi, I am trying to make my non-TLS/TLS UA register with my TLS enabled openSER. Currently I am just working on my local machine with the client UAs on the same subnet,(so there is only one domain, but its not named). Below is my configuration file: disable_tls = 0 listen = tls:10.30.100.41:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user- cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user- privkey.pem" tls_ca_list = "usr/local/etc/openser/tls/user/user-calist.pem" However, with the above configuration the client UAs couldnot register and I got 408 Request Time out Message. Is there any field that is missing to make this simple scenario work? What should be the values of "tls_client_domain" and "tls_server_domain" fields in this case? I noticed that when I start the openSER without TLS support using "openserctl start" and do "ps -e" after that, there are more openSER processes running than if I start openSER with TLS support in which case I see very few of these processes running. Your help is much appreciated.... Best regards, NCheeku

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[Users] 487 Transaction Terminated
by KOK FOON YEW 03 Jan '07

03 Jan '07

This is pertaining to call cancel operation where the calling UA sends a CANCEL request via openser to the called UA. After the UA sends ACK in response to 487 Transaction Terminated response, 487 responses are re-transmitted by openser to the UA as if the SIP proxy didn't regcognise the ACK request. I have tested with other UA and compare the differences between the ACK sent by UA that worked and UA which has this problem. The difference is that the ACK sent by the UA having this problem has a route header added. Foon-Yew Kok [This e-mail is confidential and may be priviledged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---

2 1

[Users] Accounting using dialog and sst module
by Asterisk Expert 03 Jan '07

03 Jan '07

Hello, I am currently trying to do accurate billing with openser. I found that dialog and sst modules can be used to determine failed transactions. i.e. If UA crashed and no BYE received at all. I tried to configure dialog and sst modules as specified in the documentation. But active dialogs keep incrementing even after hangup. It would be nice if someone can post a good example. I am using cvs head of openser. Snippet of config is as below: ###################### DIALOG ###################### modparam("dialog", "dlg_flag", 13) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "use_tight_match", 1) modparam("dialog", "enable_stats", 1) ###################### SST ###################### modparam("sst", "enable_stats", 1) modparam("sst", "timeout_avp", "$avp(i:10)") modparam("sst", "sst_flag", 14) route { ........ ........ if(method=="INVITE") { setflag(13); # set the dialog flag setflag(14); # Set the sst flag avp_write("15","$avp(i:10)"); }; if (method=="INVITE") { record_route_preset("192.168.1.2:5060;nat=yes;did=1"); }; ......... ......... } -- Regards Asterisk Expert

3 6

[Users] Help: how can I create one SIP account
by CESAR HUAMANI H. 03 Jan '07

03 Jan '07

Hello, Today I installed the openser in my debian etch since the CVS, I was looking information about how I can creat accounts SIP for my Endpoints (softphone and Hardphone ATA), please help me I'm new with the OPENSER Best regards

2 1

[Serusers] Detecting if username is in subscriber table
by Simon 02 Jan '07

02 Jan '07

Dear All, I would like to detect if a URI is for a user that is in the subscriber table. If I just use lookup("location") then I only detect those users that have registered. If a phone is switched off it is not registered and so will not be in the location table. However it will be in the subscriber table and I need to be able to check this from within the ser.cfg file to allow correct routing etc. Any Ideas ? Many Thanks Simon

2 1

[Users] Call forward record and Remove from callid in Openser
by Howard Tang 02 Jan '07

02 Jan '07

Hi All, I hope someone can help me with the following questions I have. 1. I have set up accounting module for record the call, simple outgoing call and incoming call are ok. However, if a sip phone activated the call forward, the call record doesn't show which sip client forward the call. e.g. A call B and B is the sip phone which forward the call to C. In the record, I can only see A call C. How can I see a record that B call C instead of A call C in the record? 2. I have connected to a PSTN gateway, and i have some internal sip users which can make a pstn call, the sip proxy will send their own private number a caller id to the pstn gateway. Since the callid is a invalid number, many Mobile phone networks provider reject the call. I want to remove the from callid when the call being send to the pstn gateway, how can i do this? If someone can send me the code that will be great. -- Howard Tang

2 1

[Users] Presence and Location
by Klaus Darilion 30 Dec '06

30 Dec '06

Hi! What kind of presence does the presence module support? Only the "basic" presence format PIDF (RFC 3863) or also PIDF extensions like the GEOPRIV Location Object Format RFC 4119? Or in other words - is the presence module aware of the XML payload or is it agnostic to the presence payload? regards klaus

2 1

[Users] totag_avpid
by Klaus Darilion 30 Dec '06

30 Dec '06

Hi! I just found this parameter in sl and presence module, but the usage is not described well. What is it's purpose? I guess that the presence module exports the totag into this AVP and the sl module use this totag in its response. I think it is not a good idea to have it configureable in 2 modules. We had the same problem with the "use_domain" parameter in several modules which was fixed recently. Maybe this can be fixed the same way - having it configurable in sl module, and presence module look ups the value on loading (or generate a error if not defined). Further, the presence module's parameter documentation lack the default values. regards klaus

2 1

[Users] Inspiring New Year wishes from Sunkara Ravi
by Sunkara Ravi 30 Dec '06

30 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/GX91230040549763 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is GX91230040549763 If you need help in viewing your card or any other assistance, please visit our Help / FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

1 0

Re: Re: [Users] how to configure failure route for b2bua or GW
by liu york 30 Dec '06

30 Dec '06

yes,the failure route works,but the client phone still receive 408 messeage. I try to use if(t_check_status(408|483)) drop; at failure route ,but faild :( ======= 2006-12-29 16:28:57 您在来信中写道：======= >Do no use forward(), just rewritehost and afterwards t_on_failure and t_relay. > >Samuel. > >2006/12/29, liu york <york(a)goldentek.biz>: >> hi all >> >> I try to build failure route to another backup b2bua or pstn gw,it seems can work after timeout ,but unfortunate,the openser send a 408 messeage to phone before failure route.,config like that,have any general config to failover or loadbalance for me.Many many thx. >> >> route[4] { >> >> log(1, "b2bua"); >> rewritehost("192.168.1.204"); >---> #forward("192.168.1.204:80"); >> t_on_failure("2"); >> t_relay(); >> } >> >> failure_route[2] { >> log(1, "b2bua"); >> rewritehost("192.168.1.226"); >---> #forward("192.168.1.226:5060"); >> t_relay(); >> } >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2006-12-30

1 0

[Serusers] Re: [Serdev] SIP Stack development
by Greger V. Teigre 29 Dec '06

29 Dec '06

Hi Alan, See inline. Alan Basinger wrote: > Thanks Greger, > I know ho hard it is to keep up with the documentation of a rapidly > changing project and also know what a challenge it is to manage the > code. I hope we can use the base of SER to create some new and > exciting applications uing SIP and add some usefull features and > functionality to the base code for everyone to benifit from. > > I appreciate the quick reply and answering a newbs questions as I may > have more as we move along. :-) Sure, I realize that lack of documentation can be replaced (partially and only temporarily) by responsive support on mailing lists. It is also a way to establish the actual scope of documentation work. > > Hi Alan, > I'm heading the new documentation effort. An updated list of the > supported rfcs and standards is on my list of needed docs... I'm on > mobile right now, so I > cannot search for you, but on iptel.org/listsearch you can search for > Janak and supported rfcs. I think he posted an update recently. > > As for the specific draft, I'm not familiar with it. In general > though, media is not handled by ser, only messaging. > The draft is for allowing messaging as well as media to be sent to a > loopback to mesure actual network conditions to and from the device > without actually setting up a call. Very usefull and we have many > vendors implimeting it in there CPE. (Linksys, Polycom, SNOM, > Panasonic, etc.) I woul like to see about developing this and many > other features / functionaility into SER. I worked with one of the > authors of this draft and it realy makes sense. Here is a link to it. > > http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-05.txt > Aha, I recognize the name now :-) Without knowing the details of the draft, I have the following comments: * In general you want your servers out of the rtp path (i.e. direct media between UAs), in fact, ser is never in the rtp path. However ser is very good at doing message processing, forwarding and all sorts of "tricks" * RTP may be handled by three separate software components: rtpproxy, mediaproxy, and sems (mediaproxy is not an iptel.org project, the others are). SER modules (nathelper and mediaproxy) are used to communicate with these external components (locally or over udp) and for mangling the SDP payload * Your measurements will probably need to follow the media path of a normal session. If not, you will not be able to measure for example real latency on your ex. rtpproxy server(s). SEMS is a very good general-purpose media server, while rtpproxy/mediaproxy are more dedicated for proxying of rtp for NAT-purposes etc * Hence, you probably need to implement sdp loopback intp the rtp handling component. SEMS may be the easiest way due to its very modular and powerful plugin interface (it's also a B2BUA). However, as sems don't stay in the media path except in conferences etc, you don't really measure "real world..." * I suspect you can create a ser module (or extend nathelper) that implements the basic loopback stuff using rtpproxy without touching rtpproxy (too much) > As for development, ser has a very efficient core exposing a module > interfaces. Modules can implement functions, parameters, avpairs (i.e. > set variables), and > selects (give access to certain info related to a message). > I found a copy of the 2002 developers guide and have been starting > there but an concerned that it may be too outdated any idea when a new > draft will be available for review? The basic concepts are still relevant. It has not been updated to cover to the attribute value pairs and the selects. We are currently focused on the user documentation for the next release, but we are debating what the next steps are. A how-to for module development is already on the list, possible with a retouch of the old developers guide. You can start out with one of the simpler modules (like textops or dispatcher) and modify to your needs. Understand the plugin interface, lumps and shared memory, and you are pretty much there ;-) > > Contributions are always welcome. We have just introduced a new > classification of modules, thus allowing experimental modules into the > cvs without too much > fuss. > Good to hear as me and my partner in crime and I hope to be able to > add some value with our ideas. :-) g-) > > BTW, start with ser ottendorf. It is getting close to release and has > many new important improvements. > > Feel free to ask questions on this list. > g-) > > Thanks again > > Alan Basinger > > ------- Original message ------- > From: Alan Basinger <droidgeneral(a)yahoo.com> > Sent: 27.12.'06, 12:43 > > > Hello all, > > I definitely am a newby with SER but not with SIP and I am trying to > find out where the supported SIP methods are? > > Specifically how would I go about finding out if SER supports the > Media Loopback draft or other ratified or non ratified components. > > Also if not supported where would I being the coder that I am not be > able to modify the code myself to add these functions and then > distribute them back to > the community? > > > > Thanks in advance > > > > Alan > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com <http://mail.yahoo.com/> > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com

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[Users] New Year's Eve invitation from Sunkara Ravi
by Sunkara Ravi 29 Dec '06

29 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123Greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/HW31229040825964 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is HW31229040825964 If you need help in viewing your ecard or any other assistance, please visit our Help/ FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

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[Users] Cheerful New Year wishes from Sunkara Ravi
by Sunkara Ravi 29 Dec '06

29 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123Greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/HW41229040814251 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is HW41229040814251 If you need help in viewing your ecard or any other assistance, please visit our Help/ FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

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[Users] how to configure failure route for b2bua or GW
by liu york 29 Dec '06

29 Dec '06

hi all I try to build failure route to another backup b2bua or pstn gw,it seems can work after timeout ,but unfortunate,the openser send a 408 messeage to phone before failure route.,config like that,have any general config to failover or loadbalance for me.Many many thx. route[4] { log(1, "b2bua"); rewritehost("192.168.1.204"); forward("192.168.1.204:80"); t_on_failure("2"); t_relay(); } failure_route[2] { log(1, "b2bua"); rewritehost("192.168.1.226"); forward("192.168.1.226:5060"); t_relay(); }

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Re: Re: [Serusers] Creating particular mysql schema with the given db user name - bad db connection url
by "tzieleniewski" 28 Dec '06

28 Dec '06

worked but another problem apeared:) the script has created the ser database user and the database ser_0_9_7 for this user Ive changed the database connection urls to the following: fifo_db_url="mysql://ser:heslo@localhost/ser0_9_7" modparam("auth_db|uri_db|usrloc|domain", "db_url", "mysql://ser:heslo@localhost/ser0_9_7") But despite that ser try too connect to the "ser" database: stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) AVPops - initializing acc - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 1(0) INFO: fifo process starting: 3499 1(0) new_connection(): Access denied for user 'ser'@'localhost' to database 'ser' 1(0) db_init(): Could not create a connection 1(0) ERROR: acc_db_init: unable to connect to the database 1(0) init_mod_child(): Error while initializing module acc 1(0) ERROR: open_uac_fifo: init_child failed 1(0) starting fifo server failed 1(0) INFO: signal 15 received > On Thu, Dec 28, 2006 at 02:14:18PM +0100, TZieleniewski wrote: > > > Ok but I generalny dont want to install ser I just want to compile it > > and then have two different instances of ser on the same machine > > (different releases)<br> > > gen_ha1.c and gen_ha1.d are in utils/gen_ha1 directory , there is a > > makefile included will it work if I will invoke make in that directory?<br> > > yep, it will works. Just compile and change the path to gen_ha1 in > ser_mysql.sh. > > > > <br> > > Tomasz<br> > > </body> > > </html> > > -- > Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info > AD1-UANIC * ICQ: 122351182 * http://www.start4.info

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[Serusers] Radius Accounting Help
by Arun Kumar 28 Dec '06

28 Dec '06

Hi, I've configured my radius + mysql and openser. Now openser is sending accounting packet to radius server and I'm able to see accounting packets coming to my radius server but the problem is : Its sending the Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Ack packet twice and then it shows rlm_radutmp: Login entry for NAS august port 5060 duplicate and then two stop packet: Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Bye and then it shows rlm_sql (sql) in sql_accounting: stop packet with zero session length and after second stop packet it shows rlm_radutmp: Logout for NAS august port 5060, but no Login record please help. thanks arun

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[Users] lock error with configuration mistake
by Jean-François SMIGIELSKI 28 Dec '06

28 Dec '06

I wrote this message a few days ago: http://openser.org/pipermail/devel/2006-December/005103.html I am sorry for the lack of informations I gave, I could investigate a little bit about it. Here is the full context: When an INVITE comes, I dispatch and t_relay it to a server. If the transaction makes an error, I wrote in the configuration to manage this whith a failure route. In this failure route, I dispatch and t_relay the transaction to a server for an other service. And then appears an strange error : * with an openser compiled with the default flags, the UDP that received the last UDP packet infinitely loops on the "sched_yield" function, in the "tsl" function in fastlock.h. * with an openser compiled with flags that make him use pthread for locking purposes, the UDP process hangs, while trying to get a mutex. It appears I forgot a call to append_branch in the failure route. Maybe the process tried to get the same lock twice. Is it known for OpenSER 1.1.0? If it is, is it corrected in later stable releases ? Thanks a lot for your answer! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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[Serusers] Presence support: PUBLISH/SUBSCRIBE
by raj gopal 28 Dec '06

28 Dec '06

Hi, I have installed ser-0.10.99-dev35-pa-4.1 and following up presence handbook. i have configured XCAP using apache. i am getting following errors with ser.cfg of ser package. For PUBLISH request : 0(13411) get_pres_uri(): Error while extracting plain URI 0(13411) handle_publish(): Error while extracting presentity URI For SUBSCRIBE request: 0(13411) ERROR: rls_handler.c:484: XCAP query problems 0(13411) XCAP query failed 0(13411) ERROR: rls_handler.c:484: XCAP query problems 0(13411) XCAP query failed have any one succeed with presence? help me... *-Venkat *

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[Serusers] AVPops and Textops to chang username of request uri
by chungyu 28 Dec '06

28 Dec '06

Hi: I use module AVPops and Textops do chang username of request uri. example: john(a)10.10.16.23 I hope chang => 4762(a)10.10.10.22 my configuration sample: if (uri=~"john@") { avp_write("4762", "s:phone"); subst_user(/(.*)/$avp(phone)/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("s:phone"); }; but reply is "Call failed:Device not found" I try this sample is OK if (uri=~"john@") { avp_write("4762", "s:phone"); subst_user(/(.*)/4762/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("s:phone"); }; OS: Freebsd 5.4 SIP server: SER-0.9.6 please tell my problem is on? Thank you Chungyu This is my Configuration: # ----------- 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" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ 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" loadmodule "/usr/local/lib/ser/modules/avp.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" #loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("xx.xx.xx.xx", "subscriber")) { www_challenge("xx.xx.xx.xx", "0"); break; }; save("location"); break; }; if (uri=~"john@") { avp_write("4762","phone"); subst_user('/(.*)/$avp(phone)/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("phone"); }; 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(); }; }

2 1

[Users] MP-104 FXO issues
by Nick De Cristofaro 27 Dec '06

27 Dec '06

Can't make outbound calls, though not related directly to OpenSER.. Maybe someone here knows how to configure these guys. I have inbound calls being automatically dialed into a * extension but If I make any outbound calls I get a syslog output: 16:45:21.609 : 192.168.1.211 : WARNING : ( lgr_call)(458 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number xxxxxx2018 16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_psbrdif)(457 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:xxxxxxx2018 Source:xxxxxxx11526 SourceIp:c0a80188 16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_TrnkGrp)(456 ) !! [ERROR] #0:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number xxxxxx2018 16:45:21.593 : 192.168.1.211 : NOTICE : ( lgr_stk_ses)(455 ) SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0 Any help would be great. Also, caller ID doesn't work - only shows the caller ID of whatever is set at the endpoint phone number settings and not what is actually coming on the line :( Nick

1 0

[Users] TLS problem, not able to start openSER
by Ncheeku Baranov 27 Dec '06

27 Dec '06

Hi, I just compiled openSER with TLS support. I checked that TLS = 1 in the Makefile when I compiled openSER. Now when I try to uncomment the parameters in the openser.cfg to enable the TLS support and restart openSER it does not start (I am using openserctl start command to start openser). It gives an error saying ERROR:PID file /var/run/openser.pid does not exist -- OpenSER start failed. I am using the following parameters in the openser.cfg file for the TLS support: disable_tls = 0 listen = tls:10.30.100.41:5061 tls_verify = 1 tls_require_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "usr/local/etc/openser/tls/user/user-calist.pem" I have checked that all the paths are correct in defining the tls_certificate, tls_private_key and tls_ca_list. I used the source tarball openser-1.1.0-tls_src.tar.gz<http://www.openser.org/pub/openser/latest/src/openser-1.1.0-tls_src.tar.gz> for installing the openser. Your help is much appreciated. Thanks NCheeku

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[Users] SER vs openSER
by pankaj.munjal＠wipro.com 27 Dec '06

27 Dec '06

Hi all, I am a SER/openSER newbie and was looking at an option of using these SIP stacks. Anyone having any doc/link that compares these two SIP servers. The major criteria that we are looking for are: - Simplicity of using the SIP stack (Most IMP) - Perfomance - Support for Presence Any pointers in this direction will be highly appreciated. Thanks, Pankaj The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Users] subst_body with nated client problem
by Oleg A Chernov 27 Dec '06

27 Dec '06

Hello everyone! My setup is as follow: for local PSTN calls UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid - PSTN gateway (G.711 codec only) for international calls UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid - ITSP gateway (G.729 codec only) B2BUA on the second leg (B2BUA - gateway) always negotiate the right codec, so no problem, but on the first leg (UA - B2BUA) it always negotiate codec that is more prefered by the client UA side. So we can get situation, when the codecs are different on the both legs. Inspired by the http://lists.iptel.org/pipermail/serusers/2006-August/030129.html I decided to normalize codec preference. Sorry for such a long introduction, at last here goes my problem. I have the following code in my openser.cfg in INVITE section # INVITE section route[3] { ... if ( route to G.711 capable gateway) { subst_body("/^m=audio(.*)RTP\/AVP(.*) 8 (.*)/m=audio\1RTP\/AVP 8 \2 \3/ig"); .... } ... } and it works like a charm for non-nated clients with real IPs. But when we should fire use_media_proxy() for a nated one client we get the problem. SDP body before our substitution (UA-Openser): v=0 o=20000 12893 12893 IN IP4 192.168.2.134 s=ATA186 Call c=IN IP4 192.168.2.134 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 After (Openser - B2BUA): v=0 o=20000 12893 12893 IN IP4 192.168.2.134 s=ATA186 Call c=IN IP4 217.77.208.184 t=0 0 m=audio 16384 RTP/AVP 8 0 18 101\01560136\012a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 So our subst_body() rearrange codecs as expected, but use_media_proxy() (it her fault, I think) inserted the mediaproxy fixed port number 60136 between symbols with ASCII codes \015 CR and \012 LF. This behavior was accepted by stable Openser 1.1 and latest cvs 1.2. So the question is : What to do and who is guilty ? :-) -- Best regards, Oleg A. Chernov OAC4-RIPE WildPark ISP, Nikolaev, Ukraine +380512 470555 +380512 500314

2 1

[Serusers] SER server as a client
by jk 27 Dec '06

27 Dec '06

Hello List, I am working on an assignment to develop a load balancing asterisk cluster. I started with google and found out that SER+ASTERISK is the best way to do it. Here is what I am trying to do. I have a sip account / DID (18xx xxx xxxx ) from an abc company. Now the company is gateway for between my asterisk server and PSTN network. My Asterisk server registers with abc company's gateway to receive and make calls. How can I do load balancing in this case. I understand that my SER server should register with the gateway, the gateway should forward call to SER server and then the server should forward call to one of the asterisk servers. Is it possible have the SER server register with another SER or Asterisk server and act like a client to receive and make calls? Can some one please point me to some documentation or case study? Thank you, -JK

2 2

[Users] LCR db_only mode problem
by Jayesh Nambiar 26 Dec '06

26 Dec '06

Hi all, A merry christmas and a great new year to all of you. I am trying to implement LCR in my openser 1.1. My problem is that I have around 40,000 prefix patterns which i need to enter into the LCR table. When I populate my LCR table and try to start my openser it dies. I can see the following messages in my log: convert_row: No memory left convert_rows: Error while converting row #9055 convert_result: Error while converting rows store_result: Error while converting result lcr_reload_gws(): Failed to query lcr data ERROR: lcr:mod_init(): failed to reload gateways and routes init_mod(): Error while initializing module lcr The openser starts properly when I tested with 1000 rows in the LCR table but with 40,000 rows it just dies. Can anyone please help me overcome this issue. Is there any parameter available to make openser properly run with these many rows in the LCR table. I am using the LCR in db_only mode. I have not tried this in caching mode. Will the caching mode work with these many rows?? w/regards, jayesh. Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php

2 1

[Serusers] Flex and Bison
by Digital Tuffy 26 Dec '06

26 Dec '06

Could anybody pls brief the use of Flex and Bison with SER. _________________________________________________________________ Christmas is drawing near and the excitement is in the air! Check out this cool photofeature on Christmas celebrations the world over http://www.msnspecials.in/photogallery/photogallery.asp?id=a200611814137906

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[Users] Sending REGISTER adn INVITE--plz help
by imran ahmed 26 Dec '06

26 Dec '06

Hi Users; I am new to use openser so having some problems. I have installed openser-1.0.0.I have the task to do load test of openser through SIPp with AUTHENTICATION. I have added some users in the subscriber table of openser. Now the problem is how to set the openser.cfg so that it can response for INVITE and REGISTER request both like: The flow goes like: 1)SIPp UAC send REGISTER -------> OPENSER 2)407 Proxy Authentication required <------ OPENSER 3)SIPp send REGISTERwith credential------>OPENSER(check from subscrible tables and on the bases of that allow or disallow) hope someone help me; Thanks and Regards; Imran __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Re: [Users] OpenSer Sending CANCEL after receiving 200 OK for a INVITE
by Klaus Darilion 26 Dec '06

26 Dec '06

Hi! TAke a look at this line: Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's destination set: $ds=Contact: sip:02111@xx.87.220.xx:6060, sip:02111@xx.87.220.xx:6060 The destination set include 2 destinations: sip:02111@xx.87.220.xx:6060 and sip:02111@xx.87.220.xx:6060 Thus, openser will fork the call with 2 branches (if you take a look at the message flow you will see 2 INVITEs from openser to Asterisk which are nearly identical, only the branch parameter in openser's Via header will be different.) Asterisk is buggy and does not recognize that the 2 INVITEs are different branches but think the second INVITE is a retransmission. Thus when Asterisk accepts the call (accepting on of the two branches), openser will cancel the other branch. As the second branch is identical to the first one, it will send the CANCEL to Asterisk. As Asterisk cann not handle multiple dialog it wil think this is the recently accepted call and will hang up. Conclusion: fix your openser config. Probably in your exec script (or somewhere in openser.cfg) you create a second branch which will be filled with the same URI as the first destination. regards klaus PS: please always Cc the mailing list. Md. Samiul Aftad Chowdhury wrote: > Sorry for quite a late reply... > > but i found same scenario when i am sending the call to an asterisk box. > > Here is the log you asked for: > > In following scenario: > > xx.35.32.xx <- My Openser IP which is running on 5060 port > > xx.87.220.xx:6060 <- Asterisk Box where i am forwarding the calls > which is running on 6060 port > > xx.88.13.xx <- ipphone ip > > Same thing is happening here just after receiving 200 OK Openser > generating a CANCEL. > > i am using : > > ============================================ > exec_dset("/usr/local/route"); # here route is a script which returns > a desired host port (URI) > append_branch(); > ============================================ > > to rewrite the host port. > > > Dec 20 14:47:45 openser[17368]: Invite -> > [CallerID=sip:707234121839@xx.35.32.xx] and [DialedNo=02111] > Dec 20 14:47:46 openser[17366]: Invite -> > [CallerID=sip:707234121839@xx.35.32.xx] and [DialedNo=02111] > > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > first branch: $br=sip:02111@xx.87.220.xx:6060 > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > all branches: $bR=sip:02111@xx.87.220.xx:6060 > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination set: $ds=Contact: sip:02111@xx.87.220.xx:6060, > sip:02111@xx.87.220.xx:6060 > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination uri: $du= > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > uri: $ru=sip:02111@xx.87.220.xx:6060 > > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > first branch: $br=<null> > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > all branches: $bR= > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination set: $ds=<null> > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination uri: $du= > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > uri: $ru=sip:02111@xx.87.220.xx:6060 > > > > On 12/7/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: >> >> Hi! >> >> I could not find any suspect messages. I guess there is some parallel >> forking inside openser, thus, the second branch is cancelled when the >> first branch picks up. >> >> Can you add this just before t_relay(): >> >> xlog("L_INFO","$ci request's first branch: $$br=$br\n"); >> xlog("L_INFO","$ci request's all branches: $$bR=$bR\n"); >> xlog("L_INFO","$ci request's destination set: $$ds=$ds\n"); >> xlog("L_INFO","$ci request's destination uri: $$du=$du\n"); >> xlog("L_INFO","$ci request's uri: $$ru=$ru\n"); >> >> >> regards >> klaus >> -- Klaus Darilion nic.at

2 1

[Users] Merry Christmas and A Happy New Year!
by Daniel-Constantin Mierla 24 Dec '06

24 Dec '06

Hello, it was a year full of achievements and events for OpenSER. We had a major release in summer (1.1.0), and a continuous increase in features set and robustness. What was new in 1.1.0 was summarized in release news: http://www.openser.org/index.php?option=com_content&task=view&id=47&Itemid=… Since then we went though many milestones. We had the first OpenSER summit where we were able to meet many folks from the lists, have nice chats and present interesting VoIP solutions based on OpenSER. http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… A very important statistic is the number of new modules and features introduced in current development version after 1.1.0: - domainpolicy - policies to connect federations - imc - instant messaging conferencing - mi_fifo and mi_xmlrpc - FIFO and XMLRPC transports for the new management interface (MI) - perl - embed perl programming in configuration file - presence - SIMPLE Presence Server implementation - pua, pua_mi, pua_usrloc - presence user agent client implementations for user location records and management interface - seas - connector to SIP Application Server - WeSIP - Java SIP Servlet Application Server (http://www.wesip.eu) - snmpstats - SNMP (Simple Network Management Interface) interface to OpenSER statistics - sst - SIP session timer support - xmpp - transparent SIP-XMPP gateway Documentation and news about all these: http://www.openser.org/docs/modules/1.2.x/ http://www.openser.org/index.php?option=com_content&task=blogsection&id=2&I… A lot of other significant improvements and addons were included in existing code (management interface, fetch database support, postgres, usrloc, dialog, enum, acc modules polishing), openser web administrator was developed. There is not very long time to the next release, along with few pending new features, the next period will be allocated to core components like timers, dns and configuration file flexibility. By end of January we should approach a new testing phase. This year the OpenSER community collaboration exploded, dokuwiki has lot of content, many tutorials were submitted, new developers and many modules sponsored by companies, mailing lists and web forum are places to get reliable information about OpenSER... We want to thank to all of you (developers, contributors with patches, tools and documentation, testers and OpenSER users) for supporting the project. 2006 was a very good year for OpenSER and 2007 looks very challenging. We wish you Merry Christmas and a very Happy New Year!

2 1

[Serusers] Service Empty
by raj gopal 23 Dec '06

23 Dec '06

Hi All, I am trying presence snapshot SER. I am getting 0(4189) Empty service! 0(4189) ERROR: rls_handler.c:484: XCAP query problems 0(4189) XCAP query failed error for SUBSCRIBE request. I am following presence handbook. Whats worng i am doing????? Can anyone help me?? Thanks in advance Venkat

1 0

[Serusers] Re: [asterisk-users] Happy X-mas
by Arun Kumar 23 Dec '06

23 Dec '06

Hi All, Wish you a very HAPPY and Merry Christmas to all and your beloved once. Arun On 12/23/06, Josué Conti <josueconti(a)gmail.com> wrote: > > Hi ALL, > > ** I like very to desire you and your family, a Merry Christmas, with much > love, peace, professional and personal success. > > Best Regards > Josue > > > 2006/12/23, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com>: > > > > > > Hello > > * * * * * * * * > > * * * * * Happy X-mas and Adv Happy New Year ....... > > ************************************ > > **** > > **** > > ******** > > ** > > > > ************* > > > > > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > > www.hyperion-tech.com > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com<http://easynews.com/>-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >

1 0

[Users] Happy X-mas
by raviprakash sunkara 23 Dec '06

23 Dec '06

Hello * * * * * * * * * * * * * Happy X-mas and Adv Happy New Year ....... ************************************ **** **** ******** ** ************* -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Users] Re: [Devel] New module for OpenSER - Application Server interconnection
by Bogdan-Andrei Iancu 22 Dec '06

22 Dec '06

Hi Elias, (also fwd. to "users" to let everybody know) thanks for the contribution. I think that the new SEAS and PERL modules will give a lot more options when comes to integrate OpenSER with other application and what is more important , they will allow non-C extensions :) !! I will generate the docs and place them on the web also. BTW, are some default application that I can give a fast try? I would like to test some real-case apps to get the feeling. Regards and Merry Christmas, Bogdan Elias Baixas wrote: > Hello everybody, > > I just submitted a new module, called SEAS (which stands for Sip > Express Application Server). > This module allows a third party to create an Application Server and > connect it to OpenSER to interoperate seamlessly. > This permits the Application Server to trigger actions on OpenSER, > such as sending out a request, or replying a UAS transaction, in some > way similarly to what can be achieved using fifo, or MI commands > "t_uac_dlg" and "t_reply", but SEAS provides a much richer API to this > kind of tasks. > The module also allows OpenSER to notify the Application Server when > requests/responses come in, and then transfer the control over the > incoming request entirely to the Application Server. > Furthermore, the module implements a binary protocol that transfers > the SIP message along with information abouth the structure of the > message's headers (ie. where each header starts, and how long it is), > so the Application Server doesn't need to reparse the entire SIP > message, thus improving Application Server's performance very much. > The SEAS protocol also carries information about the transaction being > processed at OpenSER for every message, so that the Application Server > doesn't need to implement all the SIP-transaction machinery, thus > actually profitting from the highly optimized TM module at OpenSER. > > The Application Server can be programmed in any language, because the > SEAS module comunicates with it through a TCP socket. > > At the moment, SEAS module has been written to work with an > Application Server developed by VozTelecom. It is called WeSIP and can > be downloaded freely from http://www.wesip.eu . WeSIP is an > implementation of the SipServlet and HttpServlet standards, so it > allows a programmer to easily implement Web-Sip converged applications > (such as Click2call). > WeSIP is programmed entirely in Java, so you have all the tools and > facilities from Java and J2EE available to program SIP applications, > while already using OpenSER to process the SIP messages and > transactions at the low level, thus profitting both from the high > performance of OpenSER, and the high-level SIP-Application programming > framework which is SipServlet and the JAVA languaje. > > I hope this module will help create new usage scenarios for OpenSER, > allowing it to provide high-level Telephony Applications to users. > > The module is still in a alpha stage, but should be able for > production shortly. > > > best regards, > > Elias Baixas > > VozTelecom, Sistemas > http://www.wesip.eu > http://www.voztele.com > > _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel >

3 3

[Serusers] Invite going to my gateway
by Bill 22 Dec '06

22 Dec '06

Scenario: I have two test phones setup to a SER server. The two phones are 192.168.2.133 and 192.168.2.101. My server is 192.168.2.1. I can call from .101 to .133 fine. When I try to place a call from .133 this is what I get. 0(2734) SIP Request: 0(2734) method: <INVITE> 0(2734) uri: <sip:1001@192.168.2.1> 0(2734) version: <SIP/2.0> 0(2734) parse_headers: flags=1 0(2734) Found param type 232, <branch> = <z9hG4bK7e2902f2a>; state=16 0(2734) end of header reached, state=5 0(2734) parse_headers: Via found, flags=1 0(2734) parse_headers: this is the first via 0(2734) After parse_msg... 0(2734) preparing to run routing scripts... 0(2734) parse_headers: flags=128 0(2734) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2734) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2734) parse_headers: flags=8 0(2734) DEBUG: get_hdr_body : content_length=586 0(2734) end of header reached, state=9 0(2734) DEBUG: get_hdr_field: <To> [29]; uri=[sip:1001@192.168.2.1] 0(2734) DEBUG: to body [1001 <sip:1001@192.168.2.1> ] 0(2734) DEBUG: add_param: tag=4d4e374f5ebe6ab 0(2734) end of header reached, state=29 0(2734) parse_headers: flags=256 0(2734) get_hdr_field: cseq <CSeq>: <1338505295> <INVITE> 0(2734) is_preloaded: Yes 0(2734) grep_sock_info - checking if host==us: 13==13 && [192.168.2.136] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) after_loose: Topmost route URI: 'sip:192.168.2.136:5060;lr' is me 0(2734) parse_headers: flags=256 0(2734) found end of header 0(2734) find_next_route: No next Route HF found 0(2734) after_loose: No next URI found 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) check_self: host != me 0(2734) DEBUG: t_newtran: msg id=45 , global msg id=43 , T on entrance=0xffffffff 0(2734) parse_headers: flags=-1 0(2734) parse_headers: flags=60 0(2734) t_lookup_request: start searching: hash=15747, isACK=0 0(2734) DEBUG: RFC3261 transaction matching failed 0(2734) DEBUG: t_lookup_request: no transaction found 0(2734) SER: new INVITE 0(2734) parse_headers: flags=-1 0(2734) check_via_address(192.168.2.133, 192.168.2.133, 0) 0(2734) WARNING:vqm_resize: resize(0) called 0(2734) DEBUG: reply sent out. buf=0x8104e98: SIP/2.0 1..., shmem=0xb6145728: SIP/2.0 1 0(2734) DEBUG: _reply_light: finished 0(2734) DEBUG: mk_proxy: doing DNS lookup... 0(2734) check_via_address(192.168.2.133, 192.168.2.133, 0) 0(2734) DEBUG: add_to_tail_of_timer[4]: 0xb6143eb8 0(2734) DEBUG: add_to_tail_of_timer[0]: 0xb6143ec8 0(2734) SER: new transaction fwd'ed 0(2734) DEBUG:destroy_avp_list: destroying list (nil) 0(2734) receive_msg: cleaning up 1(2737) DEBUG: timer routine:4,tl=0xb6143eb8 next=(nil) 1(2737) DEBUG: retransmission_handler : request resending (t=0xb6143da0, INVITE si ... ) 1(2737) DEBUG: add_to_tail_of_timer[5]: 0xb6143eb8 1(2737) DEBUG: retransmission_handler : done 0(2734) SIP Request: 0(2734) method: <CANCEL> 0(2734) uri: <sip:1001@192.168.2.1> 0(2734) version: <SIP/2.0> 0(2734) parse_headers: flags=1 0(2734) Found param type 232, <branch> = <z9hG4bK7e2902f2a>; state=16 0(2734) end of header reached, state=5 0(2734) parse_headers: Via found, flags=1 0(2734) parse_headers: this is the first via 0(2734) After parse_msg... 0(2734) preparing to run routing scripts... 0(2734) parse_headers: flags=128 0(2734) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2734) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2734) parse_headers: flags=8 0(2734) DEBUG: get_hdr_body : content_length=0 0(2734) end of header reached, state=9 0(2734) DEBUG: get_hdr_field: <To> [29]; uri=[sip:1001@192.168.2.1] 0(2734) DEBUG: to body [1001 <sip:1001@192.168.2.1> ] 0(2734) DEBUG: add_param: tag=4d4e374f5ebe6ab 0(2734) end of header reached, state=29 0(2734) parse_headers: flags=256 0(2734) get_hdr_field: cseq <CSeq>: <1338505295> <CANCEL> 0(2734) is_preloaded: Yes 0(2734) grep_sock_info - checking if host==us: 13==13 && [192.168.2.136] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) after_loose: Topmost route URI: 'sip:192.168.2.136:5060;lr' is me 0(2734) parse_headers: flags=256 0(2734) found end of header 0(2734) find_next_route: No next Route HF found 0(2734) after_loose: No next URI found 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) check_self: host != me 0(2734) DEBUG: t_newtran: msg id=46 , global msg id=45 , T on entrance=0xffffffff Bill Hissong

4 3

[Users] re: how to pass the Caller ID into uac_replace_from function ?
by don lin 22 Dec '06

22 Dec '06

Hi there, I saw all of you discussed about the postgres and I do have a question which somewhat related to postgres as well (please refer to the end of this email for detail). My question is that I have Caller ID (regular phone number) mapping to a SIP account stored in postgres, when I call from SIP device to a PSTN phone, telco carrier does not accept the SIP From field and requested to use regular phone number. I use exec_msg to call the external C program which read the Caller ID mapping to SIP account. However, I do not know how to pass this Caller ID into uac_replace_from. Were the postgres version work for this purpose ? Please advise if you have any suggestion. Thanks in advance, Don ------------------------------ Message: 5 Date: Thu, 21 Dec 2006 12:20:47 +0200 From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> Subject: Re: [Users] postgres To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: users(a)openser.org Message-ID: <458A5FFF.4060003(a)voice-system.ro> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello Klaus, On 12/21/06 12:05, Klaus Darilion wrote: > Hi Daniel! > > I've checked the postgres script: rel_1_1_0 works fine, but there is a > problem in unstable: The domain for the default admin user is asked > not for serweb tables but for the plain openser tables as well. Also > the subscriber table includes all the serweb columns even if serweb > tables are not installed. I think a merge with mysql version, with an option to say the underlaying DB engine is the right solution, to be easy to maintain -- same as we did with openserctl. I will have that in mind, but the best will be to register it to tracker. Cheers, Daniel > > regards > klaus > > > Daniel-Constantin Mierla wrote: >> Hello, >> >> On 12/20/06 19:36, Mark Price wrote: >>> >>> >>> On 12/20/06, *Klaus Darilion* < klaus.mailinglists(a)pernau.at >>> <mailto:klaus.mailinglists@pernau.at>> wrote: >>> >>> Hi Mark! >>> >>> Postgres should work well - I use it since ser 0.8. Just make sure >>> that >>> the hard disk does not get full, because this breaks the index >>> inside >>> postgres and postgres is getting real slow (re-create the index >>> if it >>> happens). >>> >>> Last time I tested openser_postgres.sh it worked fine. If you find >>> a bug >>> please let us know. >>> >>> Please check to use the latest versions from CVS (for 1.1 use CVS >>> rel_1_1_0) >>> >>> >>> The latest version from cvs doesn't work out of the box with openser >>> build from the release tarball, because the release tarball doesn't >>> included /usr/sbin/openser_gen_ha1 (although I could get past this >>> by replacing it with the md5sum equivelant). >>> >>> However, the release tarball doesn't work because of this bug: >>> https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_… >>> <https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_…> >>> >>> >>> So, 1. is the cvs version of rel_1_1_0 considered safe for >>> production use? >> >> yes, it is the most recommended version to use -- several issues were >> found and fixed since 1.1.0 release. >> >>> 2. if so and there are important fixes such as this in CVS, >>> would openser.org <http://openser.org> consider >>> releasing a point release containing such changes? >> It is an option, and perhaps should be taken in consideration, as >> patch release with not so much packaging. >> >> This may help some which want to maintain platforms updated, it is >> exampled with 1.0.x, bust just replace the version with 1.1.0 in the >> document and should work (rel_1_0_0 => rel_1_1_0) >> http://openser.org/dokuwiki/doku.php/install:openser-from-cvs >> >> Cheers, >> Daniel >> >> >>> Thanks, >>> >>> Mark Price don lin <don12lin(a)yahoo.com> wrote: Hi there, We have a Postgres Database to store the users' informations before we use Openser. To integrate between Postgres and Openser, we write external C programs and call them from Openser with exec_dset function. The issue we have now is with the Caller ID. When the user makes a call to PSTN number from SIP devices (soft phone or hard phone), the telco carrier requests us to send a numerical Caller ID, not the SIP format (such as john.dow(a)abc.com). What I am doing is to add the Caller ID field into the Postgres for each user and write a C program to query the Caller ID field, then Openser calls it with exec_dset functioin. However, I did not find a way to define a variable in Openser to store this Caller ID and pass it to uac_replace_from function. Appreciate for any suggestions. Jeff, You mentioned that you replaced the From Header, may I ask how you do it ? Thanks and Regards, Don ------------------------------ <Message: 4 <Date: Tue, 19 Dec 2006 11:11:09 +0900 <From: Jeff Williams <jeffw(a)globaldial.com> <Subject: [Users] (PR)ACK problem with uac_replace_from <To: users(a)openser.org <Message-ID: <45874A3D.5070206(a)globaldial.com> <Content-Type: text/plain; charset="iso-8859-1" <I seem to have an issue with uac_replace_from. <I am using openser as a media proxy. For invites I replace the numbers <with leading 0's with 61 for Australia and forward the call to our LCR <box which then forwards the call to the correct voice to pstn gateway. <I am trying to use uac_replace_from to set the from address so the <callerid appears correct on the outgoing calls. <This seems to work fine for the INVITE request, but on subsequent PRACKs <and ACKs (which just get loose routed), something strange happens to <the from address. <On invite, the From get re-written: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <sip:0892209080@sipone.com>;tag=1790577941 <For the PRACKs and ACKs, this happens: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <<\032\317\352S\232\222B\230\234K\231*BV\300\030\303\332\ah\303\306i\365\005\016\200\250\000\000\230\016\200\252j\200\002`\000\000\016>;tag=1790577 <941 <There shouldn't be anything happening to the From address on the <(PR)ACKs at all. <Anyone have any ideas? <I have attached a ethereal dump of the call setup. <Jeff __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

[Users] uri, myself, etc., variables, + is_domain_local()
by Mark Price 22 Dec '06

22 Dec '06

Hi, I've got a few questions: 1. where can I find out what all the variables like uri, myself, etc., mean? 2. how come is_domain_local($ruri) has a different result than uri==myself? In particular, I have my external IP address defined as an alias. I.e.: alias=4.2.2.2 When I say if( ! is_domain_local($ruri) ) when $ruri = user(a)4.2.2.2, the check succeeds. When I say if( ! uri==myself ), the check fails (correctly, I think). Thanks, Mark Price

2 1

[Serusers] mysql tables help
by Katty Xiong 22 Dec '06

22 Dec '06

I am tring to use Ottendorf. I have db tables for 0.9.6 on the machine, so I decide to drop the old tables and create new tables using ser_mysql.sh for Ottendorf, but the script keeps on giving me errors. Do I have to delete the entire old version of SER (0.9.6)? thanks, Joy __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] help needed on avpops
by Zoran Milic 22 Dec '06

22 Dec '06

Hi guys, I am trying to set up radius and avp for ages now, and this time i think i am close to it. I have a problem and I can't find any information about it. The thing is, my soft agent (set to uid 101) can't register, and it is getting "Forbidden IP" even with this in the cfg file (see below) I have set up freeradius and radiusclient-ng according to the manual on http://www.openser.org/docs/openser-radius-1.0.x.html. I can't find anything on the error i am getting in my log. If you need my complete ..cfg or log file, just ask, and I will send them promptly. thanks in advance, best regards, Zoran openser.cfg: .... xlog("L_ERR","rpid = $avp(s:rpid)"); xlog("L_ERR","adresa = $avp(s:adresa)"); xlog("L_ERR","adresa2 = $avp(s:adresa2)"); # check the src ip address # avp_load_radius("caller"); if(!avp_check("$adresa", "eq/$adresa/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; .... freeradius/users: .... 101 Auth-Type := Digest, User-Password == "101" Reply-Message = "Authenticated", Sip-Avp += "rpid:101", Sip-Avp += "adresa:10.0.0.157", Sip-Avp += "adresa2:10.0.0.157" .... last entries in the log after the rejected register. .... DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <rpid:101> DEBUG:auth_radius:extract_avp: AVP name is <rpid> DEBUG:auth_radius:extract_avp: AVP val is <101> DEBUG:auth_radius:generate_avps: AVP 'rpid'/0='101'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added rpid = 101 adresa = 10.0.0.157 adresa2 = 10.0.0.157 xl_get_spec_value: error - null sp->itf ERROR:avpops:ops_check_avp: cannot get src value parse_headers: flags=ffffffffffffffff check_via_address(192.168.52.145, 192.168.52.145, 0) DEBUG:destroy_avp_list: destroying list 0x404d9238 receive_msg: cleaning up

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Re: [Users] OpenSER as load balancer for several Asterisk servers
by Edoardo Serra 22 Dec '06

22 Dec '06

Hi Users, I'm here again. After having read more docs I don't feel like using rtpproxy My Asterisks have both public IP Adresses so I don't need to proxy rtp stream I just need a configuration to randomly proxy SIP requests to Asterisk boxes behind SER (for load balancing purposes) and to handle registrations of user to forward incoming calls to them (calls are coming from Asterisks to OpenSER) I'm really stuck on it because I already have such a working setup, with same config files. I'm just trying to replicate it on another site Tnx in advance for help Edoardo My last mail to the ml follow for reference Tnx again Klaus, i'll read docs more in depth and try to understand each change to openser.cfg you suggested >>U 2006/12/20 19:15:35.025446 OOO.OOO.OOO.OOO:5060 -> CCC.CCC.CCC.CCC:21722 >>SIP/2.0 183 Session Progress. >>Via: SIP/2.0/UDP >>OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-3e229802603d7c32-1--d87543-;rport=21722. >> > >^^^^^^^^^^^^^^ > >strange bug again - there must be CCC.CCC.CCC.CCC really strange, do you think it's an openser bug ? The strangest thing is that I just copied openser configs from a working system (Openser + Asterisks) changing just ip addresses OpenSER version is also the same... [...] >># !! Nathelper >>onreply_route[1] { >> # NATed transaction ? >> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >> fix_nated_contact(); >> #force_rtp_proxy(); > ^ >Are you sure it is commented? I do not believe it because the ngrep >shows that the SDP of 200 Ok is rewritten. Sure, I also tried to remove the line without solving the previous problem >Please read the Getting Started Turial from onsip.org carefully. It >describes how you handle NAT correctly and also describes how to do >NAT traversal for in-dialog messages, which is missing in your config. I will. Tnx again Regards Edoardo

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Re: [Users] Load-Balancing
by raviprakash sunkara 22 Dec '06

22 Dec '06

May i guess U Perform the Dispatcher modules ..... May this Hopes u to get On 12/22/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > > > > > On 12/22/06, Thomas Deillon < Thomas.Deillon(a)smart-telecom.ch> wrote: > > > > Hi, > > > > I'm trying to set a load balancing solution with SER and twos Asterisk. > > The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... > > > > The ser will just load balance the traffic on one Asterisk or an other. > > I use the dispatcher module for the SIP message and I use 2 public ip > > addresses for the asterisk and so, I'm able to directly send the RTP > > traffic to them (I didn't use mediaproxy or rtpproxy). > > > > On each asterisk, I set the SER as default GW to be sure that all > > message go through this computer. (Client are behind a FW/NAT and so, I > > would like to be sure that all message come from the same IP > > address:port else, the FW/NAT will DROP these messages). > > > > For the moment, with this solution, I have a problem. The message from > > client will go to SER which will resend it to the asterisk and then, the > > > > asterisk will try to answer responding on the same IP -> Loop detected > > !! > > > > On some device I saw that we can forward the request for the phone to > > the asterisk without changing the source ip address .... > > > > Do you know how can I do this ? > > > > Thanks a lot for your help, > > > > Thomas D > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com > www.hyperion-tech.com -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Users] Routing calls to a sip terminator
by Xantek, Inc. 22 Dec '06

22 Dec '06

I am sure this must have been covered somewhere, but I cannot find it. I am trying to get openser to register with username and password and forward a call to that server. This is for outbound sip termination to the PSTN. If somebody could point me to the right place, I would be grateful. Bill

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[Users] Load-Balancing
by Thomas Deillon 21 Dec '06

21 Dec '06

Hi, I'm trying to set a load balancing solution with SER and twos Asterisk. The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... The ser will just load balance the traffic on one Asterisk or an other. I use the dispatcher module for the SIP message and I use 2 public ip addresses for the asterisk and so, I'm able to directly send the RTP traffic to them (I didn't use mediaproxy or rtpproxy). On each asterisk, I set the SER as default GW to be sure that all message go through this computer. (Client are behind a FW/NAT and so, I would like to be sure that all message come from the same IP address:port else, the FW/NAT will DROP these messages). For the moment, with this solution, I have a problem. The message from client will go to SER which will resend it to the asterisk and then, the asterisk will try to answer responding on the same IP -> Loop detected !! On some device I saw that we can forward the request for the phone to the asterisk without changing the source ip address .... Do you know how can I do this ? Thanks a lot for your help, Thomas D

1 0

[Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 21 Dec '06

21 Dec '06

Hi all, Appreciate if someone can give me a suggestion or even an answer. Since the call from SIP device to PSTN phone did not have Caller ID and our Telco carriers request for the Caller ID, we have to set a number before the calls are routed to PSTN gateway. I did it in Asterisk before with the function calls SetCallerID(CLID) or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these function calls worked. But because of the scalabliity issue, I changed to SER and would like to do the similar thing as in Asterisk. I read the SER document and awared SIP_HF_FROM has the data for the entired FROM field. Is there any other variables for Caller ID only ? I searched the mailing list for SER and did not find any related topic (only found one with title: caller-id with raius using sip-rpid, but it is irrelevant). Thanks in advance. Larry _________________________________________________________________ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&source=…

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[Users] Realtime Call Monitoring
by lists＠infoway.net 21 Dec '06

21 Dec '06

We are Asterisk-only users and currently use Asterisk Flash Operator Panel (http://www.asternic.org/) to get a real-time view of the different phones/extensions, channels, etc. We are planning on enhancing our setup by adding SER/OpenSER in front of our Asterisk systems and to expand into other venues as well. If we wanted to use something like SEMS instead of Asterisk for certain services, is there any tool similar to the Flash Operator Panel (FOP) available for SER/OpenSER + SEMS? I guess we could always continue using FOP with our Asterisk systems fronted by SER/OpenSER. However, for our other plans where we prefer to use SEMS, we would love to see if there is any solution available (free, sponsored, or commercial). Thanks, Daniel

4 6

[Users] Re: any kind of statistics available
by Daniel-Constantin Mierla 21 Dec '06

21 Dec '06

Just to complete the thread, a new SNMP module is available for such purpose. In the announcement are listed several tools that can help to collect statistics and generate grphs. It is a matter of configuration and maybe small shell scripting: http://openser.org/pipermail/users/2006-December/008334.html http://cacti.net/ http://oss.oetiker.ch/mrtg/ At the first sight these tools can be used with FIFO or XMLRPC interface as well, if you write the shell script to collect the statistics. Cheers, Daniel Wienand, Joerg wrote: >> You can make a small script to pull periodically statistics via >> FiFO/XMLRPC into a database/file and from there generate graphics at >> your convenience. >> > > This is what I thought about. I expect that I'm not the only one who is > interested in statistics. So maybe anyone else has done this job before. ;-) > > regards, Jörg > >

1 0

Re: [Users] Problem with avp_check
by Bogdan-Andrei Iancu 21 Dec '06

21 Dec '06

Hi Jose, I think your regexp is not correct - you need to force the end of string, otherwise only a substring (prefix) will be matched. try avp_check("$fU","re/^[0-9]{10}$/g") regards, bogdan Jose Gil Navarrete wrote: > Hi thanks Andrei, I tried with your recomendation but I obtain the > same result, if $fu is: 012345678901, then "To-username > -012345678901-" is inserted > > if(avp_check("$fU","re/^[0-9]{10}/g")) { > insert_hf("To-username: -$rU- \r\n", "Call-ID"); > } else { > sl_send_reply("400", "Peticion Erronea, numero > invalido"); > }; > > > */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* escribió: > > Hi Jose, > > the FM operator in avp_check is not for regular expressions - FM is a > file matching- like operator (like "sip:*@domain). See "man 3 > fnmatch". > > if you want to check against an regexp, you need to use the RE > operator. > See: > http://www.openser.org/docs/modules/1.2.x/avpops.html#AEN384 > > as RE op does not support an AVP as right operand, you should try: > if(avp_check("$fU","re/^[0-9]{10}/g")) { > > regards, > bogdan > > Jose Gil Navarrete wrote: > > > Hi > > I try to check the username in the URI, this is, if the username do > > not have ten digits then send a message, I made: > > > > avp_write("^[0-9]{10}","$avp(s:fm_avp)"); > > if(avp_check("$fU","fm/$avp(s:fm_avp)/g")) { > > insert_hf("To-username: -$fU- \r\n", "Call-ID"); > > } else { > > sl_send_reply("400", "Bad Request"); > > exit; > > }; > > > > but I ever obtain 400 message. > > > > I try too with: > > if(avp_check("$fU","re/[0-9]{10}$/g")) { > > insert_hf("To-username: -$fU- \r\n", "Call-ID"); > > else{ > > sl_send_reply("400", "Bad request"); > > exit; > > } > > > > But I obtain the same message. Any idea? > > > > __________________________________________________ > > Correo Yahoo! > > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > > Regístrate ya - http://correo.yahoo.com.mx/ > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > __________________________________________________ > Correo Yahoo! > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > Regístrate ya - http://correo.yahoo.com.mx/ >

1 0

[Serusers] SER 0.9.x and 0.10.x on the same machine?
by TZieleniewski 21 Dec '06

21 Dec '06

Hi ALL!! Is it possible to have to different releases of ser on the same machine? Have any one tired that? I couldn't find any info the web, please help :) Thanks Bests tomek

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[Users] OpenSER as load balancer for several Asterisk servers
by Edoardo Serra 21 Dec '06

21 Dec '06

Hi guys, I'm having a problem with an OpenSER acting as registrar server and load balancer for many Asterisk servers. In a few words: "users are registering on openser and, when they want to make a call, OpenSER proxies the request to an Asterisk server with the dispatcher module" Here is the intended data flow (SIP goes through OpenSER and media goes directly to Asterisk) User <-- SIP --> OpenSER <-- SIP --> Asterisk User <-- RTP --> Asterisk Both, OpenSER and Asterisks have public IPs I already have a working setup of that and everything seems working correctly. I'm trying to replicate that setup on another site, same configurations of the boxes, same versions of OpenSER and Asterisk, etc... but I'm having monodirectional Audio. Having a look with tethereal I see that OpenSER, when the communication is answered, sends a SIP packet (200 OK) to the user indicating itself as media endpoint instead of the Asterisks. From that moment I see RTP packets flowing from the client to OpenSER This seems really strange to me because I just copied the same configurations file from a working setup to the new installation. Tnx in advance for help. Regards P.S.: Here is my openser.cfg ## $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 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) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #children=4 #port=5060 fifo="/tmp/ser_fifo" #uid=nobody #gid=nobody # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/exec.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser") modparam("usrloc", "timer_interval", 120) modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/voip") modparam("uri_db", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "max_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("nathelper", "natping_interval", 20) # Ping interval 20 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("dispatcher", "force_dst", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) { sl_send_reply("405", "Method Not Allowed"); exit; } if (!method=="REGISTER") { record_route(); }; if ((src_ip==xxx.xxx.xxx.xxx) || (src_ip==xxx.xxx.xxx.xxx)) { # IP of Asterisks if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; exit; }; if (nat_uac_test("3")) { if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) { fix_nated_contact(); force_rport(); setflag(6); # Mark as NATed } } # 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 (method=="REGISTER") { if (!proxy_authorize("domain", "openser_view")) { proxy_challenge("domain", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("domain", "openser_view")) { proxy_challenge("domain", "0"); exit; } if (!check_from()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } } # loose-route processing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); exit; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # ! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if ((src_ip!=xxx.xxx.xxx.xxx) && (src_ip!=xxx.xxx.xxx.xxx)) { # IP of Asterisks ds_select_dst("2", "0"); } if (!t_relay()) { sl_reply_error(); }; } # ! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

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[Serusers] Multiple domain
by TZieleniewski 21 Dec '06

21 Dec '06

Hi ! A have a small technical question:) It considers serving of multiple domain. I would like to make sure that I understand things right. After I read the user doc and README from domain module I came to the following state of art:) 1)one has to create the seperate location and subscriber table for each domain *so if i have for instance two domains : dom1.com and dom2.com then my ser has to have the according dns records in the dns server? 2) after creating the tables i need to add domains record in the database to the domain table *I dont know why but when ever i try to add or show domains o get the following error 500 command 'domain_dump' not available - why is that so? any way instead of an error the serctl command adds the domain record to the database 3) modify the ser.cfg file - add appropriate modparam commands 4) modify the ser.cfg file by adding a regular expression analizys to invoke the save, www_chalenge, proxy_chalenge and lookup correctly with the right domain? Please tell me is my thinking here right? or maybe there is another way As allway thank u for your time. Bests tomek

3 3

Re: [Users] Delayed signalling?
by Jiri Kuthan 21 Dec '06

21 Dec '06

if you happen to have a PCAP file with the incident, let me please know. -jiri p.s. even if you didn't tweak timers, the results may be suboptimal because the software version you are using is having rather indeterministic timer subsystem. For example, the recent measurements (http://www.iptel.org/ser/doc/performance) show quite scattered server responsiveness under high load. (Note though that the measurement results were achieved in a best-effort manner based on the tester knowledge and understanding of openser and that the result are not officially confirmed by the OpenSER project.) Whether it is indeed the cause is not certain though -- this looks really like a stealth bug. At 15:58 15/12/2006, Max Gregorian wrote: >Thanks very much for all the replies. I shall try and post a config and traces as soon as I can get them from the office. > >Some more information, if it helps: > >Server specs: >- HP ProLiant DL360 G4 (1U rack servers) >- 3 GHz processors (800 MHz FSB) >- 1 GB RAM >- 10K rpm SCSI HDs (in a RAID 1+0 Mirror) > ># Servers are running OpenSER 1.0.1 (no-TLS). ># Servers are listening on 3 ports (both tcp and udp for each port), so in openserctl ps I am seeing 4 child processes for each port. ># Servers running CentOS Linux 4.3 ># MySQL installed when CentOS was installed but not running and not currently being used with Openser. > > >Things I have pretty much managed to eliminate are: >1. It's doesn't seem to be hardware. The specs for the servers are more than sufficient I think. >2. It doesn't seem to be traffic/load related as I see these problems on 2 brand new servers I have just installed with no traffic on them. However, it does seem to get worse with more traffic. >3. I don't think it's database related as I have deliberately not configured mysql on any of the servers in case of database performance. >4. I haven't played with the timers at all so far. >5. I haven't configured nscd yet, but as far as I can tell it's not caching DNS. >6. Though openser is listening on tcp ports as well, currently only the udp ports are being used as most of our customers use hardware phones. In any case, I haven't as yet seen as requests on tcp. >7. I am not sure it is DNS as in the tests I ran I sent requests directly to the external IP of the server and not to the domain name it is responsible for. Also the test servers are now only responsible for one domain, but in future will have more than one. >8. Also TTL on the domain name is really short. Ping from the server itself TTL=64 and ping times are low as you would expect (< 1ms when pinging from the server itself). Ping from outside the network (from the internet - for me - tp the domain was) 12ms (average), no packet loss, TTL = 53. >9. I have not setup any internal DNS entries for the domain. Servers are resolving domain from entries in /etc/hosts. > >Like I said, it doesn't happen all the time - just maybe once or twice every hour on the servers with more traffic. > >I ran SIPp pointing at one of the new servers last week and at around 100CPS I was seeing about 2,000 out of approx. 10,000 calls were failing. Setup was UAC -> openser -> UAS (Both UAC and UAS were running on the same machine, but different ports). Again there is no traffic on these servers now so I have no idea why so many failed calls. > >I am not sure if any of this information helps, but I am certainly open to suggestions on things to try. > >Thanks in advance. > > > >On 12/14/06, samuel <<mailto:samu60@gmail.com>samu60(a)gmail.com> wrote: >It might be due to a DNS query....whenver a request has to be >forwarded to a domain, openSER makes a DNS query to resolv the IP. >During this operation, the child processing the request will not >answer to further incoming messages. > >it also can be happening due to a spiral loop that stays on the server. > >Without further information (confg,logs) it's hard to tell which is >the reason... > >hope it helps, >Samuel. > >without more information > >2006/12/14, Max Gregorian <<mailto:gregorian442@googlemail.com>gregorian442(a)googlemail.com>: >> Hi all, >> >> Just wondering if anyone else has had this problem. I have noticed while >> tracing on my OpenSER server, that every now and then the server receives a >> packet which it does to respond to immediately, resulting in a string of >> packets being sent to the server and then the server responding a few >> seconds later. This does not happen all the time, just say maybe once or >> twice every hour. The rest of the time the signaling is correct and >> responses follow request packets in the correct order. >> >> What I am trying to figure out is whether this is a load traffic issue (i.e. >> can the server not handle too much load), and if so is it OpenSER or the >> network or the server in general? I have run diagnostics on the servers and >> there is nothing wrong with the hardware. >> >> On the other hand Could this be related to any timer issues? I remember >> there was mention of timers in SER but are there any default timer settings >> that can be tweaked? >> >> Thanks in advance for any response. >> >> >> _______________________________________________ >> Users mailing list >> <mailto:Users@openser.org>Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users -- Jiri Kuthan http://iptel.org/~jiri/

4 4

[Users] Problem with avp_check
by Jose Gil Navarrete 21 Dec '06

21 Dec '06

Hi I try to check the username in the URI, this is, if the username do not have ten digits then send a message, I made: avp_write("^[0-9]{10}","$avp(s:fm_avp)"); if(avp_check("$fU","fm/$avp(s:fm_avp)/g")) { insert_hf("To-username: -$fU- \r\n", "Call-ID"); } else { sl_send_reply("400", "Bad Request"); exit; }; but I ever obtain 400 message. I try too with: if(avp_check("$fU","re/[0-9]{10}$/g")) { insert_hf("To-username: -$fU- \r\n", "Call-ID"); else{ sl_send_reply("400", "Bad request"); exit; } But I obtain the same message. Any idea? __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/

2 1

[Users] flags in reply route
by Klaus Darilion 21 Dec '06

21 Dec '06

Hi! Do I have access to the flags of a transaction in a reply route? regards klaus -- Klaus Darilion nic.at

2 1

Re: [Serusers] branch parameter
by Tuffy Dhawan 21 Dec '06

21 Dec '06

SER is identifying call even W/O branch parameter and i have not used forward(). My only question was to make sure that RTC/1.2 should generate branch parameter. >From: Martin Hoffmann <hn(a)nvnc.de> >To: Jiri Kuthan <jiri(a)iptel.org> >CC: Kapil Dhawan <digitaltuffy(a)hotmail.com>, serusers(a)lists.iptel.org >Subject: Re: [Serusers] branch parameter >Date: Mon, 18 Dec 2006 16:58:00 +0100 > >Jiri Kuthan wrote: > > At 17:05 17/12/2006, Kapil Dhawan wrote: > > > > > I am using RTC/1.2 library and its not generating branch parameter. > > > > too bad, most implementation can deal with it in backw2ards >compatibility mode > > but clients should no longer generate such branch. I would be looking >for > > a more up-to-date stack if I was you. > ><evil> >Ever used forward() in SER? ></evil> > >Regards, >Martin _________________________________________________________________ Christmas is drawing near and the excitement is in the air! Check out this cool photofeature on Christmas celebrations the world over http://www.msnspecials.in/photogallery/photogallery.asp?id=a200611814137906

4 4

Re: [Users] How to get CDR in Openser
by Klaus Darilion 21 Dec '06

21 Dec '06

Howard Tang wrote: > Hi klaus, > > I am new to the VOIP / Voice world. Thank you for your reply. I just > want openser to generate CDR. Do i need radius to do that? > > Regards, > Howard > > Klaus Darilion wrote: >> Howard Tang wrote: >>> Hi, >>> >>> Can someone give me some idea of how i can obtain CDR from Openser >>> please? >>> >>> I saw a open source software called cdrtools, however, i don't want >>> to install apache on my machine. >> >> cdrtools does not generate CDRs. It just analyzes the CDRs. >> >> Thus, what do you want? >> Creating CDRs or analyzing? >> >> openser can create CDRs for DB, Radius or syslog using the acc module >> (see module documentation on openser.org) no, you do not need radius. you can alos use syslog accounting or database accounting. http://www.openser.org/docs/modules/1.1.x/acc.html btw: please cc the mailing list regards klaus >> >> regards >> klaus >> >>> >>> Any other CDR software will work with openser >> >>> >>> Regards, >>> Howard >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > -- Klaus Darilion nic.at

2 2

[Users] A question about cancels..
by nick 21 Dec '06

21 Dec '06

When I make a call to someone, a call which is correctly connected, for example via PSTN gateway, I can watch INVITEs, ACKs, OKs and BYEs being relayed from my station the my openser to the gateway, and back.. However if the phone rings, and I decide to terminate the call (before the other person responds), I see the CANCEL going from my station to the sipserver, but I don't see it being sent to the PSTN gateway.. Is this something I have to specifically route for? The CANCEL is part of the same call sequence, is there a reason it wouldn't be relayed, other than a mistake in my routing, obviously.. Nick

2 1

Re: [Serusers] Problem with MTU size in SER with MIPv6.
by Jose Angel Calvo 21 Dec '06

21 Dec '06

Now, i'll explain and show the INVITE causing the crash: The INVITE packet is fragmented in two frames. The capture is done with the ethereal (wireshark now) and it's the next: *********************************************************************** No. Time Source Destination Protocol Info 1 0.000000 2001:720:1500:24:fcfd:ff:fe0d:101 2001:720:1500:1b:20c:76ff:fe6c:4658 IPv6 IPv6 fragment (nxt=UDP (0x11) off=1408 id=0x5) Frame 1 (431 bytes on wire, 431 bytes captured) Arrival Time: Nov 30, 2006 16:40:02.345512000 Time delta from previous packet: 0.000000000 seconds Time since reference or first frame: 0.000000000 seconds Frame Number: 1 Packet Length: 431 bytes Capture Length: 431 bytes Protocols in frame: sll:ipv6:data Linux cooked capture Packet type: Unicast to another host (3) Link-layer address type: 1 Link-layer address length: 6 Source: Cisco_17:28:70 (00:d0:06:17:28:70) Protocol: IPv6 (0x86dd) Internet Protocol Version 6 Version: 6 Traffic class: 0x00 Flowlabel: 0x00000 Payload length: 375 Next header: IPv6 fragment (0x2c) Hop limit: 61 Source address: 2001:720:1500:24:fcfd:ff:fe0d:101 (2001:720:1500:24:fcfd:ff:fe0d:101) Destination address: 2001:720:1500:1b:20c:76ff:fe6c:4658 (2001:720:1500:1b:20c:76ff:fe6c:4658) Fragmentation Header Next header: UDP (0x11) Offset: 1408 More fragments: No Identification: 0x00000005 Data (367 bytes) 0000 00 03 00 01 00 06 00 d0 06 17 28 70 00 00 86 dd ..........(p.... 0010 60 00 00 00 01 77 2c 3d 20 01 07 20 15 00 00 24 `....w,= .. ...$ 0020 fc fd 00 ff fe 0d 01 01 20 01 07 20 15 00 00 1b ........ .. .... 0030 02 0c 76 ff fe 6c 46 58 11 00 05 80 00 00 00 05 ..v..lFX........ 0040 50 44 41 54 45 0d 0a 52 65 70 6c 61 63 65 73 3a PDATE..Replaces: 0050 20 64 34 63 38 37 63 39 30 32 64 31 62 35 32 62 d4c87c902d1b52b 0060 32 38 36 62 61 64 39 36 66 33 63 62 62 36 31 34 286bad96f3cbb614 0070 34 40 5b 32 30 30 31 3a 37 32 30 3a 31 35 30 30 4@[2001:720:1500 0080 3a 31 61 3a 32 33 30 3a 31 62 66 66 3a 66 65 31 :1a:230:1bff:fe1 0090 30 3a 35 66 62 33 5d 3b 72 65 61 73 6f 6e 3d 53 0:5fb3];reason=S 00a0 49 50 20 72 6f 75 74 65 20 68 61 73 20 63 68 61 IP route has cha 00b0 6e 67 65 64 0d 0a 43 6f 6e 74 65 6e 74 2d 54 79 nged..Content-Ty 00c0 70 65 3a 20 61 70 70 6c 69 63 61 74 69 6f 6e 2f pe: application/ 00d0 73 64 70 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e sdp..Content-Len 00e0 67 74 68 3a 20 31 39 35 0d 0a 0d 0a 76 3d 30 0d gth: 195....v=0. 00f0 0a 6f 3d 69 73 61 62 65 6c 20 30 20 30 20 49 4e .o=isabel 0 0 IN 0100 20 49 50 36 20 5b 32 30 30 31 3a 37 32 30 3a 31 IP6 [2001:720:1 0110 35 30 30 3a 31 61 3a 32 33 30 3a 31 62 66 66 3a 500:1a:230:1bff: 0120 66 65 31 30 3a 35 66 62 33 5d 0d 0a 73 3d 2d 0d fe10:5fb3]..s=-. 0130 0a 63 3d 49 4e 20 49 50 36 20 32 30 30 31 3a 37 .c=IN IP6 2001:7 0140 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 30 3a 31 20:1500:1a:230:1 0150 62 66 66 3a 66 65 31 30 3a 35 66 62 33 0d 0a 74 bff:fe10:5fb3..t 0160 3d 30 20 30 0d 0a 6d 3d 61 75 64 69 6f 20 32 32 =0 0..m=audio 22 0170 32 32 34 20 52 54 50 2f 41 56 50 20 30 20 33 20 224 RTP/AVP 0 3 0180 34 20 35 20 36 20 38 20 31 35 20 31 38 0d 0a 6d 4 5 6 8 15 18..m 0190 3d 76 69 64 65 6f 20 32 32 32 32 32 20 52 54 50 =video 22222 RTP 01a0 2f 41 56 50 20 32 36 20 33 34 20 33 31 0d 0a /AVP 26 34 31.. No. Time Source Destination Protocol Info 2 0.000117 2001:720:1500:24:fcfd:ff:fe0d:101 2001:720:1500:1b:20c:76ff:fe6c:4658 SIP Request: INVITE sip:[2001:720:1500:1b:20c:76ff:fe6c:4658]:5060;transport=udp Frame 2 (1512 bytes on wire, 1512 bytes captured) Arrival Time: Nov 30, 2006 16:40:02.345629000 Time delta from previous packet: 0.000117000 seconds Time since reference or first frame: 0.000117000 seconds Frame Number: 2 Packet Length: 1512 bytes Capture Length: 1512 bytes Protocols in frame: sll:ipv6:ipv6:udp:sip Linux cooked capture Packet type: Sent by us (4) Link-layer address type: 1 Link-layer address length: 6 Source: fe:fd:00:0c:00:01 (fe:fd:00:0c:00:01) Protocol: IPv6 (0x86dd) Internet Protocol Version 6 Version: 6 Traffic class: 0x00 Flowlabel: 0x00000 Payload length: 1456 Next header: IPv6 (0x29) Hop limit: 64 Source address: 2001:720:1500:1b:fcfd:ff:fe0c:1 (2001:720:1500:1b:fcfd:ff:fe0c:1) Destination address: 2001:720:1500:24:20c:76ff:fe6c:4658 (2001:720:1500:24:20c:76ff:fe6c:4658) Internet Protocol Version 6 Version: 6 Traffic class: 0x00 Flowlabel: 0x00000 Payload length: 1416 Next header: IPv6 fragment (0x2c) Hop limit: 60 Source address: 2001:720:1500:24:fcfd:ff:fe0d:101 (2001:720:1500:24:fcfd:ff:fe0d:101) Destination address: 2001:720:1500:1b:20c:76ff:fe6c:4658 (2001:720:1500:1b:20c:76ff:fe6c:4658) Fragmentation Header Next header: UDP (0x11) Offset: 0 More fragments: Yes Identification: 0x00000005 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) Length: 1775 Checksum: 0x9301 Session Initiation Protocol Request-Line: INVITE sip:[2001:720:1500:1b:20c:76ff:fe6c:4658]:5060;transport=udp SIP/2.0 Method: INVITE Resent Packet: False Message Header Record-Route: <sip:[2001:720:1500:24:FCFD:FF:FE0D:101];ftag=12122347;lr=on> Record-Route: <sip:[2001:720:1500:1B:FCFD:FF:FE0C:103];ftag=12122347;lr=on> Record-Route: <sip:[2001:720:1500:1A:FCFD:FF:FE0A:103];ftag=12122347;lr=on> Record-Route: <sip:[2001:720:1500:19:FCFD:FF:FE0E:101];ftag=12122347;lr=on> Call-ID: 138104335821ad9b21971c7d6059b6c8@[2001:720:1500:1a:230:1bff:fe10:5fb3] CSeq: 1 INVITE From: "ueA" <sip:ueA@domainA.ammi.dit.upm.es>;tag=12122347 SIP Display info: "ueA" SIP from address: sip:ueA@domainA.ammi.dit.upm.es SIP tag: 12122347 To: <sip:ueC@domainC.ammi.dit.upm.es> SIP to address: sip:ueC@domainC.ammi.dit.upm.es Via: SIP/2.0/UDP [2001:720:1500:24:FCFD:FF:FE0D:101];branch=z9hG4bK1205.ddbf87e4.0 Via: SIP/2.0/UDP [2001:720:1500:1B:FCFD:FF:FE0C:103];branch=z9hG4bK1205.c8a66626.0 Via: SIP/2.0/UDP [2001:720:1500:1B:FCFD:FF:FE0C:102];branch=z9hG4bK1205.1cb3a0b4.0 Via: SIP/2.0/UDP [2001:720:1500:1A:FCFD:FF:FE0A:103];branch=z9hG4bK1205.44344f04.0 Via: SIP/2.0/UDP [2001:720:1500:19:FCFD:FF:FE0E:101];branch=z9hG4bK1205.a44e3b34.0 Via: SIP/2.0/UDP [2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;branch=z9hG4bK40e1844512a234e326ef8222f8b204fb Max-Forwards: 12 Contact: "ueA" <sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;transport=udp> Contact Binding: "ueA" <sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;transport=udp> URI: "ueA" <sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;transport=udp> SIP Display info: "ueA" SIP contact address: sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052 P-Preferred-Identity: "ueA" <sip:ueA@domainA.ammi.dit.upm.es>;tag=12122347 Privacy: none P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=181086 Allow: INVITE,ACK,CANCEL,BYE,PRACK,U 0000 00 04 00 01 00 06 fe fd 00 0c 00 01 00 00 86 dd ................ 0010 60 00 00 00 05 b0 29 40 20 01 07 20 15 00 00 1b `.....)@ .. .... 0020 fc fd 00 ff fe 0c 00 01 20 01 07 20 15 00 00 24 ........ .. ...$ 0030 02 0c 76 ff fe 6c 46 58 60 00 00 00 05 88 2c 3c ..v..lFX`.....,< 0040 20 01 07 20 15 00 00 24 fc fd 00 ff fe 0d 01 01 .. ...$........ 0050 20 01 07 20 15 00 00 1b 02 0c 76 ff fe 6c 46 58 .. ......v..lFX 0060 11 00 00 01 00 00 00 05 13 c4 13 c4 06 ef 93 01 ................ 0070 49 4e 56 49 54 45 20 73 69 70 3a 5b 32 30 30 31 INVITE sip:[2001 0080 3a 37 32 30 3a 31 35 30 30 3a 31 62 3a 32 30 63 :720:1500:1b:20c 0090 3a 37 36 66 66 3a 66 65 36 63 3a 34 36 35 38 5d :76ff:fe6c:4658] 00a0 3a 35 30 36 30 3b 74 72 61 6e 73 70 6f 72 74 3d :5060;transport= 00b0 75 64 70 20 53 49 50 2f 32 2e 30 0d 0a 52 65 63 udp SIP/2.0..Rec 00c0 6f 72 64 2d 52 6f 75 74 65 3a 20 3c 73 69 70 3a ord-Route: <sip: 00d0 5b 32 30 30 31 3a 37 32 30 3a 31 35 30 30 3a 32 [2001:720:1500:2 00e0 34 3a 46 43 46 44 3a 46 46 3a 46 45 30 44 3a 31 4:FCFD:FF:FE0D:1 00f0 30 31 5d 3b 66 74 61 67 3d 31 32 31 32 32 33 34 01];ftag=1212234 0100 37 3b 6c 72 3d 6f 6e 3e 0d 0a 52 65 63 6f 72 64 7;lr=on>..Record 0110 2d 52 6f 75 74 65 3a 20 3c 73 69 70 3a 5b 32 30 -Route: <sip:[20 0120 30 31 3a 37 32 30 3a 31 35 30 30 3a 31 42 3a 46 01:720:1500:1B:F 0130 43 46 44 3a 46 46 3a 46 45 30 43 3a 31 30 33 5d CFD:FF:FE0C:103] 0140 3b 66 74 61 67 3d 31 32 31 32 32 33 34 37 3b 6c ;ftag=12122347;l 0150 72 3d 6f 6e 3e 0d 0a 52 65 63 6f 72 64 2d 52 6f r=on>..Record-Ro 0160 75 74 65 3a 20 3c 73 69 70 3a 5b 32 30 30 31 3a ute: <sip:[2001: 0170 37 32 30 3a 31 35 30 30 3a 31 41 3a 46 43 46 44 720:1500:1A:FCFD 0180 3a 46 46 3a 46 45 30 41 3a 31 30 33 5d 3b 66 74 :FF:FE0A:103];ft 0190 61 67 3d 31 32 31 32 32 33 34 37 3b 6c 72 3d 6f ag=12122347;lr=o 01a0 6e 3e 0d 0a 52 65 63 6f 72 64 2d 52 6f 75 74 65 n>..Record-Route 01b0 3a 20 3c 73 69 70 3a 5b 32 30 30 31 3a 37 32 30 : <sip:[2001:720 01c0 3a 31 35 30 30 3a 31 39 3a 46 43 46 44 3a 46 46 :1500:19:FCFD:FF 01d0 3a 46 45 30 45 3a 31 30 31 5d 3b 66 74 61 67 3d :FE0E:101];ftag= 01e0 31 32 31 32 32 33 34 37 3b 6c 72 3d 6f 6e 3e 0d 12122347;lr=on>. 01f0 0a 43 61 6c 6c 2d 49 44 3a 20 31 33 38 31 30 34 .Call-ID: 138104 0200 33 33 35 38 32 31 61 64 39 62 32 31 39 37 31 63 335821ad9b21971c 0210 37 64 36 30 35 39 62 36 63 38 40 5b 32 30 30 31 7d6059b6c8@[2001 0220 3a 37 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 30 :720:1500:1a:230 0230 3a 31 62 66 66 3a 66 65 31 30 3a 35 66 62 33 5d :1bff:fe10:5fb3] 0240 0d 0a 43 53 65 71 3a 20 31 20 49 4e 56 49 54 45 ..CSeq: 1 INVITE 0250 0d 0a 46 72 6f 6d 3a 20 22 75 65 41 22 20 3c 73 ..From: "ueA" <s 0260 69 70 3a 75 65 41 40 64 6f 6d 61 69 6e 41 2e 61 ip:ueA@domainA.a 0270 6d 6d 69 2e 64 69 74 2e 75 70 6d 2e 65 73 3e 3b mmi.dit.upm.es>; 0280 74 61 67 3d 31 32 31 32 32 33 34 37 0d 0a 54 6f tag=12122347..To 0290 3a 20 3c 73 69 70 3a 75 65 43 40 64 6f 6d 61 69 : <sip:ueC@domai 02a0 6e 43 2e 61 6d 6d 69 2e 64 69 74 2e 75 70 6d 2e nC.ammi.dit.upm. 02b0 65 73 3e 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e es>..Via: SIP/2. 02c0 30 2f 55 44 50 20 5b 32 30 30 31 3a 37 32 30 3a 0/UDP [2001:720: 02d0 31 35 30 30 3a 32 34 3a 46 43 46 44 3a 46 46 3a 1500:24:FCFD:FF: 02e0 46 45 30 44 3a 31 30 31 5d 3b 62 72 61 6e 63 68 FE0D:101];branch 02f0 3d 7a 39 68 47 34 62 4b 31 32 30 35 2e 64 64 62 =z9hG4bK1205.ddb 0300 66 38 37 65 34 2e 30 0d 0a 56 69 61 3a 20 53 49 f87e4.0..Via: SI 0310 50 2f 32 2e 30 2f 55 44 50 20 5b 32 30 30 31 3a P/2.0/UDP [2001: 0320 37 32 30 3a 31 35 30 30 3a 31 42 3a 46 43 46 44 720:1500:1B:FCFD 0330 3a 46 46 3a 46 45 30 43 3a 31 30 33 5d 3b 62 72 :FF:FE0C:103];br 0340 61 6e 63 68 3d 7a 39 68 47 34 62 4b 31 32 30 35 anch=z9hG4bK1205 0350 2e 63 38 61 36 36 36 32 36 2e 30 0d 0a 56 69 61 .c8a66626.0..Via 0360 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 5b 32 : SIP/2.0/UDP [2 0370 30 30 31 3a 37 32 30 3a 31 35 30 30 3a 31 42 3a 001:720:1500:1B: 0380 46 43 46 44 3a 46 46 3a 46 45 30 43 3a 31 30 32 FCFD:FF:FE0C:102 0390 5d 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 4b ];branch=z9hG4bK 03a0 31 32 30 35 2e 31 63 62 33 61 30 62 34 2e 30 0d 1205.1cb3a0b4.0. 03b0 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 .Via: SIP/2.0/UD 03c0 50 20 5b 32 30 30 31 3a 37 32 30 3a 31 35 30 30 P [2001:720:1500 03d0 3a 31 41 3a 46 43 46 44 3a 46 46 3a 46 45 30 41 :1A:FCFD:FF:FE0A 03e0 3a 31 30 33 5d 3b 62 72 61 6e 63 68 3d 7a 39 68 :103];branch=z9h 03f0 47 34 62 4b 31 32 30 35 2e 34 34 33 34 34 66 30 G4bK1205.44344f0 0400 34 2e 30 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e 4.0..Via: SIP/2. 0410 30 2f 55 44 50 20 5b 32 30 30 31 3a 37 32 30 3a 0/UDP [2001:720: 0420 31 35 30 30 3a 31 39 3a 46 43 46 44 3a 46 46 3a 1500:19:FCFD:FF: 0430 46 45 30 45 3a 31 30 31 5d 3b 62 72 61 6e 63 68 FE0E:101];branch 0440 3d 7a 39 68 47 34 62 4b 31 32 30 35 2e 61 34 34 =z9hG4bK1205.a44 0450 65 33 62 33 34 2e 30 0d 0a 56 69 61 3a 20 53 49 e3b34.0..Via: SI 0460 50 2f 32 2e 30 2f 55 44 50 20 5b 32 30 30 31 3a P/2.0/UDP [2001: 0470 37 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 30 3a 720:1500:1a:230: 0480 31 62 66 66 3a 66 65 31 30 3a 35 66 62 33 5d 3a 1bff:fe10:5fb3]: 0490 36 31 30 35 32 3b 62 72 61 6e 63 68 3d 7a 39 68 61052;branch=z9h 04a0 47 34 62 4b 34 30 65 31 38 34 34 35 31 32 61 32 G4bK40e1844512a2 04b0 33 34 65 33 32 36 65 66 38 32 32 32 66 38 62 32 34e326ef8222f8b2 04c0 30 34 66 62 0d 0a 4d 61 78 2d 46 6f 72 77 61 72 04fb..Max-Forwar 04d0 64 73 3a 20 31 32 0d 0a 43 6f 6e 74 61 63 74 3a ds: 12..Contact: 04e0 20 22 75 65 41 22 20 3c 73 69 70 3a 5b 32 30 30 "ueA" <sip:[200 04f0 31 3a 37 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 1:720:1500:1a:23 0500 30 3a 31 62 66 66 3a 66 65 31 30 3a 35 66 62 33 0:1bff:fe10:5fb3 0510 5d 3a 36 31 30 35 32 3b 74 72 61 6e 73 70 6f 72 ]:61052;transpor 0520 74 3d 75 64 70 3e 0d 0a 50 2d 50 72 65 66 65 72 t=udp>..P-Prefer 0530 72 65 64 2d 49 64 65 6e 74 69 74 79 3a 20 22 75 red-Identity: "u 0540 65 41 22 20 3c 73 69 70 3a 75 65 41 40 64 6f 6d eA" <sip:ueA@dom 0550 61 69 6e 41 2e 61 6d 6d 69 2e 64 69 74 2e 75 70 ainA.ammi.dit.up 0560 6d 2e 65 73 3e 3b 74 61 67 3d 31 32 31 32 32 33 m.es>;tag=121223 0570 34 37 0d 0a 50 72 69 76 61 63 79 3a 20 6e 6f 6e 47..Privacy: non 0580 65 0d 0a 50 2d 41 63 63 65 73 73 2d 4e 65 74 77 e..P-Access-Netw 0590 6f 72 6b 2d 49 6e 66 6f 3a 20 33 47 50 50 2d 55 ork-Info: 3GPP-U 05a0 54 52 41 4e 2d 54 44 44 3b 75 74 72 61 6e 2d 63 TRAN-TDD;utran-c 05b0 65 6c 6c 2d 69 64 2d 33 67 70 70 3d 31 38 31 30 ell-id-3gpp=1810 05c0 38 36 0d 0a 41 6c 6c 6f 77 3a 20 49 4e 56 49 54 86..Allow: INVIT 05d0 45 2c 41 43 4b 2c 43 41 4e 43 45 4c 2c 42 59 45 E,ACK,CANCEL,BYE 05e0 2c 50 52 41 43 4b 2c 55 ,PRACK,U *********************************************************************** If you want the .cap file for open it with ethereal, ask it for me. Thanks for your help. ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

1 0

Re: [Serusers] Problem with MTU size in SER with MIPv6.
by Jose Angel Calvo 21 Dec '06

21 Dec '06

I'm sorry for the delay, but we had to do a deliverable in this date, and it was imposible to answer before. >Could you try again after runing: >ulimit -c unlimited # you need bash for this >ser -f ser.cfg -w /tmp/ The parameter "unlimited" already was in this node, so I think this is not the problem. The scenario used in our test is the following: - There are two user terminals that are communicated by SIP Communicator program. The INVITE message (problematic paquet) travels through the nodes until it arrive at the Proxy node. In this node is the problem with the core shown in my previous message. We have MIPv6 in our scenario. In our case, we use mipv6-2.0.2 from http://www.mobile-ipv6.org/ . We have ubuntu with kernel 2.6.16.24 Next, i'll show you the configuration file in the Proxy node: ****************************************************************** # # P-CSCF # # ----------- global configuration parameters ------------------------ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=yes # (cmd. line: -R) fifo="/tmp/ser_fifo" listen="pcscf" # ------------------ module loading ---------------------------------- 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/textops.so" loadmodule "/usr/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- # -- 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") { # Append the Path header # FIXME: "Require" and "Supported" issues append_hf("Path: <sip:term@pcscf.domainA.ammi.dit.upm.es;lr>\r\n"); append_hf("Require: path\r\n"); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (method != "REGISTER") { record_route(); }; if (loose_route()) { route(1); break; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } **************************************************************** In the next message i'll show the packet INVITE that produces the error becuase. Otherwise, the message would be very big (>40 KB) and it would not be shown. Thanks. ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

1 0

[Serusers] editing the location table.
by Natambu Obleton 21 Dec '06

21 Dec '06

I want to edit the mysql location table myself, but it seems that SER doesn't read the table everytime a call is received. How can I make it refresh this table? Right now if I stop and restart SER then it will load my new entries. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

3 3

[Serusers] Maximum Time of the call
by Digital Tuffy 21 Dec '06

21 Dec '06

How can we restrict a user for the maximum time allowed for a call. _________________________________________________________________ Get up-to-date with movies, music and TV. Its happening on MSN Entertainment http://content.msn.co.in/Entertainment/Default

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RE: [Serusers] editing the location table.
by Digital Tuffy 21 Dec '06

21 Dec '06

use serctl ul command only which will update location table as well as DATA in Memory which is in the form of linked lists etc. >From: "Natambu Obleton" <nobleton(a)fasttrackcomm.net> >To: <serusers(a)iptel.org> >Subject: [Serusers] editing the location table. >Date: Wed, 20 Dec 2006 11:13:01 -0700 > >I want to edit the mysql location table myself, but it seems that SER >doesn't read the table everytime a call is received. How can I make it >refresh this table? Right now if I stop and restart SER then it will load >my >new entries. > > > >Natambu Obleton > >Network Engineer > >FastTrack Communications > >nobleton(a)fasttrackcomm.net > >(970) 247-3366 office > >(970) 247-2426 fax > > > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Want to know what your favourite stars are up to? Catch the latest gossip in Bollywood on MSN Entertainment! http://content.msn.co.in/Entertainment/Default

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Re: [Users] Delayed signalling?
by Jiri Kuthan 20 Dec '06

20 Dec '06

At 16:03 14/12/2006, samuel wrote: >It might be due to a DNS query....whenver a request has to be >forwarded to a domain, openSER makes a DNS query to resolv the IP. >During this operation, the child processing the request will not >answer to further incoming messages. Hard to say without more input what the cause may be. Indeed common suspects are DNS (or any other blocking operation, such as dataabse) or timers, but again it is hard to say without more input. If it is DNS or TCP-based blocking, it may be worthwhile trying genuine SER/ottendorf which is having IP blacklisting and DNS cachine in there -- this is indeed something which has been pressing us for a while. If it is timer system, then SER should address it it (it must be the latest ottendorf version though). If it is database that blocks itself and SER consequently too, you have to look at the database. (Even if SER would do something about it, it could do hardly better then sending a negative reply "DB unavailable"). -jiri >it also can be happening due to a spiral loop that stays on the server. > >Without further information (confg,logs) it's hard to tell which is >the reason... > >hope it helps, >Samuel. > >without more information > >2006/12/14, Max Gregorian <gregorian442(a)googlemail.com>: >>Hi all, >> >>Just wondering if anyone else has had this problem. I have noticed while >>tracing on my OpenSER server, that every now and then the server receives a >>packet which it does to respond to immediately, resulting in a string of >>packets being sent to the server and then the server responding a few >>seconds later. This does not happen all the time, just say maybe once or >>twice every hour. The rest of the time the signaling is correct and >>responses follow request packets in the correct order. >> >>What I am trying to figure out is whether this is a load traffic issue (i.e. >>can the server not handle too much load), and if so is it OpenSER or the >>network or the server in general? I have run diagnostics on the servers and >>there is nothing wrong with the hardware. >> >>On the other hand Could this be related to any timer issues? I remember >>there was mention of timers in SER but are there any default timer settings >>that can be tweaked? >> >>Thanks in advance for any response. >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] OT: network debugging
by Klaus Darilion 20 Dec '06

20 Dec '06

Hi! As SIP debugging often includes low-level network debugging I think this tutorial can give some valuable inputs how to debug network problems. http://penguin-it.co.il/presentations/BrainShare2006/Troubleshooting%20LAN%… regards klaus -- Klaus Darilion nic.at

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[Serusers] Recommend ATA adaptor ?
by johnk＠koreanet.us 20 Dec '06

20 Dec '06

Hi, Can someone recommend ATA models that work easy with Openser, based on experience? I am currently using Sipura SPA3000 and PAP2 which are highly configurable but at the same time, need much works with trial and error fashion to get them work with Openser, Freeradius, Mediaproxy and CDRTool combination. Especially, I am not quite sure if these are handling "VIA" header correctly. Because setting up Outbound proxy was the only solution I found to get these units send "BYE" to the server, otherwise it will send "BYE" directly to PSTN gateway, resulting forward_reply error, killing a child process and mediaproxy never had a chance to mark "stop time" for acc. Can anyone suggest ATAs that imposes least burden of configuration to work with the setting above? Thanks mush in advance. John K.

3 2

[Users] New OpenSER Module: The SNMPStats Module
by jmagder 20 Dec '06

20 Dec '06

Hello everybody, I recently submitted a new module known as the "SNMPStats module" to the OpenSER CVS head. The SNMPStats module provides an SNMP management interface to OpenSER, and is intended for those who want better manageability of their systems. The module provides general SNMP queryable scalar statistics (useful for discovering system trends), table representations of more complicated data such as user and contact information, and alarm monitoring capabilities. The scalar statistics are very useful for discovering trends in the ways your systems are used. For example, using a tool such as mrtg or Cacti, you can collect SNMP based information over long periods of time, and automatically graph statistics. Some examples of good graphs would be: - The number of messages received each hour of the day. (Using the openserSIPSummaryInRequests scalar) - The number of active dialogs at any time of the day. (Using the openserCurNumDialogs scalar) These graphs may reveal that most SIP traffic happens between say, 14:00-16:00, and that perhaps it would be beneficial to change rate plans based on this data. SNMPStats alarm monitoring provides for a useful tool in maintaining system stability. Currently there are two alarms implemented: - Dialog Alarms: If the number of active dialogs passes a certain threshold, then a message can be sent to a network operation center. The idea is that an administrator should me made aware a system is becoming too heavily loaded before a failure occurs. - Message Queue Alarms: If the number of bytes waiting to be consumed by an OpenSER instance exceeds a threshold, then a message can be sent to a network operation center. The idea is to find out why the system is not able to handle the load, and to take corrective actions before a failure occurs. In order to get the SNMPStats module up and running, you'll be required to do some extra configuration steps that at this time are not automated. The full installation procedure is documented on the OpenSER web site at: http://www.openser.org/docs/modules/1.2.x/SNMPStats.html Since this is the first time the SNMPStats module will be tried outside of my work at SOMA Networks, there may be a few rough edges I've overlooked. I welcome all to take out their sand paper help me polish the module. All suggestions and criticisms are always welcome. Good luck all, and cheers! - Jeffrey Magder - SOMA Networks Inc.

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[Users] Trying to understand difference between forward() and t_relay()
by Juan Carlos Castro y Castro 20 Dec '06

20 Dec '06

I was googling for that and found a reply in the SER list that links to a page that doesn't exist anymore: http://lists.iptel.org/pipermail/serusers/2004-October/011924.html My searchs have been unsuccesful so far. Can anyone give me a pointer? Thanks, Juan

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[Users] postgres
by Mark Price 20 Dec '06

20 Dec '06

Hi, What is the best way to use postgres with openser? Is postgres a viable option? I ask, because the install script appears to be broken. If I am able to fix it well enough create the necessary tables, will it be fit for use? As an example, there are several errors in the openser_postgres.sh that comes with openser 1.1 In particular, there are instances where the following is done: create table a ( field type, index a_idx(field) ); However, postgres requires syntax like this: create table a ( field type ); create index a_idx on a (field); Thanks.

2 1

[Serusers] How to change charset in SER for IM
by Killer 20 Dec '06

20 Dec '06

Hi ! Can someone tell how to change the charset in SER ? I want to send IMs which also shows german letters like "ß" ... I also get these Errors and even google couldnt help me :-( , how to erase these erros WARNING: run_actions: null action list (rec_level=1) 2(4154) ERROR: fifo_server: command must have at least 3 chars 2(4154) ERROR: fifo_server: command must have at least 3 chars Thanks a lot for your answers Ahsen ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

2 1

[Users] Mysql reconnect problem
by Dimo 20 Dec '06

20 Dec '06

Hi All, I have a problem with openser reconnecting to my redundant mysql servers which hold the openser database. The database servers are in a cluster and if one fails the other takes over its IP address. In openser however the connection to the database is established only (i think) at the beginning. So when openser is running and one of my sql servers drops, and the other takes over the IP, i get an error in openser saying that the mysql connection does not exist. Is there a way to make openser reconnect to mysql automatically without restarting it or some other way I can make this work? Best, Dimo

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[Serusers] cannot find directory /src/sip_router/tools/serct
by Franz-Ferdinand Lehnert 20 Dec '06

20 Dec '06

Hello fellows I was looking for the 'magic' scripts (http://ftp.iptel.org/pub/ser/presence/presence-handbook/ar01s07.html) to initialize the postgres data base for ser 0.10xxx (latest snapshot from today) quite some time, but could not find the folder 'src/sip_router/tools' to use the new serctl. Even my search with google gave no result. Can someone help me please? Kind regards leffe

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Re: [Serusers] Help! Run SER with MySQL to do authentication
by Greger V. Teigre 20 Dec '06

20 Dec '06

Copy the list. Well, "libmysqlclient.so.14: cannot open shared object file: No such file or directory" pretty much says it all, right? Your LD_LIBRARY_PATH environment variable must include your library dirs before starting ser (cumbersome) or you need to add /usr/lib/mysql to your /etc/ld.so.conf file (which btw should have been updated when installing mysql, I don't know why it didn't happen for you). g-) Tako Chang wrote: > Thanks!! > I ran make_and_install script > But i still can't run 'auth-mysql.cfg' > Error message : > load_module: could not open module > </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot > open shared object file: No such file or directory > > but I have libmysqlclient.so.14 at "/usr/local/lib/mysql" > What should I do?? > > Thanks for your help <(_ _)> > > I mentioned that someone talked about the version of mysql > So, i download the older version 4.1.22 > 2006/12/6, Greger V. Teigre < greger(a)teigre.com > <mailto:greger@teigre.com>>: > > Look at the make_and_install script in the scripts package (same > place). It does this for you. mysql-devel must be installed. > g-) > > Tako Chang wrote: >> Hi, all >> >> First of all, I read [SER - Getting Started >> <http://siprouter.onsip.org/doc/gettingstarted/>] and downloaded >> [ SER Source package >> <ftp://siprouter.onsip.org/pub/gettingstarted/packages/ser-0.9.7-pre1.Gettin…>] >> >> after I make the package under debug mode, it runs well. >> Now, I can receive REGISTER requests and send 200 OK Responses >> and play a SIP proxy server >> >> I want to test "auth_mysql.cfg" in Chapter 7.... so, >> i remove "exclude_modules?= jabber mysql\" in Makefile >> but i still can't compile the mysql.so >> >> Is there some more detail documents or suggestions for me? >> >> Thank! >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> >> http://lists.iptel.org/mailman/listinfo/serusers >> > >

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[Users] (PR)ACK problem with uac_replace_from
by Jeff Williams 20 Dec '06

20 Dec '06

I seem to have an issue with uac_replace_from. I am using openser as a media proxy. For invites I replace the numbers with leading 0's with 61 for Australia and forward the call to our LCR box which then forwards the call to the correct voice to pstn gateway. I am trying to use uac_replace_from to set the from address so the callerid appears correct on the outgoing calls. This seems to work fine for the INVITE request, but on subsequent PRACKs and ACKs (which just get loose routed), something strange happens to the from address. On invite, the From get re-written: From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 to From: Jeff <sip:0892209080@sipone.com>;tag=1790577941 For the PRACKs and ACKs, this happens: From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 to From: Jeff <\032\317\352S\232\222B\230\234K\231*BV\300\030\303\332\ah\303\306i\365\005\016\200\250\000\000\230\016\200\252j\200\002`\000\000\016>;tag=1790577 941 There shouldn't be anything happening to the From address on the (PR)ACKs at all. Anyone have any ideas? I have attached a ethereal dump of the call setup. Jeff

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[Users] RE: [Serusers] Recommend ATA adaptor ?
by johnk＠koreanet.us 19 Dec '06

19 Dec '06

Thanks for quick reponse. Actually I agree with you fully but the only reason for looking at other choice is that I ran out of ideas to resolve this Sipura "hang up" situations where OpenSER logs "forward_reply" error and Mediaproxy's missing stop-time for the call. The only way of getting around was to setup Sipura to use Outbound proxy of same server, which I dearly want to avoid in production. Any idea how to resolve the issue? > You won't find anything better than Sipura/ Linksys. I tried a lot and it > is > worth it to make a script to configure instead of looking for others. > There > is a lot of trash around also. > > Regards CFM > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of johnk(a)koreanet.us > Sent: Tuesday, December 19, 2006 3:04 PM > To: serusers(a)iptel.org > Subject: [Serusers] Recommend ATA adaptor ? > > Hi, > > Can someone recommend ATA models that work easy with Openser, based on > experience? > > I am currently using Sipura SPA3000 and PAP2 which are highly configurable > but at the same time, need much works with trial and error fashion to get > them work with Openser, Freeradius, Mediaproxy and CDRTool combination. > > Especially, I am not quite sure if these are handling "VIA" header > correctly. Because setting up Outbound proxy was the only solution I > found > to get these units send "BYE" to the server, otherwise it will send "BYE" > directly to PSTN gateway, resulting forward_reply error, killing a child > process and mediaproxy never had a chance to mark "stop time" for acc. > > Can anyone suggest ATAs that imposes least burden of configuration to work > with the setting above? > > Thanks mush in advance. > > John K. > > > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: > 12/18/2006 > 1:45 PM > > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: > 12/18/2006 > 1:45 PM > > >

1 0

[Users] Recommend ATA adaptor ?
by johnk＠koreanet.us 19 Dec '06

19 Dec '06

Hi, Can someone recommend ATA model that works easy with Openser, based on experience? I am currently using Sipura SPA3000 and PAP2 which are highly configurable but at the same time, need much works with trial and error fashion to get them work with Openser, Freeradius, Mediaproxy and CDRTool combination. Especially, I am not quite sure if these are handling "VIA" header correctly. Because setting up Outbound proxy was the only solution I found to get these units send "BYE" to the server, otherwise it will send "BYE" directly to PSTN gateway, resulting forward_reply error, killing a child process and mediaproxy never had a chance to mark "stop time" for acc. Can anyone suggest ATAs that imposes least burden of configuration to work with the setting above? Thanks mush in advance. John K.

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[Users] re: how to pass the Caller ID into uac_replace_from function ?
by don lin 19 Dec '06

19 Dec '06

Hi there, We have a Postgres Database to store the users' informations before we use Openser. To integrate between Postgres and Openser, we write external C programs and call them from Openser with exec_dset function. The issue we have now is with the Caller ID. When the user makes a call to PSTN number from SIP devices (soft phone or hard phone), the telco carrier requests us to send a numerical Caller ID, not the SIP format (such as john.dow(a)abc.com). What I am doing is to add the Caller ID field into the Postgres for each user and write a C program to query the Caller ID field, then Openser calls it with exec_dset functioin. However, I did not find a way to define a variable in Openser to store this Caller ID and pass it to uac_replace_from function. Appreciate for any suggestions. Jeff, You mentioned that you replaced the From Header, may I ask how you do it ? Thanks and Regards, Don ------------------------------ <Message: 4 <Date: Tue, 19 Dec 2006 11:11:09 +0900 <From: Jeff Williams <jeffw(a)globaldial.com> <Subject: [Users] (PR)ACK problem with uac_replace_from <To: users(a)openser.org <Message-ID: <45874A3D.5070206(a)globaldial.com> <Content-Type: text/plain; charset="iso-8859-1" <I seem to have an issue with uac_replace_from. <I am using openser as a media proxy. For invites I replace the numbers <with leading 0's with 61 for Australia and forward the call to our LCR <box which then forwards the call to the correct voice to pstn gateway. <I am trying to use uac_replace_from to set the from address so the <callerid appears correct on the outgoing calls. <This seems to work fine for the INVITE request, but on subsequent PRACKs <and ACKs (which just get loose routed), something strange happens to <the from address. <On invite, the From get re-written: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <sip:0892209080@sipone.com>;tag=1790577941 <For the PRACKs and ACKs, this happens: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <<\032\317\352S\232\222B\230\234K\231*BV\300\030\303\332\ah\303\306i\365\005\016\200\250\000\000\230\016\200\252j\200\002`\000\000\016>;tag=1790577 <941 <There shouldn't be anything happening to the From address on the <(PR)ACKs at all. <Anyone have any ideas? <I have attached a ethereal dump of the call setup. <Jeff __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] How can we do Calling Card System with OpenSER
by raviprakash sunkara 19 Dec '06

19 Dec '06

Hello Users First of All Advance "Marry Chrimas " Is How can we do the Calling Card System with OpenSER, I don't want the route every call to Asterisk ( PBX feature ) and other How can we perform the Billing System in OpenSER for Post Paid anf PrePaid Billing Systeem In Asterisk it can do it... by thrid party open Source .... My main intension is with OpenSER with Calling Card System -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 1

[Users] ACK after INVITE not reach callee
by somsak vattanavakin 19 Dec '06

19 Dec '06

Dear List if ACK message after INVITE message (UAC caller ) can not send to UAC callee. can i do something on openser to complete this 3 way handshaking. openser server bedind NAT best Somsak Vattanavakin 2753-7 Ladprao 99-101 Bangkok.10310 Thailand Tel. (66)0-29350016 , 0-25386251 Fax. (66)0-25140253 www.infowavenet.com

2 1

[Users] client_nat_test
by Andreas Granig 19 Dec '06

19 Dec '06

Hi, Provided that my clients are advised to not use STUN or any other NAT traversal mechanism, is it enough to just use client_nat_test("2"), which checks if source-ip is different from the one in the Via-Header? Should be sufficient for detecting NAT, right? Andy

6 16

[Users] Install SER
by Danilo Vilela Dantas 19 Dec '06

19 Dec '06

Hi How does it install SER 0.9.6 in the Debian 3.1? And your modules? Througth apt-get tool? Thanks! Danilo

3 2

[Users] openser-1.1.0-notls not compile in NetBSD
by Roberto Pereyra 19 Dec '06

19 Dec '06

Hi all ! I trying to complie openser-1.1.0-notls in my netbsd host and not compile. My system is: NetBSD 3.1_RC4 Thanks for any help !! roberto My output is: smtp2: {6} gmake mode=debug all Compiling cfg.tab.c gcc -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"i386"' -DOS='"netbsd"' -DCOMPILER='"gcc 3.3.3"' -D__CPU_i386 -D__OS_netbsd -D__SMP_no -DCFG_DIR='"/usr/pkg/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DEXTRA_DEBUG -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -DHAVE_TIMEGM -DHAVE_SELECT -c cfg.tab.c -o cfg.tab.o cfg.y: In function `yyparse': cfg.y:859: error: `IPTOS_MINCOST' undeclared (first use in this function) cfg.y:859: error: (Each undeclared identifier is reported only once cfg.y:859: error: for each function it appears in.) gmake: *** [cfg.tab.o] Error 1 smtp2: {7} -- Ing. Roberto Pereyra ContenidosOnline Looking for Linux Virtual Private Servers ? Click here: http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bid=56

1 0

[Users] How to get CDR in Openser
by Howard Tang 19 Dec '06

19 Dec '06

Hi, Can someone give me some idea of how i can obtain CDR from Openser please? I saw a open source software called cdrtools, however, i don't want to install apache on my machine. Any other CDR software will work with openser? Regards, Howard

2 1

[Serusers] Missed calls in serweb
by hiSIPatH 19 Dec '06

19 Dec '06

Hi all, i have ser-0.9.6 and serweb (0.9.6), with mysql support and SQL accounting enabled (DEFS+=-DSQL_ACC). Unfortunately i am not able to configure missed calls correctely (not being displayed in serweb and no entry in mysql). mysql> select * from missed_calls; Empty set (0.01 sec) what's wrong with my config? thx a lot... my config looks like: [...] loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|acc|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "log_missed_flag", 3 ) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "log_level", 1) route{ [....] if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Not Found"); acc_log_request("404 Not Found"); acc_db_request("404 missed call", "missed_calls"); break; }; sl_send_reply("404", "User Not Found"); break; }else { setflag(3); t_relay(); break; };

1 0

[Users] avp concat
by vallimamod abdullah 19 Dec '06

19 Dec '06

Hello list, I have 2 avps containing a username ($avp(s:usr))and a domain ($avp(s:domain)) retrieved from db and I would like to build an uri by joining this 2 avps with an '@' in the middle. I have checked avp_subst and avp_printf but it doesn't work as, for what I have understood, they only accept pseudo variables (and not other avps) for the replacement. For example, avp_printf("$avp(s:uri)", "$avp(s:usr)@$avp(s:domain)") is not valid. Is there a (simple) way to achieve what I want (I am using version 1.1.0-notls) ? Thank you, -vma .

2 3

[Users] How to get statistics from the dialog module.
by Leonid Fainshtein 19 Dec '06

19 Dec '06

Hi, I can't get dialog:active_dialogs statistic in my custom module. There is a fragment of my code: stat_var *psvActive_dlg_stat; str sActive_dlg; long lCurrentNumActiveDlg =0; char szStatisticName[ 25]; strcpy( szStatisticName, "dialog:active_dialogs"); sActive_dlg.s = szStatisticName; sActive_dlg.len = strlen( szStatisticName); psvActive_dlg_stat = get_stat( &sActive_dlg ); if (psvActive_dlg_stat) lCurrentNumActiveDlg = get_stat_val( psvActive_dlg_stat ); .... The problem is that get_stat() always returns 0. Also I get error when I try to get the statistic by using openserctl: # openserctl fifo get_statistics dialog:active_dialogs 404 Statistic not found But the following works ok: ClaroSIS22:/usr/local/sbin# openserctl fifo get_statistics dialog: 200 OK Module name = dialog; statistics=3 dialog:active_dialogs = 0 dialog:processed_dialogs = 0 dialog:expired_dialogs = 0 What could be the problem? Thank you in advance, Leonid Fainshtein ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine %tn%

2 1

[Serusers] append_rpid_hf() replacement in CVS-HEAD
by Hendrik Scholz 18 Dec '06

18 Dec '06

Hi! append_rpid_hf() was removed from CVS in 2005 with the following note: "Rpid support dropped, it will be implemented using AVPs." Is there a standard way of implementing this? OpenSER utilizes avp_printf() for this which isn't part of SER: http://www.mail-archive.com/devel@openser.org/msg02248.html What is the Best Current Practice to add the Rpid header? If there is none I'd help implementing something new since I'm already tackling some Rpid to P-Asserted-Identity transformation things. It would also be handy to add the privacy tag according to another AVP (although I'm not using it). Cheers, Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

2 2

[Users] Delayed signalling?
by Max Gregorian 18 Dec '06

18 Dec '06

Hi all, Just wondering if anyone else has had this problem. I have noticed while tracing on my OpenSER server, that every now and then the server receives a packet which it does to respond to immediately, resulting in a string of packets being sent to the server and then the server responding a few seconds later. This does not happen all the time, just say maybe once or twice every hour. The rest of the time the signaling is correct and responses follow request packets in the correct order. What I am trying to figure out is whether this is a load traffic issue (i.e. can the server not handle too much load), and if so is it OpenSER or the network or the server in general? I have run diagnostics on the servers and there is nothing wrong with the hardware. On the other hand Could this be related to any timer issues? I remember there was mention of timers in SER but are there any default timer settings that can be tweaked? Thanks in advance for any response.

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[Serusers] SIPSAK error in username
by "tzieleniewski" 18 Dec '06

18 Dec '06

Hi! I was testing my ser proxy with sipsak and i found one error. invoking sipsak with the following command: # sipsak -IU -vvv -s sip:sipsak@scscf.ims.touk.pl -l 5061 -H 192.168.0.138 -e 1 -S -a sipsak sipsak inserts an extratrailing zero in the username this is both in the to and from headers as well in the authentication header. U 2006/12/18 12:52:48.594546 192.168.0.138:5061 -> 192.168.0.74:5060 REGISTER sip:scscf.ims.touk.pl SIP/2.0. Authorization: Digest username="sipsak0", uri="sip:scscf.ims.touk.pl", algorithm=MD5, realm="scscf.ims.touk.pl", nonce="4586820caf30028588ff88825a90f86abbdb598d", response="66968ba65014ad848a3fbe0cc9331daf". Via: SIP/2.0/UDP 192.168.0.138:5061;branch=z9hG4bK.5fcbc0b0;rport;alias. From: sip:sipsak0@scscf.ims.touk.pl;tag=391d3b1a. To: sip:sipsak0@scscf.ims.touk.pl. Call-ID: 958216986(a)192.168.0.138. CSeq: 2 REGISTER. Content-Length: 0. Max-Forwards: 70. User-Agent: sipsak 0.9.6. Expires: 15. Contact: sip:sipsak0@192.168.0.138:5061. Ive put the whole log in the attached file. Bests Tomasz

2 1

[Users] The usrloc table, Oracle, and Asterisk
by Juan Carlos Castro y Castro 18 Dec '06

18 Dec '06

I'm working on an Asterisk+OpenSER setup and I am in the process of implementing the following: registration has to be handled by OpenSER, but it has to be recognized by Asterisk. For that purpose, OpenSER is authenticating against Asterisk's "sipfriends" realtime table. That works OK. But OpenSER should also update fields "ipaddr", "port", and "regseconds" in the same table, or else Asterisk won't be able to direct calls to the logged SIP phones. I'm thinking of implementing an Oracle trigger (or rather asking the Oracle DBA to do that) so all updates/inserts into the location table will cause updates in sipfriends. As far as I know, I need to take the IP and port from the "contact" field of the location table. Or should I use the "received" field? In my tests, the "received" field gets set to NULL, although "contact" has the info I need. Can I use the data from "contact" always? For the regseconds field, I'll just have to make a conversion from the local time format (used by the "expires" field of location) to the unix timestamp format (used by "regseconds" in Asterisk). Oh, by the way, this is the location table definition that worked OK with Oracle. I had to ask for a login trigger so the date fields would accept the 'YYYY-MM-DD HH:MM:SS' format. Also, Oracle refuses to accept an empty string as a non-null value. CREATE TABLE location ( username varchar2(64) DEFAULT '', domain varchar2(128) DEFAULT 'vono.net.br', contact varchar2(255) DEFAULT '', received varchar2(255) DEFAULT '', path varchar2(255) DEFAULT '', expires date DEFAULT '2020-05-28 21:32:15' NOT NULL, q decimal(10,2) DEFAULT 1.0 NOT NULL, callid varchar2(255) DEFAULT 'Default-Call-ID' NOT NULL, cseq int DEFAULT 42 NOT NULL, last_modified date DEFAULT '1900-01-01 00:00' NOT NULL, flags int DEFAULT 0 NOT NULL, user_agent varchar2(255) DEFAULT '', socket varchar2(128) DEFAULT '', methods int DEFAULT NULL, PRIMARY KEY(username, domain, contact) );

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[Users] Is there any module for P&IM?
by Danilo Vilela Dantas 18 Dec '06

18 Dec '06

Hi all, Congratulations! I use Openser. I have got a question. Has it Instant Messaging? It works like IM? Is there any module for P&IM? Thanks! Danilo

3 2

[Users] calculating phplib_id
by Bill Neely 18 Dec '06

18 Dec '06

Can anybody explain to me how to calculate phplib_id? I have been aboe to calculate ha1 and ha1b, but don't see any instructions on calculating the phplib_id. --

2 1

[Users] OpenSER
by Danilo Vilela Dantas 18 Dec '06

18 Dec '06

Hi all What is version OpenSER for the module IMC? Can it use in the OpenSER 1.0.1? And when will be ready the version 1.2.0? Delay? When time delay? Danilo

2 1

Re: [Serusers] branch parameter
by Tuffy Dhawan 18 Dec '06

18 Dec '06

www.adoreinfotech.com is offering a Softphone based on Microsoft RTC/1.2 which is not generating branch parameter while sending INVITE but call is still going through. How is branch parameter used as per RFC 3261. >From: "Pat Wang" <wangyu39(a)hotmail.com> >To: "Kapil Dhawan" <digitaltuffy(a)hotmail.com>,<serusers(a)lists.iptel.org> >Subject: Re: [Serusers] branch parameter >Date: Mon, 18 Dec 2006 00:17:26 -0000 > >branch in Via header is a MUST have parameter by RFC 3261. > >----- Original Message ----- From: "Kapil Dhawan" ><digitaltuffy(a)hotmail.com> >To: <serusers(a)lists.iptel.org> >Sent: Sunday, December 17, 2006 4:05 PM >Subject: [Serusers] branch parameter > > >>Whats the importance of branch parameter. I am using RTC/1.2 library and >>its not generating branch parameter. >> >>_________________________________________________________________ >>Tried the new MSN Messenger? Its cool! Download now. >>http://messenger.msn.com/Download/Default.aspx?mkt=en-in >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> > _________________________________________________________________ Stay in tune with the latest styles! Check out the happening fashion trends on 'Wardrobe watch'. Only on MSN Lifestyle http://content.msn.co.in/Lifestyle/Moreonlifestyle/LifeStyle_091006_0551.htm

2 1

[Serusers] Denial-of_service
by Tuffy Dhawan 18 Dec '06

18 Dec '06

How do you prevent Denial-Of-service in case of a stateful Proxy. _________________________________________________________________ Catch all the cricketing action right here. Live score, match reports, photos et al. http://content.msn.co.in/Sports/Cricket/Default.aspx

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[Serusers] Request Response Query
by Kapil Dhawan 17 Dec '06

17 Dec '06

As per RFC 3261, SIP Trapezoid UAC1 - INVITE - UAS1 UAS1 - INVITE - UAS2 UAS1 - 100 Trying - UAC1 But i tried to check logs in SER which shows UAC1 - INVITE - UAS1 UAS1 - 100 Trying - UAC1 UAS1 - INVITE - UAS2 Why is it different. _________________________________________________________________ Mega Fare sale on domestic airlines. Call 1-800-11-8747 now! http://india.makemytrip.com/?Camp=MSN&attrib=Dec/Tag/Mega

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Re: [Users] Duplicate INVITEs - t_lookup_request and t_check_trans
by Mahesh Paolini-Subramanya 16 Dec '06

16 Dec '06

quite quite positive. the reason i am so certain is that there is only *one* place in my entire config file that i actually do a t_relay() (or equivalent). I isolated all the code into route[1], to ease maintainability of my (admittedly complex) config file. In route[1], the relevant section is pretty damn close to if (!t_relay()) { if (!is_method("ACK")) { sl_reply_error(); } } fwiw, its 1.1.0 (latest cvs version thereof) cheers ----- Original Message ----- From: Bogdan-Andrei Iancu < bogdan(a)voice-system.ro > To: mahesh(a)aptela.com Cc: Klaus Darilion < klaus.mailinglists(a)pernau.at >, users < users(a)openser.org > Sent: Thursday, December 14, 2006 12:39:04 PM GMT-0600 Subject: Re: [Users] Duplicate INVITEs - t_lookup_request and t_check_trans Hi Mahesh, I do not think you need t_check_tran() as t_relay() automatically absorbs the retransmissions. are you sure you are not calling t_relay() twice for the same requests? or calling a tm function (t_newtran) before the t_relay ?? regards, bogdan Mahesh Paolini-Subramanya wrote: >Well, i read the documentation again, and found t_lookup_request, and t_check_trans. >Question 1 - t_lookup_request() creates the transaction if it doesn't exist, correct? However, t_check_trans only checks to see if the transaction already exists, correct? > >Question 2 - Assuming that t_check_trans only checks to see if the transaction already exists, then is the following correct? > >Currently I do the following (after stripping out a bunch of stuff) > >if (!t_relay()) { > if (!is_method("ACK")) { > sl_reply_error(); > } >} > >should this, instead, actually look like the following? >if (is_method("INVITE")) { > if (if !t_check_tran()) { > t_relay(); > } >} else if (!is_method("ACK")) { > sl_reply_error(); >} > > > >----- Original Message ----- >From: Klaus Darilion < klaus.mailinglists(a)pernau.at > >To: mahesh(a)aptela.com >Cc: users < users(a)openser.org > >Sent: Thursday, December 14, 2006 2:06:01 AM GMT-0600 >Subject: Re: [Users] Duplicate INVITEs > >Mahesh Paolini-Subramanya wrote: > > >>We're seeing sporadic situations where fones >>a) send an INVITE >>b) ignore the 100-Trying response, and instead >>c) send another INVITE with the same sequence number. >> >>The question is, What should happen here? >> >> > >If the message is completely identical then it is a retransmission. >Retransmissions should be detected by tm module (e.g. t_relay()) and >absorbed, > >regards >klaus > > > > >>Currently, we send back a 500 response (server error). >>Is this correct? (i think not, cause it invariably causes the fone to go fast busy). >>Is this some other response that should occur? >> >>poring over 3261 resulted in a headache, and no additional clarity... >> >> >> >> >> >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" ******************************************* -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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Re: usrloc loading (was Re: [Serusers] Re: Fw: [Users] TM : retransmission timers
by Jiri Kuthan 16 Dec '06

16 Dec '06

Thanks, how could I have missed that! Is what it does it loads usrloc table piece-wise? And if so, why does it do only for mysql? And also, it is saying that old version would have increased memory consumption with TCP/TLS -- does it mean that it does not increase with the new version? -jiri At 14:32 30/11/2006, Ovidiu Sas wrote: >Maybe you are looking for this one: >http://www.openser.org/index.php?option=com_content&task=view&id=48&Itemid=9 > > >Regards, >Ovidiu Sas > >On 11/30/06, Jiri Kuthan <jiri(a)iptel.org> wrote: >>At 12:53 22/11/2006, Weiter Leiter wrote: >>>I know that OpenSER loads (only?) faster. >> >>Can folks share with me what the fast-usrloc-loading feature is about? >>I was not successful finding it out. >> >>Thanks! >> >>-jiri >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users > >-- >Jiri Kuthan http://iptel.org/~jiri/

10 18

[Users] determining if From==To
by Mark Price 15 Dec '06

15 Dec '06

Hi, How would one determine if the From address equals the To address, or even better, if the To address is either the same as or an alias of the From address? Thanks Mark Price

4 5

RES: [Users] OpenSER
by Danilo Vilela Dantas 15 Dec '06

15 Dec '06

List of contacts and instant message. ________________________________ De: Adrian Georgescu [mailto:ag@ag-projects.com] Enviada em: sexta-feira, 15 de dezembro de 2006 17:16 Para: Danilo Vilela Dantas Cc: users(a)openser.org Assunto: Re: [Users] OpenSER Hi Danilo, Could you please tell us what features from 1.2.0 do you need so badly? Adrian On Dec 15, 2006, at 8:05 PM, Danilo Vilela Dantas wrote: Hi all What is version OpenSER for the module IMC? Can it use in the OpenSER 1.0.1? And when will be ready the version 1.2.0? Delay? When time delay? Danilo _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] new MediaProxy release 1.8.0
by Adrian Georgescu 15 Dec '06

15 Dec '06

Hello, There is a new MediaProxy release available. To upgrade go to http://mediaproxy.ag-projects.com or download the software directly from: http://mediaproxy.ag-projects.com/mediaproxy-1.8.0.tar.gz Changes from version 1.7.2 to 1.8.0 ----------------------------------- - Added support for radius accounting. The accounting type can be specified in the configuration file. To support radius accounting the pyrad module must be installed (available at http://www.wiggy.net/code/pyrad/). - The old boolead switch dbaccounting in the [Accounting] section of mediaproxy.ini was replaced by accounting which is a string specifying the accounting type to be used: none, radius or database. For those using CDRTool, release 5.0 is available, which works with the new Radius accounting of MediaProxy 1.8.0. To upgrade go to http://cdrtool.ag-projects.com/ or download the software directly from: http://download.dns-hosting.info/CDRTool/cdrtool_5.0.tar.gz Kind regards, Adrian Georgescu

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[Users] NEW FEATURE: new pseudo-variables from dialog module
by Bogdan-Andrei Iancu 15 Dec '06

15 Dec '06

Hi everybody, the dialog module exports 2 new pseudo variables: *DLG_lifetime* - the duration (in seconds) of the dialog. The duration is calculated from the dialog confirmation and the current moment. This PV will be available only for sequential requests, after doing loose_route(). So, it can be used to get timestamps foe sequential requests, or to get the overall dialog duration and to use it for accounting. Closes Feature Request 1611930. https://sourceforge.net/tracker/index.php?func=detail&aid=1611930&group_id=… *DLG_status* - Returns the status of the dialog. This PV will be available only for sequential requests, after doing loose_route(). Value may be: 3 - Confirmed by a final reply but no ACK received yet. 4 - Confirmed by a final reply and ACK received. 5 - Dialog ended. Can be used to delay re-INVITEs until ACK is received and avoid signalling races. Closes Feature Request 1610630. https://sourceforge.net/tracker/index.php?func=detail&aid=1610630&group_id=… regards, bogdan

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[Serusers] Interdomain remote disconnect
by Maciej Żwirski 15 Dec '06

15 Dec '06

Hi, I've a following problem: I'm dialing over voip to another domain. Everything swings except that when a remote party hangs up, I don't get a disconnect signal at my caller. Instead I'm getting ". I'm pretty sure I messed up something with my force/unforce proxy but I just can't get it working right. Note that when I'm calling to the PSTN (same transit proxy) through Asterisk I do get the disconnect signal. Any help would be appreciated :). Below is the config of my local proxy. I can post my transit proxy conf if needed. Thanks, Maciek debug=3 fork=no log_stderror=yes listen=<local_proxy> port=5060 children=4 mpath="/usr/local/lib/openser/modules/" dns=no rev_dns=no alias="yam" fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openserrw@localhost:3306/openser" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "mysql.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "permissions.so" loadmodule "uac.so" loadmodule "acc.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" loadmodule "xlog.so" loadmodule "cpl-c.so" loadmodule "exec.so" #modparam("auth_db","calculate_ha1",1) #modparam("auth_db","password_column","password") modparam("auth_db|permissions|uri_db|usrloc|cpl-c","db_url","mysql://openser:openserrw@localhost:3306/openser") modparam("usrloc", "db_mode", 2) modparam("usrloc|registrar","use_domain",1) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 120) #modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("nathelper","rtpproxy_sock","unix:/var/run/rtpproxy.sock") modparam("nathelper","natping_interval",0) modparam("nathelper","ping_nated_only",1) modparam("registrar","nat_flag",6) modparam("registrar","sip_natping_flag",7) modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) modparam("acc", "radius_flag",1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc", "report_ack",1) modparam("acc|auth_radius|group_radius|avp_radius","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) modparam("avpops", "avp_aliases", "day=i:101;time=i:102") modparam("cpl-c","cpl_table","cpl") modparam("cpl-c","cpl_dtd_file","/usr/local/etc/openser/cpl-06.dtd") #modparam("cpl-c","log_dir","/var/log/openser") modparam("cpl-c","lookup_domain","location") modparam("cpl-c","use_domain",1) modparam("cpl-c","nat_flag",6) modparam("cpl-c","proxy_recurse",2) modparam("cpl-c","proxy_route", 0) #modparam("cpl-c","timer_avp","i:5") route{ #-------------------------------------- # Sanity Check Section #-------------------------------------- if (!mf_process_maxfwd_header("10")){ sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len){ sl_send_reply("513", "Message Overflow"); return; }; #-------------------------------------- # Record Route Section #-------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL"){ xlog("L_INFO","$tu\n $fu\n $ru\n"); unforce_rtp_proxy(); }; #-------------------------------------- # Loose Route Section #-------------------------------------- if (loose_route()) { log(1,"loose route\n"); if (is_method("BYE")) { unforce_rtp_proxy(); sl_send_reply("200","Bye"); # acc_rad_request("200"); # acc_log_request("200"); } if((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403","Forbidden"); return; }; if (method=="INVITE") { if (!allow_trusted()) { if (!radius_www_authorize("")) { www_challenge("","0"); return; } else if(!check_from()) { sl_send_reply("403","Use From=ID"); return; }; consume_credentials(); }; if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; # force_rtp_proxy("1"); force_rtp_proxy("s"); }; route(1); return; }; #-------------------------------------- # Call Type Processing Section #-------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; return; }; if (method=="ACK") { route(1); return; } else if(method=="CANCEL"){ route(1); return; } else if (method=="INVITE"){ if(!cpl_run_script("incoming", "is_stateless")) { # script execution failed t_reply("500", "CPL script execution failed"); }; route(3); return; } else if (method=="REGISTER") { cpl_process_register(); route(2); return; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); return; }; if (!lookup("location")){ # xlog("L_INFO"," $ru \n, $fu \n, $tu \n"); sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { #-------------------------------------- # Default Message Handler #-------------------------------------- t_on_reply("1"); if (!t_relay()) { if(method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #-------------------------------------- # REGISTER Message Handler #-------------------------------------- sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100","Trying"); if(!radius_www_authorize("")) { www_challenge("","0"); return; }; if(!check_to()){ sl_send_reply("401","Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #----------------------- # INVITE Message Handler #----------------------- if (!allow_trusted()) { if(!radius_www_authorize("")) { www_challenge("","0"); return; } else if (!check_from()){ sl_send_reply("403","Use From=ID"); return; }; consume_credentials(); }; if(nat_uac_test("19")) { setflag(6); }; avp_load_radius("caller"); # setflag(1); # setflag(2); route(9); # CLIP/CLIR route(8); # Check if number is blocked route(7); # Set/remove call forwarding route(10); # VPABX lookup("aliases"); if(!is_uri_host_local()) { route(4); route(1); return; }; if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { setflag(22); avp_pushto("$ru", "$avp(s:callfwd)"); route(6); return; }; if (!lookup("location")) { #Outbound domain if (uri=~"^sip:[0-9]{10}@") { route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; if (avp_db_load("$ru/username", "$avp(s:fwdonbusy)")) { if (!avp_check("$ru(s:fwdonbusy)", "eq/$ru/i")) { setflag(26); }; }; if (avp_db_load("$ru/username", "$avp(s:fwdnoanswer)")) { if (!avp_check("$avp(s:fwdnoanswer)", "eq/$ru/i")) { setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { # ---------------------- # NAT Traversal Section # ---------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); # force_rtp_proxy("i"); force_rtp_proxy(); }; } route[5] { #---------------------- # Main Domain Handler #---------------------- if (uri=~"^sip:022448755[0-9]@"){ sl_send_reply("404","User Not Found"); drop; } xlog("L_INFO","Main domain Handler: address changed to <outbound_proxy>\n"); #rewritehostport("<outbound_proxy>:5061"); rewritehost("<outbound_proxy>"); t_on_failure("1"); route(1); } route[6] { # ------------------------- # Call Forwarding Handler # ------------------------- if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); revert_uri(); append_branch(); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); return; }; if (uri=~"^sip:00[0-9]{10}@") { route(4); route(1); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); route(1); return; }; sl_send_reply("404", "3User Not Found"); }; route(4); route(1); } route[7] { ####################################### # Call forwarding setters ####################################### if(uri=~"^sip:\*21\*"){ subst_uri('/^sip:\*21\*(.*)@(.*)$/sip:\1@yam/'); avp_db_delete("$fu","$avp(s:callfwd)"); avp_write("$ru","$avp(s:callfwd)"); avp_db_store("$fu","$avp(s:callfwd)"); sl_send_reply("603", "Call forwarding set"); drop; }; if(uri=~"^sip:\*31\*"){ subst_uri('/^sip:\*31\*(.*)@(.*)$/sip:\1@yam/'); avp_db_delete("$fu","$avp(s:fwdonbusy)"); avp_write("$ru","$avp(s:fwdonbusy)"); avp_db_store("$fu","$avp(s:fwdonbusy)"); exec_msg("/home/pgrabowski/busy_fwd_cpl"); sl_send_reply("603", "Call forwarding on busy set"); drop; }; if(uri=~"^sip:\*41\*"){ subst_uri('/^sip:\*31\*(.*)@(.*)$/sip:\1@yam/'); avp_db_delete("$fu","$avp(s:fwdnoanswer)"); avp_write("$ru","$avp(s:fwdnoanswer)"); avp_db_store("$fu","$avp(s:fwdnoanswer)"); exec_msg("/home/pgrabowski/noanswer_fwd_cpl"); sl_send_reply("603", "Call forwarding on no answer set"); drop; }; if(uri=~"^sip:\*22\*"){ avp_db_delete("$fu","$avp(s:callfwd)"); sl_send_reply("603", "Call forwarding removed"); drop; }; if(uri=~"^sip:\*32\*"){ avp_db_delete("$fu","$avp(s:fwdonbusy)"); exec_msg("/home/pgrabowski/ser_work/rm_busy_fwd_cpl"); sl_send_reply("603", "Call forwarding on busy removed"); drop; }; if(uri=~"^sip:\*42\*"){ avp_db_delete("$fu","$avp(s:fwdnoanswer)"); exec_msg("/home/pgrabowski/ser_work/rm_noanswer_fwd_cpl"); sl_send_reply("603", "Call forwarding on no answer removed"); drop; }; } route[8] { ################################## # Outgoing number blocking ################################## if(avp_check("$avp(s:caller_blocked)","eq/$tu")){ sl_send_reply("403", "Forbidden"); return; }; } route[9] { ################################# # CLIP/CLIR ################################# if(is_avp_set("$avp(s:caller_clip)")){ uac_replace_from("$avp(s:caller_clip)",""); }; if(is_avp_set("$avp(s:clir)")){ uac_replace_from("Restricted","sip:restricted@<local_proxy>"); }; } route[10] { ################################# # VPABX ################################# if(uri=~"^sip:[0-9]{3}@"){ if (is_avp_set("$avp(s:caller_vpabx)")) { xlog("L_INFO","VPABX\n"); subst_uri('/^sip:(.*)$/sip:$avp(s:caller_vpabx)\1/'); xlog("$ru $tu\n"); } } } onreply_route[1] { if (isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1] { if (t_check_status("487")) { return; }; subst_uri('/^sip:(.*)@(.*)$/sip:100.\1@<local_proxy>/'); rewritehostport("<local_proxy>:5061"); append_branch(); t_relay(); } ---------------------------------------------------------------------- smieszne, muzyka, pilka, sexy, kibice, kino, ciekawe, extreme, kabaret http://link.interia.pl/f19d4 - najlepsze filmy w intermecie

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RE: [Users] DNS queries and TLS
by Papadopoulos Georgios 15 Dec '06

15 Dec '06

Hi Klaus, Here are the packets. The ngrep of the DNS lookup does not show much. I can only see the client IP (in reverse order). client: 213.5.17.236 Openser: 213.5.43.134, 213.5.8.5 DNS: 213.5.41.8 thank you U 2006/12/14 12:00:09.612841 213.5.17.236:5060 -> 213.5.43.134:5060 INVITE sip:2116872933@i-call.gr;user=phone SIP/2.0. Via: SIP/2.0/UDP 213.5.17.236:5060;branch=z9hG4bK-nykmi2fjal1m;rport. From: "George Papadopoulos" <sip:geop@i-call.gr>;tag=pt3u7ggbqm. To: <sip:2116872933@i-call.gr;user=phone>. Call-ID: 3c28be1c1adb-0vv5406jgtfv@213-5-17-236. CSeq: 2 INVITE. Max-Forwards: 70. Contact: <sip:geop@213.5.17.236:5060;line=9yt55o5b>. P-Key-Flags: keys="3". User-Agent: snom190-3.56w. Accept-Language: en. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. Supported: timer, 100rel, replaces. Session-Expires: 3600. Content-Type: application/sdp. Content-Length: 268. . v=0. o=root 886017132 886017132 IN IP4 213.5.17.236. s=call. c=IN IP4 213.5.17.236. t=0 0. m=audio 10190 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. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2006/12/14 12:00:09.614000 213.5.8.5:50120 -> 213.5.41.8:53 .............236.17.5.213.in-addr.arpa..... # U 2006/12/14 12:00:09.618660 213.5.41.8:53 -> 213.5.8.5:50120 .............236.17.5.213.in-addr.arpa.............*0.7.ns0.altectelecom s.gr. hostmaster.;w.e...p.... ..:...Q. # U 2006/12/14 12:00:09.619325 213.5.43.134:5060 -> 213.5.17.236:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 213.5.17.236:5060;branch=z9hG4bK-nykmi2fjal1m;rport=5060. From: "George Papadopoulos" <sip:geop@i-call.gr>;tag=pt3u7ggbqm. To: <sip:2116872933@i-call.gr;user=phone>. Call-ID: 3c28be1c1adb-0vv5406jgtfv@213-5-17-236. CSeq: 2 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. . > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Friday, December 15, 2006 12:42 PM > To: Papadopoulos Georgios > Cc: Bogdan-Andrei Iancu; users(a)openser.org > Subject: Re: [Users] DNS queries and TLS > > Can you also post the INVITE which is processd and the ngrep > of the DNS lookup (which domain is lookup up) > > regards > klaus > > Papadopoulos Georgios wrote: > > Hi Bogdan, > > > > I am not using any nat test functions, at least at the part of the > > script that handles INVITEs. I am using allow_trusted() in > caching mode. > > But if allow_trusted was doing the DNS query then I should > also see it > > after the first INVITE and before the "407 Proxy Auth > Reqd". I am also > > using lcr module. So, is it possible that load_gws() or > next_gw() is > > responsible for the DNS query? > > > > Regards, > > > > George > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Thursday, December 14, 2006 7:14 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] DNS queries and TLS > >> > >> Hi George, > >> > >> > >> TLS = Thread Local Storage > >> > >> > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hello, > >>> > >>> I am having some performance issues with Openser and I tend > >> to believe > >>> they are related with DNS. So I am trying to figure out > >> when and why > >>> Openser is doing DNS queries. Doing ngrep on port 53 I > >> realized that > >>> right before sending out the "100 trying -- your call is > >> important to > >>> us" message, Openser is doing a reverse DNS lookup for > the IP where > >>> the INVITE came from. So the sequence is something like: > >>> client Openser DNS > >>> |---INVITE------------------------>| > >>> |<---407 Proxy Auth Reqd---| > >>> |---ACK--------------------------->| > >>> |---INVITE------------------------>| > >>> |---client > IP?-------->| > >>> > >> |<-------------------------| > >>> |<--------------------100 trying---| > >>> > >>> > >>> I am using Openser-1.1.0-notls and in my script I have dns=no > >>> rev_dns=no So first question is whether this DNS query is > necessary > >>> and how I could avoid it. > >> the configuration options are ok (as time you do not use > the command > >> line -r or -R). I would say the rev dns query is not > triggered by the > >> t_relay() (actually the params control what DNS queries should be > >> done when testing the "received" VIA param) - I have > tested and I see > >> no query before 100 trying, so it should be ok. > >> > >> maybe you are using some nat test functions (like > >> client_nat_test) or any other script functions that mask a dns > >> query...can you check on this? > >> > >>> > >>> What is furthermore confusing is that I have a test > system with the > >>> same Openser version and same script, where this DNS query is not > >>> happening. Looking into the production system I found the > following: > >>> ser2:/usr/local/openser-1.1.0-notls/sbin# ldd openser > >>> linux-gate.so.1 => (0xffffe000) > >>> libdl.so.2 => /lib/tls/libdl.so.2 (0x55571000) > >>> libresolv.so.2 => /lib/tls/libresolv.so.2 (0x55574000) > >>> libc.so.6 => /lib/tls/libc.so.6 (0x55587000) > >>> /lib/ld-linux.so.2 => /lib/ld-linux.so.2 > >> (0x55555000) whereas > >>> the test system shows: > >>> sertest:/usr/local/openser-1.1.0-notls/sbin# ldd openser > >>> libdl.so.2 => /lib/libdl.so.2 (0x7002c000) > >>> libresolv.so.2 => /lib/libresolv.so.2 (0x70040000) > >>> libc.so.6 => /lib/libc.so.6 (0x70064000) > >>> /lib/ld-linux.so.2 (0x70000000) So the two > systems link to > >>> different libresolv.so libraries. Is the > tls/libresolve.so that is > >>> responsible for the DNS query? Given that in both cases I > >> am using the > >>> notls version of Openser 1.1, why is there a difference > between the > >>> two? > >> the "tls" frm /lib/tls comes from "Thread Local Storage" and there > >> are libraries implementations for thread env. It has nothing to do > >> with TLS (Transport Layer Security) > >> > >> regards, > >> bogdan > >> > >>> > >>> thank you for any help > >>> > >>> 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 > >>> > >>> > >> > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > -- > Klaus Darilion > nic.at > >

1 0

RE: [Users] DNS queries and TLS
by Papadopoulos Georgios 15 Dec '06

15 Dec '06

Hi Bogdan, I am not using any nat test functions, at least at the part of the script that handles INVITEs. I am using allow_trusted() in caching mode. But if allow_trusted was doing the DNS query then I should also see it after the first INVITE and before the "407 Proxy Auth Reqd". I am also using lcr module. So, is it possible that load_gws() or next_gw() is responsible for the DNS query? Regards, George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Thursday, December 14, 2006 7:14 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] DNS queries and TLS > > Hi George, > > > TLS = Thread Local Storage > > > > Papadopoulos Georgios wrote: > > > Hello, > > > > I am having some performance issues with Openser and I tend > to believe > > they are related with DNS. So I am trying to figure out > when and why > > Openser is doing DNS queries. Doing ngrep on port 53 I > realized that > > right before sending out the "100 trying -- your call is > important to > > us" message, Openser is doing a reverse DNS lookup for the IP where > > the INVITE came from. So the sequence is something like: > > client Openser DNS > > |---INVITE------------------------>| > > |<---407 Proxy Auth Reqd---| > > |---ACK--------------------------->| > > |---INVITE------------------------>| > > |---client IP?-------->| > > > |<-------------------------| > > |<--------------------100 trying---| > > > > > > I am using Openser-1.1.0-notls and in my script I have dns=no > > rev_dns=no So first question is whether this DNS query is necessary > > and how I could avoid it. > > the configuration options are ok (as time you do not use the > command line -r or -R). I would say the rev dns query is not > triggered by the > t_relay() (actually the params control what DNS queries > should be done when testing the "received" VIA param) - I > have tested and I see no query before 100 trying, so it should be ok. > > maybe you are using some nat test functions (like > client_nat_test) or any other script functions that mask a > dns query...can you check on this? > > > > > What is furthermore confusing is that I have a test system with the > > same Openser version and same script, where this DNS query is not > > happening. Looking into the production system I found the following: > > ser2:/usr/local/openser-1.1.0-notls/sbin# ldd openser > > linux-gate.so.1 => (0xffffe000) > > libdl.so.2 => /lib/tls/libdl.so.2 (0x55571000) > > libresolv.so.2 => /lib/tls/libresolv.so.2 (0x55574000) > > libc.so.6 => /lib/tls/libc.so.6 (0x55587000) > > /lib/ld-linux.so.2 => /lib/ld-linux.so.2 > (0x55555000) whereas > > the test system shows: > > sertest:/usr/local/openser-1.1.0-notls/sbin# ldd openser > > libdl.so.2 => /lib/libdl.so.2 (0x7002c000) > > libresolv.so.2 => /lib/libresolv.so.2 (0x70040000) > > libc.so.6 => /lib/libc.so.6 (0x70064000) > > /lib/ld-linux.so.2 (0x70000000) So the two systems link to > > different libresolv.so libraries. Is the tls/libresolve.so that is > > responsible for the DNS query? Given that in both cases I > am using the > > notls version of Openser 1.1, why is there a difference between the > > two? > > the "tls" frm /lib/tls comes from "Thread Local Storage" and > there are libraries implementations for thread env. It has > nothing to do with TLS (Transport Layer Security) > > regards, > bogdan > > > > > thank you for any help > > > > 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 > > > > > >

2 1

[Users] Sip Message
by Howard Tang 15 Dec '06

15 Dec '06

Hi, Can someone tell me how i can see the SIP message going through in openser please? I want to debug why openser return "Address Incomplete". Regards, Howard

4 3

[Serusers] click2dial issue
by hiSIPatH 15 Dec '06

15 Dec '06

Hi all, i am trying to use click 2 dial in phonebook (serweb). But unfortunately i am not able to get this to work. I always get the following error in the popup window: 407 Proxy Authentication Required The sip trace gives the following ( i have sniffed eth0 and lo) on the eth0 (with public ip@ the dns query and response) No. Time Source Destination Protocol Info 16 3.618415 89.xxx.xxx.218 89.xxx.xxx.129 DNS Standard query SRV _sip._udp.yyyy.zzz No. Time Source Destination Protocol Info 17 3.618868 89.xxx.xxx.129 89.xxx.xxx.218 DNS Standard query response SRV 200 0 5060 proxy.yyyy.zzz on my loopback interface No. Time Source Destination Protocol Info 1 0.000000 89.xxx.xxx.218 89.xxx.xxx.218 SIP/SDP Request: INVITE sip:firstname.lastname@yyyy.zzz, with session description No. Time Source Destination Protocol Info 2 0.000251 89.xxx.xxx.218 89.xxx.xxx.218 SIP Status: 407 Proxy Authentication Required No. Time Source Destination Protocol Info 3 0.000332 89.xxx.xxx.218 89.xxx.xxx.218 SIP Request: ACK sip:firstname.lastname@yyyy.zzz i can see several INVITE-407-ACK but no INVITE with proxy-authorization from controller(a)yyyy.zzz have i forgotten to configure something. thx in advance

2 2

[Serusers] transfer to a speeddial number
by jorain 15 Dec '06

15 Dec '06

Hi all, How to get the referred-by value from sip message header? I need it to transfer the call by dialing speeddial number which was stored in speeddial table. Because it will take the original caller (first caller) as the username instead of the second party who did the transfer. For example, speeddial table: ----------------------------------------------------------- username sd_number new_uri 55555555 123 55555550(a)205.5.5.5 33333333 123 33333330(a)205.5.5.5 When i call from 33333333 to 55555555, then 55555555 transfer the call to 123, by right it will call 55555550(a)205.5.5.5 , but now my system call 33333330(a)205.5.5.5 How can i solve it? Thanks. Regards, jorain

1 0

[Serusers] SER + Asterisk - Help needed please
by Mr Raams 15 Dec '06

15 Dec '06

Hello All, I am having problems with my SER + Asterisk combination. I have SER and Asterisk working well on their own, however my issue is as follows. I have asterisk running in realtime with a users table of values accountcode & secret for its authentication. I have the SER subscriber table as normal. I want to have the same SIPID in both tables and the secret=password. However when I register to SER as the SIPID and with the password register is ok, local calls are ok, but when I call asterisk it wont go, it times out after some time. -Note: This is with the password set in the asterisk table as secret (remember this works fine if i register direct to asterisk) The funny part is if I remove the secret (password) from the asterisk account, leaving only the accountcode, but leave the username & password in SER the PSTN call works fine. SER passes the call to Asterisk, and Asterisk will terminate the call. I really need SER to pass the correct authentication details on to Asterisk. I have attached the ser.cfg for you to view and comment. Thank you, Send instant messages to your online friends http://au.messenger.yahoo.com

1 0

[Serusers] click to dial
by jorain 15 Dec '06

15 Dec '06

Hi all, Is it possible to enable the conference using click-to-dial script? Thanks Regards, jorain

1 0

[Users] Duplicate INVITEs
by Mahesh Paolini-Subramanya 14 Dec '06

14 Dec '06

We're seeing sporadic situations where fones a) send an INVITE b) ignore the 100-Trying response, and instead c) send another INVITE with the same sequence number. The question is, What should happen here? Currently, we send back a 500 response (server error). Is this correct? (i think not, cause it invariably causes the fone to go fast busy). Is this some other response that should occur? poring over 3261 resulted in a headache, and no additional clarity... -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

3 4

[Users] Call based log
by ibrahim tunali 14 Dec '06

14 Dec '06

Hi everone, I want to trace previous or completed SIP dialogs so i need to keep log every sip activate to separate log files which name is <sip_callid>.log . Do you suggest any how to or tool for this issue. Best regards, ibrahim tunali

4 3

[Users] DNS queries and TLS
by Papadopoulos Georgios 14 Dec '06

14 Dec '06

Hello, I am having some performance issues with Openser and I tend to believe they are related with DNS. So I am trying to figure out when and why Openser is doing DNS queries. Doing ngrep on port 53 I realized that right before sending out the "100 trying -- your call is important to us" message, Openser is doing a reverse DNS lookup for the IP where the INVITE came from. So the sequence is something like: client Openser DNS |---INVITE------------------------>| |<---407 Proxy Auth Reqd---| |---ACK--------------------------->| |---INVITE------------------------>| |---client IP?-------->| |<-------------------------| |<--------------------100 trying---| I am using Openser-1.1.0-notls and in my script I have dns=no rev_dns=no So first question is whether this DNS query is necessary and how I could avoid it. What is furthermore confusing is that I have a test system with the same Openser version and same script, where this DNS query is not happening. Looking into the production system I found the following: ser2:/usr/local/openser-1.1.0-notls/sbin# ldd openser linux-gate.so.1 => (0xffffe000) libdl.so.2 => /lib/tls/libdl.so.2 (0x55571000) libresolv.so.2 => /lib/tls/libresolv.so.2 (0x55574000) libc.so.6 => /lib/tls/libc.so.6 (0x55587000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x55555000) whereas the test system shows: sertest:/usr/local/openser-1.1.0-notls/sbin# ldd openser libdl.so.2 => /lib/libdl.so.2 (0x7002c000) libresolv.so.2 => /lib/libresolv.so.2 (0x70040000) libc.so.6 => /lib/libc.so.6 (0x70064000) /lib/ld-linux.so.2 (0x70000000) So the two systems link to different libresolv.so libraries. Is the tls/libresolve.so that is responsible for the DNS query? Given that in both cases I am using the notls version of Openser 1.1, why is there a difference between the two? thank you for any help 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] Request for Discussion: MI and FIFO
by Bogdan-Andrei Iancu 14 Dec '06

14 Dec '06

Hi everybody, with the adding of asynchronous reply feature in MI and MI_FIFO implementation, the transition to MI was completed - all former FIFO functions (old implementation) were ported to MI. Also all FIFO capabilities were also ported to the new MI_FIFO module. so, currently there are two FIFO implementations: - the old one, directly in core. - the new one, based on MI interface. the old implementation is obsolete (all external management will be done via MI) , so, at some point, it will be removed from OpenSER in order to maintain consistency and avoid confusion between the two implementations (as they are equivalent from user point of view - at the moment, at least). my question is when to drop the old implementation? my personal preference will be do doing ASAP as : 1) no functionality is lost 2) avoid confusion among users and the explanation overhead 3) it will force the testing of the new FIFO and MI implementation, which will shorter the way to a stable form. I would like to know also some other opinions. regards, bogdan

4 4

[Serusers] send IM before operation in next shell script ends...
by Webnutzer 14 Dec '06

14 Dec '06

Hi ! I have an if block in my ser.cfg. in this block at first I send an IM with an "thanks for input", after that a shell script should start with some operations. SER sends the IM, but after the operations in shell script ends... How can I send the IM , before the shell script with operations finishs or SER start with operations in the shell script ? here is the code: if (method == "Message" && search("[a-zA-Z0-9/.+ ]{1,}-*,[a-zA-Z0-9]{1,}")) { sl_send_reply("200", "OK"); exec_msg("cd /etc/ser/;sh zweiteEingabe"); # send the thanks... log(1, "zweiter Schritt\n"); exec_msg("cd /root/workspace/Check_Name/bin/;sh Check_Name_Start.sh"); # other operations log(1, "Daten uebergabe\n"); break; }; Thanks ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

1 0

Re: [Users] full SIP message in perl
by vallimamod abdullah 14 Dec '06

14 Dec '06

Hello, On Thu, 14 Dec 2006 09:31:22 +0100, Bastian Friedrich <bastian.friedrich(a)collax.com> wrote: > Hi > > On Wednesday 13 December 2006 20:12, Greg Fausak wrote: > > I can see where the sip message is exported to perl. > > Can perl modify the message and send it back? > > Yes, modification is possible. I did not implement too many functions > though; you have two means of modification: > * Use the one or two I wrote. The most important probably is > rewrite_ruri, just as you asked for. > * Use the functions that are exported by other modules (esp. > the "textops" module) for message modification > > [...] > A question: is it possible to change the sip message using Perl's regular expression power and send it back to openser ? That would be a killer feature as it will give me the ability to make complex routing without developping a new module in c ! -- Regards, -vma .

2 1

[Users] Pike module
by Stefano Capitanio 14 Dec '06

14 Dec '06

Hi, i want to use pike module to avoid brute-force attack on Register. are the values of the parameters expressed in seconds? it seems that there is a problem because i've set: modparam("pike", "sampling_time_unit", 30) modparam("pike", "reqs_density_per_unit", 10) modparam("pike", "remove_latency", 600) and the routing logic is: if (method=="REGISTER") { # Checks brute-force attacks if (!pike_check_req()) { sl_send_reply("403", "Too many attempts!"); break; }; if (!www_authorize("caspur.it", "subscriber")) { www_challenge("caspur.it", "0"); break; }; save("location"); break; }; but when I reach the limit it blocks the requests only for less than 1 minute... where is the mistake?any idea? thanks, Stefano

2 3

[Users] full SIP message in perl
by Klaus Darilion 14 Dec '06

14 Dec '06

Hi! m->getFullHeader returns only the headers. How can I access the whole message (headers + all bodies)? regards klaus -- Klaus Darilion nic.at

4 6

[Serusers] Daily snapshots
by Jan Janak 13 Dec '06

13 Dec '06

Hello, Daily snapshots of SER, SERctl, SERweb, and RTPproxy are now available on the FTP server, for more information see: http://www.iptel.org/daily_snapshots_of_software Jan.

1 0

[Serusers] Voice Mail
by Jesús Terán 13 Dec '06

13 Dec '06

Hello I am using ser with asterisk as voicemal, i have the solution when the number is busy and when the number is not registered, but my question is how can i do when no body answer after 3 rings, because right now the mailbox starts after one minute that the phone is ringing. Thanks in advance Jesús Terán

2 1

[Users] perl problems
by Klaus Darilion 13 Dec '06

13 Dec '06

Hi! I try the new perl module, but openser does not start: openser[28662]: perl: Setting lib path: '/home/darilion/software/openser-cvs/sip-server/modules/perl/lib/perl/OpenSER.pm' openser[28662]: Error loading perl file "/etc/openser/myperl.pl". openser[28662]: init_mod(): Error while initializing module perl my perl related config: loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/perl.so" ... modparam("perl", "filename", "/etc/openser/myperl.pl") modparam("perl", "modpath", "/home/darilion/software/openser-cvs/sip-server/modules/perl/lib/perl/OpenSER.pm") /etc/openser/myperl.pl is identical to modules/perl/doc/samples/messagedump.pl Any hints what the problem is? thanks klaus -- Klaus Darilion nic.at

5 9

[Serusers] XML-RPC - how to make working on 0.10?
by Libor Chocholaty 13 Dec '06

13 Dec '06

Hi all, I am struggling with xmlrpc module. It is loaded already but ser reports that do not know create_via() during parsing ser.cfg. Could anybody show me the right way to configure xmlrpc.so properly? BR, Libor

2 1

[Serusers] About rewriteuser()
by chungyu 13 Dec '06

13 Dec '06

Hi I have a field called "phone_number" in subscriber,and would append to rewriteuser(). But rewriteuser() type is string. So have a problem. problem: I used AVpops. modparam("avpops","avp_aliases","phone=i:660") modparam("avpops","db_scheme","phone_scheme:table=subscriber;value_col=phone_number;value_type=string") avp_db_load("$ruri","$phone/$phone_scheme"); rewriteuser("$phone"); result is error. because rewriteuser(string). and i want if value of "$phone" is 4761 the result is 4761@domain How should I use rewriteuser() Thank you!! Chungyu 2006/12/13

1 1

[Users] OpenSER.pm perl library installation
by Klaus Darilion 13 Dec '06

13 Dec '06

Hi! I wonder how the OpenSER.pm is handled during installation. Is it copied into a certain directory (e.g. /usr/lib/openser/modules/) or do I have to do it manually? regards klaus -- Klaus Darilion nic.at

2 1

[Users] Re: check if an AVP is blank
by Daniel-Constantin Mierla 13 Dec '06

13 Dec '06

Hello, see is_avp_set() from avpops modules: http://www.openser.org/docs/modules/1.2.x/avpops.html#AEN594 Cheers, Daniel PS. please start a new thread on mailing list if you have questions not related to the subject of the message you reply On 12/13/06 14:29, Fernando Freitas(Brastel.com.br) wrote: > Hi, > > I need to check if an AVP is blank, how can I do it? > > This is what I have in my script: > > avp_write("tU","$avp(tmp)"); > > > I need to check if the content of this tmp AVP is blank, how to check this?? > > What I need is to check if the TO Username from SIP URI is blank if so, I > need to push the To URI into the To URI Username. > > Thanks, > > Fernando > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Users] Tutorial: OpenSER modular installation
by Daniel-Constantin Mierla 13 Dec '06

13 Dec '06

Hello, this might be useful for many people around, and the input from others would improve the content ... On the OpenSER Web Forum was posted a small tutorial about OpenSER modular installation, courtesy of Alistair Doswald: http://www.voipuser.org/forum_topic_8040.html Cheers, Daniel

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[Serusers] Starting SER in fork=no mode.
by Atle Samuelsen 13 Dec '06

13 Dec '06

Hi Guys, I was wondering if there is a technical reason why ser does not startup in fork=no mode if you only listen to TCP. - Atle

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[Users] Question about dispatcher module
by Jose Gil Navarrete 13 Dec '06

13 Dec '06

Hi, I try to configure openser with the dispatcher module like only ponit of contact and load balancer, it is: Register UAC ---Register--> LB ---Register--> proxy UAC <--200 ok----- LB <--200 ok----- proxy INVITE UAC1 ---Invite--> LB ---Invite--> proxy UAC2 <--Invite--- LB <--Invite--- proxy My load balancer works ok, my question is: May I have a problem with the transactions in the load balacer? In my proxy i use openser, then in my openser.cfg I make t_relay("IPLoadBalancer:5060") and the openser.cfg in LB: if ( !is_method("REGISTER") ){ record_route(); }; if (loose_route()) { if (search("Path")) { route(1); }; if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); exit; }; route(1); }; if (!uri==myself) { route(1); } if ( !is_method("REGISTER") ){ if (search("Path")) { route(1); }; if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); }; }; ds_select_domain("2", "0"); t_on_failure("1"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Thanks in advance. --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Users] Resource List Server/XCAP Clients
by Reinhold Buchinger 13 Dec '06

13 Dec '06

Hi, I try to build a presence infrastructure based on OpenSER. Do you know any free RLS and XCAP Clients I can use for this purpose? I hope this is not too much off-topic. Thanks in advance! regards, Reinhold

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[Serusers] IP Based Auth
by Arun Kumar 13 Dec '06

13 Dec '06

HI, I've running ser with mysql and radius. My user are stored in mysql db. my users are able to do the auth againg radius. I'm looking how to setup ip based auth. any kind of doc or link. thanks arun

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[Serusers] reinvite problem...
by Natambu Obleton 12 Dec '06

12 Dec '06

Ok. I have SER working with mediaproxy and it rewrites the requests to goto the Mediaproxy server as it should, but after a call has been up for a bit.. it gets another invite from my PSTN gateway and when this invite comes through it doesnt rewrite the media addresses why? My config: http://laplata.fasttrackcomm.net/ser_cfg.txthttp://laplata.fasttrackcomm.net /ser_cfg.txt <http://laplata.fasttrackcomm.net/ser_cfg.txthttp:/laplata.fasttrackcomm.net /ser_cfg.txt> My transaction log: http://laplata.fasttrackcomm.net/translog.txt About ¾s of the way down you will see where 206.123.213.84 sends another invite for the call and SER doesnt change the media information on it. Thanks for your help. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Users] New module: perl
by Bastian Friedrich 12 Dec '06

12 Dec '06

Hi everybody, since last friday, OpenSER HEAD has a new module, called perl. As the name suggests, it incorporates a Perl interpreter within OpenSER. What can you do with it? ======================== The perl module lets you request information from the SIP message currently active, lets you make decisions based on that information and finally modify the SIP message. In other words, you can do almost everything you could have done in the OpenSER configuration file, plus much, much more. How do I use it? ================ The perl module should be automatically compiled in current CVS checkouts of the HEAD branch, as long as you have a fairly recent Perl version (including possibly necessary development packages) installed on your system. Load it in your OpenSER configuration just like all the other modules. Set the two module parameters "filename" and "modpath"; the first one points to your perl script (samples are included in the doc directory), the second points to the path where the Perl packages reside (they currently are not "installed" anywhere automatically. Feel free to do that yourself). In the configuration file, you may then call the functions defined in your Perl script with the "perl_exec" and "perl_exec_simple" functions. The samples and the README file should tell you more about it. What should the Perl script look like? ====================================== Please take a look at the doc/samples directory located in the perl module directory. They should give you an introduction of how such a script could look like. The full API is described in the README file. The module documentation on the OpenSER website is not 100% apt as of now, but should be updated during the next days. Great! And now? =============== Please feel free to test the perl module. Although it currently runs fine for me, deployment in production environments is not recommended. Tell me about your experiences and any bugs you found. If you write a test script, I'd be thankfull if you send it to me for inclusion as a sample. If you have any questions or remarks, please send a mail to the users or developer mailing lists (depending on the focus), or directly to me. Thx for your attention, Regards, Bastian -- Collax GmbH . Burkheimer Straße 3 . 79111 Freiburg . Germany p: +49 (0) 761-45684-24 f: +49 (0) 761-45684-10 www.collax.com \ Things you never wanted to know: \ In Minnesota it is illegal to cross state lines with a duck on your \ head.

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[Users] Presence Publish/Subscribe Timeout
by christian l 12 Dec '06

12 Dec '06

Hello, we want to use the presence module of openser in combination with eyebeam pa mode. In order to save bandwith, we want the clients to subscribe in an intervall of 1 hour. At the same time, we want to detect almost instantly when I client timeouts, i.e. has a network failure. Is there any way I can detect client timeouts rather fast other than reducing the max_expires value, which also affects the SUBSCRIBE timeout and therefore increasing the network utiliztion again? openser version is 1.2.0-dev12-notls , eyebeam version is 1.5.10.2 build 33793 Thanks in advance for your help Schnell und einfach ohne Anschlusswechsel zur Lycos DSL Flatrate wechseln und 3 Monate kostenlos ab effektiven 5,21 EUR pro Monat im ersten Jahr surfen. http://www.lycos.de/startseite/online/dsl/index.html?prod=DSL&trackingID=em…

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[Users] sent notify after registration
by Pavel A. Siderov 12 Dec '06

12 Dec '06

Hi guys, I'm trying to generate (read data from custom database) and send a notify message to user agent when it registers and there is a voicemail left. I don't want to use an external script due to some reasons. I had a look on the msilo and uac modules but they don't seem to be suitable for the purpose. Is it possible with some of the modules or currently I need to use an external script? Thanks in advance, Pavel

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[Serusers] ser 0.9.6 # mediaproxy # one-way audio
by davor jovanovic 12 Dec '06

12 Dec '06

Hi all, I’m using ser 0.9.6 with mediaproxy and latest version of XLite client. I have a problem with one-way audio when calling client on another domain (not served by our SER). When Xlite uses public address (not behind NAT) I can't receive audio from other side, but when I connect that same XLite client by way of dial-up or ADSL everything is OK. I suppose that in second case (ADSL) mediaproxy is included in the process and that is way call succeeds. I don't know what is a problem when client isn't behind NAT, but it has public IP address. All combinations of calls are working with no problem when all users are using only our SER. I'm using configuration similar to that in SER Getting started (NAT part is same), but if needed I can email it. If anyone had similar problem, please help, a would appreciated it! Thanks /davor

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[Users] Problem with $dlg_count
by Leonid Fainshtein 12 Dec '06

12 Dec '06

Thank you, Daniel. Your recommendation helped. Leonid ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine %tn%

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[Serusers] Idea for concurrent calls statistics?
by Tobias Lindgren 12 Dec '06

12 Dec '06

Hi all, I'd like to somehow count concurrent outgoing calls through my SER from my customers, based on the source ip-address in the SIP-request. I'm running with the TM-module. Does anyone have an idea on how this could be done? Br, Tobias

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[Users] openserctl fifo
by Klaus Darilion 12 Dec '06

12 Dec '06

Hi! I can nowhere found the setting for the fifo use inside openserctl. The default is /tmp/openser_fifo - where can I change this? thanks klaus -- Klaus Darilion nic.at

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Re: [Serusers] Ser accounting question?
by Greger V. Teigre 12 Dec '06

12 Dec '06

Copy the list. Sending private emails just make them add to the end of my big inbox. I don't know where the service type comes from. That's where you need to define it. g-) Lokesh Kumar wrote: > > Where I can define service type? If this not from ser then must be > conflicting from radius dictionary. Can you suggest something.? > > > > Thanks > > > > Lokesh > > > > ------------------------------------------------------------------------ > > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Thursday, December 07, 2006 10:11 AM > *To:* Lokesh Kumar > *Cc:* serusers(a)iptel.org > *Subject:* Re: [Serusers] Ser accounting question? > > > > The service-type is set by the sender of the accounting packet. I'm > not aware of such a service-type in ser. > g-) > > Lokesh Kumar wrote: > > Hello List, > > > > Below mentioned is my accounting packet, I would like to know what is > IAPP-AP-Check, what this check do , and where its configuration > present. Can I change Service-Type ? > > > > Wed Dec 6 20:08:23 2006 > > Acct-Status-Type = Stop > > Service-Type = IAPP-AP-Check > > Sip-Response-Code = 200 > > Sip-Method = 8 > > User-Name = "69021(a)xxx.yyy.pt" <mailto:69021@xxx.yyy.pt> > > Calling-Station-Id = "sip:69021@xxx.yyy.pt" > <mailto:sip:69021@xxx.yyy.pt> > > Called-Station-Id = "sip:30612@xxx.yyy.pt" > <mailto:sip:30612@xxx.yyy.pt> > > Sip-Translated-Request-URI = "sip:30612@212.13.xx.yy:5060" > <mailto:sip:30612@212.13.xx.yy:5060> > > Acct-Session-Id = > "D4808CE2-1BA9-4572-B82B-5A0BCF6AB7CB(a)192.168.0.47" > <mailto:D4808CE2-1BA9-4572-B82B-5A0BCF6AB7CB@192.168.0.47> > > Sip-To-Tag = "86949D74-2699" > > Sip-From-Tag = "67434843439" > > Sip-Cseq = "3" > > NAS-Port = 5060 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 127.0.0.1 > > Client-IP-Address = 127.0.0.1 > > Acct-Unique-Session-Id = "c6adb5da735b1150" > > Timestamp = 1165435703 > > > > Thanks > > > > Lokesh > > > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006 > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006 > > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006 >

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Re: [Users] SIP message being sent twice
by Klaus Darilion 12 Dec '06

12 Dec '06

Fernando Freitas wrote: > Hi all, > > I am having a situation that every SIP message is being sent twice when > reaching openser. It seems that the message when hits proxy it is processed > twice. I double checked my cfg file, but didn't find anything. Please help > me. Maybe you add another branch in your openser config. Are they really the same (also the branch parameter in the Via header)? regards klaus > > Thanks, > > Fernando Freitas. > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- Klaus Darilion nic.at

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Re: Re: [Serusers] SIPSAK - wrong behaviou r??
by tzieleniewski 12 Dec '06

12 Dec '06

yes you are right:) but the stange situation is the following (version from svn): UA1 SIPProxy | | |INVITE 3 | |------------->| | | |407 3 | |<-------------| | | |ACK 3 | |------------->| | | |INVITE 4 | |------------->| | | |TRYING 4 | |<-------------| | | |INVITE 4 | |<-------------| | | |ACK 3 | |------------->| | | |INVITE 4 | |<-------------| | | after the end of dialog related to cseq 3 ua (sipsak) sends an invite (with cseq 4) to sipproxy then there is the provisional response trying and sipproxy sends invite further (actually the same ua) to sipsak and the reaction of sipsak is: ACK with with cseq 3 instead of OK or 4xx, this scenarion is repeated until sipproxy sends the 408 timeout with cseq 4 after wich ua (sipsak) resposes with ACK with cseq 4 what do you think about this?:) bests tomek ---- Wiadomość Oryginalna ---- Od: Michal Matyska <michal(a)iptel.org> Do: serusers(a)lists.iptel.org Data: 11 grudnia 2006 20:59 Temat: Re: [Serusers] SIPSAK - wrong behaviour?? > I was just curious, so I took look on the ngrep.... > > there is short jump in the description between points 2 and 3.... > 2a. sipsak sends ACK upon 407 reply, which is 100% correct > 2b. sipsak resends INVITE with correct credentials > 2c. ser forwards the INVITE to the sipsak @ port 5060 > 3. sipsak sends ACK (with the CSeq: 3, Via: headers look like > combination of original ACK and passed INVITE Vias), which you've found > inappropriate, right? Just to make sure we talk about the same ACK :-D > > Michal > > On Mon, 2006-12-11 at 19:12 +0100, Nils Ohlmeier wrote: > > Hi Tomasz, > > > > On Monday 11 December 2006 17:21, tzieleniewski wrote: > > > Hi ALL! > > > I am trying to run some tests under sipsak but I obtain strange result > > > I invoke sip stack with the following command: > > > sipsak -IU -vv -s sip:sipsak@scscf.ims.touk.pl -a sipsak -l 5061 -H > > > 192.168.0.129 The registration proces runs without any problem, user > > > registers and then sipsak sends invite to ser. And here comes the strange > > > thing > > > 1. sipsak sends invite to ser: > > > 2. Ser sends the proxy authentication required response 407 (from ngrep) > > > 3. sipsak receives the message andresponses with the ACK???? > > > Ive placed the ngrep from the sipsak socket in the attached file. > > > > that looks like a bug. I'll try to look into that later. > > It would be great if you could try if the same behavior is still present in > > the latest version from subversion (let me know if I should send you a > > tarball of that). And the output from sipsak with -vvv would be a great help. > > Just send that directly to me. > > > > Thanks > > Nils > > _______________________________________________ > > 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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[Users] SIP message being sent twice
by Fernando Freitas 12 Dec '06

12 Dec '06

Hi all, I am having a situation that every SIP message is being sent twice when reaching openser. It seems that the message when hits proxy it is processed twice. I double checked my cfg file, but didn't find anything. Please help me. Thanks, Fernando Freitas.

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[Serusers] Monitoring
by Peter Kuebler 11 Dec '06

11 Dec '06

Hello Hendrik, I was reading your post and wondering if you have made any progress on SER monitoring? Especially in regard to enhancing the snmpd to monitor the number of users and transactions? http://lists.iptel.org/pipermail/serusers/2006-January/026771.html Hello Hendrik, What about VRRP for failover ? SER1 (master) sip agents == || === Asterisk Farm==sip/pstn gateway SER2 (slave) Harry --- Hendrik Scholz <hendrik.scholz at freenet-ag.de <http://lists.iptel.org/mailman/listinfo/serusers> > a écrit : > Hi! > > hgaillac-sip at yahoo.fr <http://lists.iptel.org/mailman/listinfo/serusers> wrote: > > Yes I know about snmp module. > > There is no way to monitor SER vi snmp !? > > As others have noted SER does not directly support > SNMP but you can > easily enhance your snmpd to support it. > You add serctl fifo commands using the snmp exec > directive > to monitor the number of users, transactions stuff, > ... > The meminfo patch (now included) is handy and you > can directly pipe > serctl fifo meminfo into SNMP and create nifty > graphs. > > sipsak comes in handy, i.e. sending an OPTIONS > request and measuring > a) if a response comes back and b) the time > processing took. > > Much better would be monitoring INVITE/REGISTER > response behaviour since > that usually requires a whole lot more interaction > (radius, db, ..) > on SER side. > > Hendrik > Peter Kuebler Director Network Reliability tele: 703-636-1114 ext: 112 tele: 703-734-4112(alt) cell: 240-506-5703 SunRocket 8045 Leesburg Pike, Suite 300 Vienna, VA 22182 peter.kuebler(a)sunrocket.com SunRocket NOC 703-637-9970

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[Users] problem with forking...
by Frogger 11 Dec '06

11 Dec '06

I am seeing a problem forking more than 7. Any phones that are beyond the 7th device get 180 ringin but Openser cannot receive the invite to answer the call. Any thoughts on why I cannot increase the children? (BTW, "openserctl ps" shows the proper number of children. For example, when set to 14, "ps" shows 14 children. However any UA beyond 7 cannot answer the forked call.) Thanks, MC ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

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