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[Serusers] Telegram between 2 Clients
by Christian Mickler 13 Sep '06

13 Sep '06

Hi, how can i see the Telegram between 2 Clients by INVITE? Thanks!!!

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[Serusers] SERWEB: classkit.so file can not be loaded. Why? Please help..Please
by Ferianto siregar 13 Sep '06

13 Sep '06

Dear all, Thank you very much to Karel Kozlik and all of you who always kind-hearted to help me. Thank you very much. Karel, I have tried your suggestion in installing serweb. I have installed the DB-1.7.6 and classkit-0.4 . This is the installation message : Build complete. (It is safe to ignore warnings about tempnam and tmpnam). running: make INSTALL_ROOT="/var/tmp/pear-build-root/install-classkit-0.4" insta ll Installing shared extensions: /var/tmp/pear-build-root/install-classkit-0.4/ usr/lib/php/modules/ running: find "/var/tmp/pear-build-root/install-classkit-0.4" -ls 362402 8 drwxr-xr-x 3 root root 4096 Sep 12 16:54 /var/tmp/pear -build-root/install-classkit-0.4 362423 8 drwxr-xr-x 3 root root 4096 Sep 12 16:54 /var/tmp/pear -build-root/install-classkit-0.4/usr 362424 8 drwxr-xr-x 3 root root 4096 Sep 12 16:54 /var/tmp/pear -build-root/install-classkit-0.4/usr/lib 362425 8 drwxr-xr-x 3 root root 4096 Sep 12 16:54 /var/tmp/pear -build-root/install-classkit-0.4/usr/lib/php 362426 8 drwxr-xr-x 2 root root 4096 Sep 12 16:54 /var/tmp/pear -build-root/install-classkit-0.4/usr/lib/php/modules 362422 60 -rwxr-xr-x 1 root root 49885 Sep 12 16:54 /var/tmp/pear -build-root/install-classkit-0.4/usr/lib/php/modules/classkit.so Build process completed successfully Installing '/var/tmp/pear-build-root/install-classkit-0.4//usr/lib/php/modules/c lasskit.so' install ok: classkit 0.4 Segmentation fault But when I tried to run it in the browser, the error message. The message says : Function aggregate_methods() doesn't exists. Try install Classkit extension. http://pecl.php.net/package/classkit I have add "extension=classkit.so" in php.ini file. But why this message still appear? When I tried to look the error_log file in /var/log/httpd/, I got this error message: [Tue Sep 12 16:59:45 2006] [notice] Digest: generating secret for digest authentication ... [Tue Sep 12 16:59:45 2006] [notice] Digest: done [Tue Sep 12 16:59:45 2006] [notice] LDAP: Built with OpenLDAP LDAP SDK [Tue Sep 12 16:59:45 2006] [notice] LDAP: SSL support unavailable PHP Warning: PHP Startup: Unable to load dynamic library '/usr/lib/php/modules/classkit.so' - /usr/lib/php/modules/classkit.so: cannot open shared object file: Permission denied in Unknown on line 0 [Tue Sep 12 16:59:45 2006] [notice] mod_python: Creating 4 session mutexes based on 256 max processes and 0 max threads. [Tue Sep 12 16:59:45 2006] [notice] Apache/2.0.54 (Fedora) configured -- resuming normal operations [Tue Sep 12 17:00:04 2006] [error] [client 202.95.149.251] File does not exist: /var/www/serweb/html/favicon.ico [Tue Sep 12 17:00:12 2006] [error] [client 202.95.149.251] File does not exist: /var/www/serweb/html/favicon.ico Sep 12 11:56:07 sip kernel: audit(1158036967.878:16): avc: denied { read } for pid=2958 comm="httpd" name="classkit.so" dev=hdb6 ino=1499281 scontext=root:system_r:httpd_t:s0 tcontext=root:object_r:tmp_t:s0 tclass=file Sep 12 11:56:07 sip kernel: audit(1158036967.878:17): avc: denied { read } for pid=2958 comm="httpd" name="classkit.so" dev=hdb6 ino=1499281 scontext=root:system_r:httpd_t:s0 tcontext=root:object_r:tmp_t:s0 tclass=file Sep 12 11:56:11 sip kernel: audit(1158036971.482:18): avc: denied { read } for pid=2978 comm="httpd" name="classkit.so" dev=hdb6 ino=1499281 scontext=root:system_r:httpd_t:s0 tcontext=root:object_r:tmp_t:s0 tclass=file Sep 12 11:56:11 sip kernel: audit(1158036971.482:19): avc: denied { read } for pid=2978 comm="httpd" name="classkit.so" dev=hdb6 ino=1499281 scontext=root:system_r:httpd_t:s0 tcontext=root:object_r:tmp_t:s0 tclass=file As I read from the error message, the file classkit.so can not be loaded? Why? Would you mind give me a solution? Please.... Thank you very much, Regards, Ferianto Karel Kozlik <karel(a)iptel.org> wrote: Hi, > Thank you much for your help. I am very happy to read this relpy message > from you . Thanks you..thank you very much. > Karel, may I ask you again..Please.. > What you mean by pear library? > I am using Fedora Core 5 and I found these packages are installed in my sorry, I don't know Fedora > > Do you mean that pear library is *php-pear-1.4.6-2 *that is installed? I think so. Karel > or I have install pear again? > Please help me..Karel.. I too lack. Please... > > Thank you very much..Thank you. > > > Regards, > > Ferianto > > */Karel Kozlik /* wrote: > > Hi, > > > 1. I have downloaded from pear.net the DB packet. The version of the > > packet is DB-1.7.6. I have untar it in root directory. I can > found the > > file DB.php in it. But I am confused, where should I place this > > DB.php file? So I can solve this error? Would ypu mind telling > me..Please.. > > Sould I have internet connection to install it? Because I do not > have > > internet connection now. > > Please read the documentation! > http://pear.php.net/manual/en/installation.cli.php > > You should not untar the file in clasic way. You should use: > pear install .tgz > instead. You have to have 'pear' binary installed before, of course. I > don't know what OS are you useing. There should be some package > containing the 'pear' binary. If there is not, read the instructions > here: > http://pear.php.net/manual/en/installation.getting.php > > > > 2. Would you mind telling me what you mean by "fix include_path > in your > > php.ini". > > Would you mind give me an example? Please... > > This mean if you have your pear library in diferent directory (e.g. > /usr/share/foo/bar/pear) you have to append this directory to > 'include_path' variable within php.ini (php configuration file). Read > the comments within php.ini for more info. > > regards Karel > > > ------------------------------------------------------------------------ > Do you Yahoo!? > Everyone is raving about the all-new Yahoo! Mail. > > --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

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[Serusers] Newbie question
by Mario François Jauvin 13 Sep '06

13 Sep '06

Hi, I have installed SER and asterisk on a redhat enterprise 4 linux box. I would like to find out how to: 1. configure SER so that when people dial 9,2226665555 it will redirect the call to asterisk for processing and forwarding to PSTN 2. I heard mentioned several times how asterisk can provide voicemail for SER clients and I would like to know how to set this up so that user can login to their mailbox 3. I would also like to know how users can be notified of their message indicators. 4. In addition to the info above I would like to know if there are links that describe this in greater details. Thanks, Mario

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[Serusers] ser & sems
by Andy Thomas 13 Sep '06

13 Sep '06

Im running the latest ser with sems. Whenever I dial 958 I get the number reader announcement everytime. However when I dial an offline user or conference room, many times I dont get any speech. The call is connected but I just get silence. The server is on a public IP, so no nat issues. Firewall is correct, otherwise I wouldnt get the number reader working! All my fifo's are correct with the correct permissions etc, and my config file must be ok, as sometimes voicemail & conference announcements are played, sometimes not. Sems in debug mode displays this error everytime I get silence rather than the voicemail announcement- (4836) ERROR: stop (AmThread.cpp:99): pthread_detach failed with code EINVAL: thread already in detached state. Any ideas anyone?? Andy Thomas

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[Serusers] Handling 302 responses
by Roger Lewau 13 Sep '06

13 Sep '06

Hello list, I have searched this list for an answer to the 302 response handling problem but found no real solution. It seems no one actually has an aswer for this so I studied the RFC 3261 and found the following statement: The requesting client SHOULD retry the request at the new address(es) given by the contact header field. In my mind that statement is completely off the wall, it is not the requesting client that should be responsible for establishing the forwarded call, it never is in the rest of the telecom industry so why should it be the case for SIP? Instead, this should ofcourse be the responsibility of the forwarding client or the service provider on behalf of the forwarding client. But as the RFC is not crafted that way I need to find a way to handle call forwarding in a proper way so that the cost for the forwarded call ends up on the forwarding clients bill. As call forwarding is a basic requirement in any phone network there must be some one reading this list who has solved this issue that can share there insight. Any help on this issue is highly apreciated. Kind regards Roger Lewau

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[Serusers] SER as fake SBC...
by Natambu Obleton 13 Sep '06

13 Sep '06

I understand that SER is not an SBC, but I can see how it could maybe do it. So I started with hello-world.cfg from the onsip.org crowd and started playing. So I rewrite the contact address to send the address and port of the SER server and that works. I can make calls out, but when call come in they get routed to SER and it fails because my UA is not in the location database. # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { if (method=="REGISTER") { if (!subst('/Contact: <sip:(.*)@(.*)>/Contact: <sip:\1@206.123.214.55:5060>/ig')) { log("******************************SUBST FAILED ************************************\n"); }; # save("location"); }; if (!t_relay()) { sl_reply_error(); }; break; }; When I have 'save("location")' in my configuration, Then I don't get registered with my softswitch. So what does 'save("location")' do exactly? Beside put an entry in my location database? Thanks. In my setup I am not using NAT at all. UA, SER, and my softswitch all have public ip addresses. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Serusers] Need to work on subst().
by ravi reddy 13 Sep '06

13 Sep '06

Hi Ricardo , I too need the same process which you had done by subst(); function. two weeks back i had posted a mail to ser-users . regarding this issue. because one call-shop with the small sip-proxy using private ip-adress controls the remaing phones . and with one username i am giving access to all phones so that i can bill for one account only . but when a sip-phone registered to that callshop make call it forwards to my SER with contact id as eg: 12345(a)192.168.2.101 .so, the call comes in as it just requires authorise but when we hung the phone the "BYE" messages are not recieving by both end parties because SER is sending "BYE" messages to that contact-id that is private ip so call became "idle" for 30-40 seconds and after it shut down. so now i need to rewrite the contact part of that private ip-address and just use NAT ipadress of that sip-proxy(call-shop) i saw your subst(); funct. in your mail but i dont know where and how to use that,(I am not a programmer) So how you suggest me to do that for rewriting any private ip-adresss and replacing NAT-address so that the call will ends when it recieve "BYE" message . below is the trace of that: U 82.102.69.105:39871 -> 81.21.33.35:5060 INVITE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35 :5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102 ;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip: 192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23 . Content-Length: 361. Ricardo here i need to replace contact<sip:192.168.1.100> with above NAT ip address that is with 82.102.69.105:39871 how ??/ please assist me so that i can solve my problem. Thank You. Regards, Ravi.

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[Users] jabber
by Javier Ramirez 13 Sep '06

13 Sep '06

Anyone has experience with jabber module ? I have a jabber server working, and openser working but I can't connect both. please, send me an example of openser.cfg if you want. best regards Javier

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[Serusers] jabber
by Javier Ramirez 13 Sep '06

13 Sep '06

Anyone has experience with jabber module ? I have a jabber server working, and ser working but I can't connect both. please, send me an example of ser.cfg if you want. best regards Javier

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Re: [Users] 513 Message too big
by Greg Fausak 13 Sep '06

13 Sep '06

Bill, I think there is different syntax for this, I used to use something like: if(len_gt(max_len)) { } But, I don't check for max length any more. Operating system assembly/disassembly for packets bigger than the mtu is transparent to me, I often work with big packets. Also, the mtu is different depending on where you look. I think some of the gig-ethernets use jumbo packets, then there is the 1500 byte or so regular mtu, then there is also possible modifications to the mtu depending on what the local transport is (cable, dsl, frame relay, etc). I still check for max hops :-) -g On Sep 12, 2006, at 8:38 PM, Bill Zhang wrote: > OK, I solved the problem(though I don't understand how it solved) > by adding > alias=" sip.mydomain.com", thanks to my old experience with SER:). > > Bill > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] > On Behalf > Of Bill Zhang > Sent: Tuesday, September 12, 2006 2:18 PM > To: users(a)openser.org > Subject: RE: [Users] 513 Message too big > > It does not help me here:(. I will get "Too Many Hops" (SIP > response 483), > it's something that seems to get my SIP message stuck in the first 2 > routines. > > Bill > -----Original Message----- > From: Ka Lun Chan (KC) [mailto:kchan1028@gmail.com] > Sent: Tuesday, September 12, 2006 1:04 PM > To: 'Bill Zhang' > Subject: RE: [Users] 513 Message too big > > I have the exact problem. I fixed by changing it to the following > instead: > > if (msg:len >= max_len ) { > sl_send_reply("513", "Message too big"); > return; > }; > > KC > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] > On Behalf > Of Bill Zhang > Sent: Tuesday, September 12, 2006 12:59 PM > To: users(a)openser.org > Subject: [Users] 513 Message too big > > Hi, > > I've exhausted my knowledge on openser:(. Following is my > openser.cfg which > is pretty much copy directly from sample but I kept getting "513 > Message too > big" response from this server for SIP registration, I know the SIP > registration message size is ONLY 555 bytes, what could be wrong here? > > Thanks in Advance! > > # > # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > debug=4 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=yes # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > fork=no > log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > children=4 > fifo="/tmp/openser_fifo" > > # > # uncomment the following lines for TLS support > #disable_tls = 0 > #listen = tls:your_IP: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" > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > loadmodule "/usr/local/lib64/openser/modules/mysql.so" > loadmodule "/usr/local/lib64/openser/modules/sl.so" > loadmodule "/usr/local/lib64/openser/modules/tm.so" > loadmodule "/usr/local/lib64/openser/modules/rr.so" > loadmodule "/usr/local/lib64/openser/modules/maxfwd.so" > loadmodule "/usr/local/lib64/openser/modules/usrloc.so" > loadmodule "/usr/local/lib64/openser/modules/registrar.so" > loadmodule "/usr/local/lib64/openser/modules/textops.so" > loadmodule "/usr/local/lib64/openser/modules/auth.so" > loadmodule "/usr/local/lib64/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) > modparam("usrloc", "db_url", "mysql://openser:passwd@localhost/ > openser") > > # -- auth params -- > # Uncomment if you are using auth module > # > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url", "mysql://openser:passwd@localhost/ > openser") > > # -- 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; > }; > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!method=="REGISTER") > record_route(); > > # subsequent messages withing a dialog should take the > # path determined by record-routing > if (loose_route()) { > # mark routing logic in request > append_hf("P-hint: rr-enforced\r\n"); > route(1); > }; > > if (!uri==myself) { > # mark routing logic in request > append_hf("P-hint: outbound\r\n"); > # if you have some interdomain connections via TLS > #if(uri=~"@tls_domain1.net") { > # t_relay("tls:domain1.net"); > # exit; > #} else if(uri=~"@tls_domain2.net") { > # t_relay("tls:domain2.net"); > # exit; > #} > route(1); > }; > > # if the request is for other domain use UsrLoc > # (in case, it does not work, use the following command > # with proper names and addresses in it) > if (uri==myself) { > > if (method=="REGISTER") { > if (!www_authorize("sip.mydomain.com", > "subscriber")) { > www_challenge("sip.mydomain.com", > "0"); > exit; > }; > > save("location"); > exit; > }; > > lookup("aliases"); > if (!uri==myself) { > append_hf("P-hint: outbound alias\r\n"); > route(1); > }; > > # native SIP destinations are handled using our > USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > exit; > }; > append_hf("P-hint: usrloc applied\r\n"); > }; > > route(1); > } > > > route[1] { > # send it out now; use stateful forwarding as it works > reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > exit; > } > > Bill > > > _______________________________________________ > 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 > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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RE: [Users] 513 Message too big
by Bill Zhang 13 Sep '06

13 Sep '06

OK, I solved the problem(though I don't understand how it solved) by adding alias=" sip.mydomain.com", thanks to my old experience with SER:). Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, September 12, 2006 2:18 PM To: users(a)openser.org Subject: RE: [Users] 513 Message too big It does not help me here:(. I will get "Too Many Hops" (SIP response 483), it's something that seems to get my SIP message stuck in the first 2 routines. Bill -----Original Message----- From: Ka Lun Chan (KC) [mailto:kchan1028@gmail.com] Sent: Tuesday, September 12, 2006 1:04 PM To: 'Bill Zhang' Subject: RE: [Users] 513 Message too big I have the exact problem. I fixed by changing it to the following instead: if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; KC -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, September 12, 2006 12:59 PM To: users(a)openser.org Subject: [Users] 513 Message too big Hi, I've exhausted my knowledge on openser:(. Following is my openser.cfg which is pretty much copy directly from sample but I kept getting "513 Message too big" response from this server for SIP registration, I know the SIP registration message size is ONLY 555 bytes, what could be wrong here? Thanks in Advance! # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP: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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib64/openser/modules/mysql.so" loadmodule "/usr/local/lib64/openser/modules/sl.so" loadmodule "/usr/local/lib64/openser/modules/tm.so" loadmodule "/usr/local/lib64/openser/modules/rr.so" loadmodule "/usr/local/lib64/openser/modules/maxfwd.so" loadmodule "/usr/local/lib64/openser/modules/usrloc.so" loadmodule "/usr/local/lib64/openser/modules/registrar.so" loadmodule "/usr/local/lib64/openser/modules/textops.so" loadmodule "/usr/local/lib64/openser/modules/auth.so" loadmodule "/usr/local/lib64/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) modparam("usrloc", "db_url", "mysql://openser:passwd@localhost/openser") # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:passwd@localhost/openser") # -- 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; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("sip.mydomain.com", "subscriber")) { www_challenge("sip.mydomain.com", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Bill _______________________________________________ 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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RE: [Serusers] Redirect on Busy with 486 or Voicemail
by Tracy Lofton 13 Sep '06

13 Sep '06

Hi All, I am still a bit new to openser and I am trying to get the call to respond with busy if caller calls themself from their own UA. I created in the table grp - "voicemail" with corresponding usernames. While in the failure route, I am requesting to route calls to voicemail if the caller is calling a user that is busy or unavailable (but i don't want caller = self in this category), but I get this error for the first openser.cfg snippet: ( 0(31978) parse error (140,19-20): unknown command, missing loadmodule? Also, when I run the avp_check condition by iself without && my program seems to just skip over it and shoots right to the failure route and then redirects to voicemail when busy or unavailable, whether it is the caller calling from their own UA or not. ------------------------------------------------------------------------------------------- Route { ... if (method=="INVITE") { avp_write("$from", "i:34"); if (is_in_group("voicemail") && avp_check("i:34", "eq/$ruri/i")) { <- this is the parse error, line 140 sl_send_reply("486", "Busy Here"); exit; }; avp_delete("i:34"); ....... } ... # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if(method=="INVITE" && !isflagset(10)) { t_on_failure("1"); }; ...... failure_route[1] { if (!t_was_cancelled() | (t_check_status("486") ) { revert_uri(); rewritehostport("mydomain.com:5065"); append_branch(); xlog("L_INFO", "Temporarily Unavailable or Busy: Transfer to Voicemail"); setflag(10); route(1); } ---------------------------------------------------------------------------------------------end of first openser.cfg snippet Prior to the above, I have this openser.cfg working somewhat to my wishes by forwarding to the failure route any busy UA and returning a 486 (whether it as caller=callee or not) and for unavailable it would forward to voicemail. Also, I have a backup conditional statement (not shown) for users to dial *981 to reach a UA's voicemail directly to leave voicemail because I cannot get around the busy 486. ________________________________ failure_route[1] { if (t_check_status("486")) { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 486 Busy Here\n"); exit; } else if (!t_was_cancelled() ) { revert_uri(); rewritehostport("mydomain.com:5065"); append_branch(); xlog("L_INFO", "Temporarily Unavailable #1: Transfer to Voicemail"); setflag(10); route(1); } ---------------- end of 2nd openser.cfg snippet Actually, i need help with trying to get it to work right by if caller = self (as a separate condition) then return with 486 busy, and another condition if UA is unavailable or busy then forward to voicemail. I really would like a bit of help. I I hope this makes sense. Thanks, Tracy (The Practitioner) _________________________________________________________________ Check the weather nationwide with MSN Search: Try it now! http://search.msn.com/results.aspx?q=weather&FORM=WLMTAG

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[Users] Redirect on Busy with 486 or Voicemail
by Tracy Lofton 13 Sep '06

13 Sep '06

Hi All, I apologize for rewriting most of this, but I do not know how to continue a thread from a prior email. I am new. I fixed the parse error from a prior email request by adding the R-URI but I am still trying to get the call to respond with busy if caller calls themself from their own UA. I created in the table grp - "voicemail" with corresponding usernames. While in the failure route, I am requesting to route calls to voicemail if the caller is calling a user that is busy or unavailable (but i don't want caller = self in this category). When I run the avp_check condition, the program seems to just skip over it and shoots right to the failure route and then redirects to voicemail when busy or unavailable, whether it is the caller calling from their own UA or not. ------------------------------------------------------------------------------------------- Route { ..... if (method=="INVITE") { avp_write("$from", "i:34"); if (is_in_group("Request-URI", "voicemail") && avp_check("i:34", "eq/$ruri/i")) { sl_send_reply("486", "Busy Here"); exit; }; avp_delete("i:34"); ......... } ..... # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if(method=="INVITE" && !isflagset(10)) { t_on_failure("1"); }; ........ failure_route[1] { if (!t_was_cancelled() | (t_check_status("486") ) { revert_uri(); rewritehostport("mydomain.com:5065"); append_branch(); xlog("L_INFO", "Temporarily Unavailable or Busy: Transfer to Voicemail"); setflag(10); route(1); } ---------------------------------------------------------------------------------------------end of first openser.cfg snippet Prior to the above, I have this openser.cfg working somewhat to my wishes by forwarding to the failure route any busy UA and returning a 486 (whether it as caller=callee or not) and for unavailable it would forward to voicemail. Also, I have a backup conditional statement (not shown) for users to dial *981 to reach a UA's voicemail directly to leave voicemail because I cannot get around the busy 486. ________________________________ failure_route[1] { if (t_check_status("486")) { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 486 Busy Here\n"); exit; } else if (!t_was_cancelled() ) { revert_uri(); rewritehostport("mydomain.com:5065"); append_branch(); xlog("L_INFO", "Temporarily Unavailable #1: Transfer to Voicemail"); setflag(10); route(1); } ---------------- end of 2nd openser.cfg snippet Actually, i need help with trying to get it to work right by if caller = self (as a separate condition) then return with 486 busy, and another condition if UA is unavailable or busy then forward to voicemail. I really would like a bit of help. I I hope this makes sense. Thanks, Tracy (The Practitioner) _________________________________________________________________ Get real-time traffic reports with Windows Live Local Search http://local.live.com/default.aspx?v=2&cp=42.336065~-109.392273&style=r&lvl…

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RE: [Users] 513 Message too big
by Bill Zhang 12 Sep '06

12 Sep '06

It does not help me here:(. I will get "Too Many Hops" (SIP response 483), it's something that seems to get my SIP message stuck in the first 2 routines. Bill -----Original Message----- From: Ka Lun Chan (KC) [mailto:kchan1028@gmail.com] Sent: Tuesday, September 12, 2006 1:04 PM To: 'Bill Zhang' Subject: RE: [Users] 513 Message too big I have the exact problem. I fixed by changing it to the following instead: if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; KC -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, September 12, 2006 12:59 PM To: users(a)openser.org Subject: [Users] 513 Message too big Hi, I've exhausted my knowledge on openser:(. Following is my openser.cfg which is pretty much copy directly from sample but I kept getting "513 Message too big" response from this server for SIP registration, I know the SIP registration message size is ONLY 555 bytes, what could be wrong here? Thanks in Advance! # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP: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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib64/openser/modules/mysql.so" loadmodule "/usr/local/lib64/openser/modules/sl.so" loadmodule "/usr/local/lib64/openser/modules/tm.so" loadmodule "/usr/local/lib64/openser/modules/rr.so" loadmodule "/usr/local/lib64/openser/modules/maxfwd.so" loadmodule "/usr/local/lib64/openser/modules/usrloc.so" loadmodule "/usr/local/lib64/openser/modules/registrar.so" loadmodule "/usr/local/lib64/openser/modules/textops.so" loadmodule "/usr/local/lib64/openser/modules/auth.so" loadmodule "/usr/local/lib64/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) modparam("usrloc", "db_url", "mysql://openser:passwd@localhost/openser") # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:passwd@localhost/openser") # -- 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; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("sip.mydomain.com", "subscriber")) { www_challenge("sip.mydomain.com", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Bill _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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Re: [Users] 513 Message too big
by Norman Brandinger 12 Sep '06

12 Sep '06

Hi Bill, My msg:len line is as follows: if (msg:len >= max_len) { Perhaps there is an issue with the parsing of the size value you have specified. Regards, Norm Bill Zhang wrote: > Hi, > > I've exhausted my knowledge on openser:(. Following is my openser.cfg which > is pretty much copy directly from sample but I kept getting "513 Message too > big" response from this server for SIP registration, I know the SIP > registration message size is ONLY 555 bytes, what could be wrong here? > > Thanks in Advance! > > # > # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > debug=4 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=yes # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > fork=no > log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > children=4 > fifo="/tmp/openser_fifo" > > # > # uncomment the following lines for TLS support > #disable_tls = 0 > #listen = tls:your_IP: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" > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > loadmodule "/usr/local/lib64/openser/modules/mysql.so" > loadmodule "/usr/local/lib64/openser/modules/sl.so" > loadmodule "/usr/local/lib64/openser/modules/tm.so" > loadmodule "/usr/local/lib64/openser/modules/rr.so" > loadmodule "/usr/local/lib64/openser/modules/maxfwd.so" > loadmodule "/usr/local/lib64/openser/modules/usrloc.so" > loadmodule "/usr/local/lib64/openser/modules/registrar.so" > loadmodule "/usr/local/lib64/openser/modules/textops.so" > loadmodule "/usr/local/lib64/openser/modules/auth.so" > loadmodule "/usr/local/lib64/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) > modparam("usrloc", "db_url", "mysql://openser:passwd@localhost/openser") > > # -- auth params -- > # Uncomment if you are using auth module > # > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url", "mysql://openser:passwd@localhost/openser") > > # -- 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; > }; > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!method=="REGISTER") > record_route(); > > # subsequent messages withing a dialog should take the > # path determined by record-routing > if (loose_route()) { > # mark routing logic in request > append_hf("P-hint: rr-enforced\r\n"); > route(1); > }; > > if (!uri==myself) { > # mark routing logic in request > append_hf("P-hint: outbound\r\n"); > # if you have some interdomain connections via TLS > #if(uri=~"@tls_domain1.net") { > # t_relay("tls:domain1.net"); > # exit; > #} else if(uri=~"@tls_domain2.net") { > # t_relay("tls:domain2.net"); > # exit; > #} > route(1); > }; > > # if the request is for other domain use UsrLoc > # (in case, it does not work, use the following command > # with proper names and addresses in it) > if (uri==myself) { > > if (method=="REGISTER") { > if (!www_authorize("sip.mydomain.com", > "subscriber")) { > www_challenge("sip.mydomain.com", "0"); > exit; > }; > > save("location"); > exit; > }; > > lookup("aliases"); > if (!uri==myself) { > append_hf("P-hint: outbound alias\r\n"); > route(1); > }; > > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > exit; > }; > append_hf("P-hint: usrloc applied\r\n"); > }; > > route(1); > } > > > route[1] { > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > exit; > } > > Bill > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

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[Serusers] function capable of identifying the domain of the callee
by Ricardo Carvalho 12 Sep '06

12 Sep '06

Hi, I'm looking for a function capable of identifying the domain of the callee so I can implement something like this: if(domain_of_callee == domain_A) { do this } if(domain_of_callee == domain_B) { do that } I know that there are some functions like those in textops module, but the problem is that I can't search in the INVITE messages for that because I what that to be searched in INVITE messages sent from Ser to Asterisk for Voicemail purpose, and in that cases, the domain of the callee was already replaced by the IP of Asterisk server in those messages, and To and From parameters can have other domains because I'm using multi-domain implementation. Is there a function that does this? Thanks, Ricardo.

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[Users] presence server
by henri.keski-sikkila＠sesca.com 12 Sep '06

12 Sep '06

Hello, Are there any plans that OpenSER could be used as presence server? I'm using SER presence server, but I would like to use OpenSER. There is pa module in OpenSER, but it is some how broken? Best Regards Henri Keski-Sikkila

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[Serusers] Call generator
by Daniel Mirrha Santos 12 Sep '06

12 Sep '06

Does anybody know a software that generates simultaneous calls in sip or h323 for tests? Thanks, Daniel

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[Users] jabber module
by Javier Ramirez 12 Sep '06

12 Sep '06

Anyone has experience with jabber module ? I have a jabber server working, and openser working but I can't connect both. please, send me an example of openser.cfg if you want. best regards Javier

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[Users] Can someone explain
by nick 12 Sep '06

12 Sep '06

or point me to some documentation as to what append_hf() actually does? I see it throughout my openser.cfg, but it's not in the openser cookbook, or other visible documentation. Does it add a hint to log files, or actually include information in some other place? Thanks for any clues.. Nick

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Re: [Serusers] Reload Ser configuration and DB in runtime?
by Andrey Kouprianov 12 Sep '06

12 Sep '06

Hi, You certainly cannot make ser accept changes made to a configuration file during runtime. Take a look for instance at apache http server or should i say almost any server on Linux.. None of them allows accepting configurations dynamically. :) Besides, some threads in this list already discussed this issue. As for database, Im not sure.. But what exactly are you expecting to be reloaded automatically? Andrey.

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[Serusers] how to setup mediaproxy on remote server
by Ranjeet Kumar 12 Sep '06

12 Sep '06

Hi, I am able to setup mediaproxy on same sip proxy server. I want to setup mediaproxy on different server. Could you please help me in setting up mediaproxy on remote server. Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com

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[Serusers] can ser support IPV6??
by P Jayasree 12 Sep '06

12 Sep '06

Hi, can somebody tell me which version of ser supports IPV6? Thanks in advance, Regards, Jayasree. The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail immediately and permanently delete the message and any attachments.

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[Serusers] Can't cancel a forwarding call to Asterisk
by Paco Hsiao 12 Sep '06

12 Sep '06

Hi all, I configure ser to forward a call to Asterisk when callee is busy or unanswered. But, when the call is forwarding and the caller cancels the call, ser still forward the call to Asterisk and won't cancel the call. How do I really cancel the call and stop forwarding to Asterisk? I will be appreciate for any help! Here are my ser.cfg: # $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=192.168.10.114 # INSERT YOUR IP ADDRESS HERE port=5060 children=8 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 20) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # xlog("L_INFO", "<%mb>\n"); # ----------------------------------------------------------------- # 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!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # 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 (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("1"); }; 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=="BYE") { 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" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; #sl_send_reply("100", "Trying"); 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 (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:999[0-9]*@") { strip(3); route(4); route(5); break; }; if (!lookup("location")) { route(4); route(5); break; }; t_on_failure("1"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.168.10.124"); avp_write("i:90", "inv_timeout"); route(1); } 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")) { unforce_rtp_proxy(); break; } if (t_check_status("486") || t_check_status("408")) { revert_uri(); rewritehost("192.168.10.124"); avp_write("i:90", "inv_timeout"); append_branch(); t_relay(); break; }; unforce_rtp_proxy(); }

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[Serusers] Help with error
by Steve Blair 12 Sep '06

12 Sep '06

I am trying to relay calls from an on campus IP phone to proxy #1 then to proxy #2 then to an on campus gateway. The INVITE reaches the gateway but I get the "I'm sorry we cannot complete your call" recording from my carrier. Proxy #1's log shows the following error. "t_newtran: transaction already in process 0xb61d9a30" I found the following via google but the post doesn't describe how to fix the error. Is the only recourse to restructure the existing ser.cfg file or does the uac module provide a function to handle this situation? http://www.openser.org/pipermail/users/2006-April/003899.html Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

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Fwd: [Users] Sip trace on ...
by raviprakash sunkara 12 Sep '06

12 Sep '06

---------- Forwarded message ---------- From: raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> Date: Sep 12, 2006 8:49 AM Subject: Re: [Users] Sip trace on ... To: Alberto Pacifico <alberto_pacifico(a)yahoo.com> Hi Alberto , Thanks For reply, and Nice to met U , What u'd told , I had done before only, Can U tell, Who to process openser with sip_trace .... On 9/11/06, Alberto Pacifico <alberto_pacifico(a)yahoo.com> wrote: > > > Hi Sunkara, > > If you are working with Openser 1.1.0, you need to > configure the file named "openserctlrc" with the > database's informations. > > Good Luck! > > Alberto > > --- raviprakash sunkara > <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > > Hi Users, > > > > I'm trying the sip_traces ,,,, > > > > When I typed below command , I got these errors > > openserctl fifo sip_trace on > > database engine 'MYSQL' loaded > > Control engine 'FIFO' loaded > > entering fifo_cmd sip_trace on > > ERROR: Error opening openser's FIFO FIFO > > ERROR: Make sure you have line fifo=FIFO in your > > config > > > > > > > > > > -- > > Thanks and Regards with cheers > > Sunkara Ravi Prakash (Voip Developer) > > Hyperion Technology > > www.hyperion-tech.com > > +91-9985077535 > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

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[Users] Cisco 5350
by Dan 12 Sep '06

12 Sep '06

Anybody able to get cisco 5350 register with openser using username and password. Am using openser 1.0.1 with mysql authentication. When I try to send calls from cisco , I get 401 unauthorised. Have made entries in sip-ua : authentication username 90557 password 06575A751F1B504C5D42455C28 realm xxx.xxx.xxx.xxx registrar ipv4:xxx.xxx.xxx.xxx expires 3600 Under dialpeer : session target ipv4:xxx.xxx.xxx.xxx Where xxx.xxx.xxx.xxx is the ip address of openser. Can anyone point out the problem? Regards DAN

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[Users] Suppression of Silence Suppression
by Federico Giannici 12 Sep '06

12 Sep '06

Is there any way to force the disabling of the "silence suppression" for an RTP stream? I'd like to disable it at the SIP proxy, maybe modifying some SDP attribute... Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Users] Q:ABOUT:DETECT IF A PART IS KEEP ALIVE
by Alejandro Sanchez 12 Sep '06

12 Sep '06

Hello. My scenario is the next. I have two sip phones and i need when one of them dissconnect (for reasons different to the request user posible errors) stop the accounting. What's the correct form tio implement this? Thank's in advanced. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] Reload Ser configuration and DB in runtime?
by Ricardo Carvalho 12 Sep '06

12 Sep '06

How may I reload Ser configuration file, or alterations made in DB tables in Ser runtime, without killing the Call leg of transactions in progress at that time? Thanks, Ricardo.

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[Users] Help in redirection server
by Michel Zenone 12 Sep '06

12 Sep '06

Hi! I try to make my openser act as a redirection server when a customer is unavailable. And I need to know, receiving an INVITE, if this INVITE is due to a redirection or not. Does somebody have an idea to get this information? Thanks in advance, Michel

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[Serusers] Help in redirection server
by Michel Zenone 12 Sep '06

12 Sep '06

Hi! I try to make my SIP proxy act as a redirection server when a customer is unavailable. And I need to know, receiving an INVITE, if this INVITE is due to a redirection or not. Does somebody have an idea to get this information? Thanks in advance, Michel

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[Users] action.c:1: error: bad value (x86_64) for -mtune= switch
by Ronald Wiplinger 12 Sep '06

12 Sep '06

First try to compile Openser on an AMD Semperon CPU: openser-1.1.0-tls> make all Compiling action.c gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=x86_64 -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"x86_64"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.0"' -D__CPU_x86_64 -D__OS_linux -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_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c action.c -o action.o action.c:1: error: bad value (x86_64) for -mtune= switch make: *** [action.o] Error 1 How can I solve that? bye Ronald

2 3

[Serusers] Show registered users in ser
by Jan Schumacher 12 Sep '06

12 Sep '06

Hallo, how can I show the registered users in ser ? Is there a way like ''sip show registry'' on the terminal? thanks in advance Jan

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[Serusers] Can I have separate sip proxy & sip registrar?
by Ranjeet Kumar 12 Sep '06

12 Sep '06

Hi, I want to deploy the sip server, where I want one server as sip proxy and other as registrar. Is it possible in SER? Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com

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[Serusers] Re: rewrite the FROM part of SIP INVITE
by Alex Fler 12 Sep '06

12 Sep '06

Hello guys, Sorry for a stupid question, but how do you actually rewrite the FROM part of SIP INVITE ? Do you use avp ? Could someone give me an example ? Thanks to all Alex Fler

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[Serusers] deference between rtpproxy and mediaproxy
by vijay tiwari 12 Sep '06

12 Sep '06

hi what is the deference between rtpporxy and mediaproxy with regards Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

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[Serusers] SER - Avoid some sip message
by Damian Nardelli 11 Sep '06

11 Sep '06

How can i transform a 183 Session Progress into a 180 Ringing? is it possible not to redirect a message SIP from the configuration file (for example the "183 Session Progress")? How can i do that? Thanks. Damián

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[Serusers] Grandstream Budgetone phones don't show alphanumeric caller right
by Ricardo Carvalho 11 Sep '06

11 Sep '06

I have tested "Grandstream Budgetone 102" and "Grandstream Budgetone 200" and with both, if they are called from a caller that is an alphanumeric user, their display shows a unintelligible name impossible to figure out who is calling!! If the caller is a numeric one, in both phones their display shows correctly the caller's contact. I've updated their firmwares to the latest ones and that problem persists... Does anybody also experienced this? Thanks, Ricardo.

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[Users] Sip trace on ...
by raviprakash sunkara 11 Sep '06

11 Sep '06

Hi Users, I'm trying the sip_traces ,,,, When I typed below command , I got these errors openserctl fifo sip_trace on database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd sip_trace on ERROR: Error opening openser's FIFO FIFO ERROR: Make sure you have line fifo=FIFO in your config -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

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[Users] openser dialog statistics
by Greg Fausak 11 Sep '06

11 Sep '06

I've been attempting to get statistics for dialogs to work for me: I've loaded the dialog.so module, and set the dlg_flag to 2. However, I've not been able to see dialogs open or close when I use the openserctl get_statistics all call. I know statistics are working because I put them in, increment them in the routing script, and I see them in the openserctl get_statistics output. Can somebody post an example of how to get dialog counting to work? Thanks, -g -- Greg Fausak greg(a)thursday.com

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[Users] Re: [Serweb-users] Can serweb-0.9.4 run in openser? Please help....
by Ferianto siregar 11 Sep '06

11 Sep '06

Dear Karel, Thank you very much again for your kind-hearted. Thank you very much. I am sorry Karel. If I can make a wish to you again, I have some questions anymore. Can I? Please... :) 1. I have downloaded from pear.net the DB packet. The version of the packet is DB-1.7.6. I have untar it in root directory. I can found the file DB.php in it. But I am confused, where should I place this DB.php file? So I can solve this error? Would ypu mind telling me..Please.. Sould I have internet connection to install it? Because I do not have internet connection now. 2. Would you mind telling me what you mean by "fix include_path in your php.ini". Would you mind give me an example? Please... I do hope you can help me. I know that I am still nwe in this system and I have lack skill in this system. Please help me...Please.. Thank you very much... Regards, Ferianto Karel Kozlik <karel(a)iptel.org> wrote: Hi, it looks like php-pear is not installed or php can't find it. Check if file DB.php (from the PEAR library) is in dir /usr/share/pear. If not, install php pear into this dir, or fix include_path in your php.ini. More info about php-pear you could find at http://pear.php.net regards Karel Ferianto siregar napsal(a): > Dear all, > > Thanks to every body in this forum for help me.. Thank you. > > All, now I try to build serweb-0.9.4 in openser-1.1.0-tls. I have ever > got a message from this forum that the serweb-0.9.4 can work sucessfully > in openser-1.1.0. > So, I tried to install it in openser-1.1.0. But, When I tried, I got the > error message. > This is the error message that I have got. I do hope anybody can give me > a suggestion to solve this problem. Please... > > The error message is : > [Sat Sep 09 13:02:56 2006] [notice] mod_python: Creating 4 session > mutexes based on 256 max processes and 0 max threads. > [Sat Sep 09 13:02:56 2006] [notice] Apache/2.2.0 (Fedora) configured -- > resuming normal operations > [Sat Sep 09 13:03:10 2006] [error] [client 202.95.149.251] PHP Warning: > require_once(DB.php) [ > href='function.require-once'>function.require-once]: failed to open > stream: No such file or directory in > /var/www/serweb-0.9.4/html/main_prepend.php on line 38 > [Sat Sep 09 13:03:10 2006] [error] [client 202.95.149.251] PHP Fatal > error: require_once() [function.require]: > Failed opening required 'DB.php' (include_path='.:/usr/share/pear') in > /var/www/serweb-0.9.4/html/main_prepend.php on line 38 > [root@sip httpd]# > > I have tried to analyze the error. But, I can not understand what > happen. Maybe because my lack skill in php programming. > I do hope anybody can help me..Please... > > > Thanks > Regards, > > > Ferianto > > Note : > 1. I am using Fedora Core 5 > 2. The php and mysql packet are : > > > php-pdo-5.1.2-5 > php-soap-5.1.2-5 > php-bcmath-5.1.2-5 > php-ncurses-5.1.2-5 > php-gd-5.1.2-5 > php-mbstring-5.1.2-5 > php-xmlrpc-5.1.2-5 > php-ldap-5.1.2-5 > php-mysql-5.1.2-5 > php-devel-5.1.2-5 > php-imap-5.1.2-5 > php-pgsql-5.1.2-5 > php-xml-5.1.2-5 > php-5.1.2-5 > php-pear-1.4.6-2 > php-dba-5.1.2-5 > php-odbc-5.1.2-5 > php-snmp-5.1.2-5 > > libdbi-dbd-mysql-0.8.1a-1.2.1 > mysql-5.0.18-2.1 > mysql-server-5.0.18-2.1 > mod_auth_mysql-3.0.0-3 > php-mysql-5.1.2-5 > mysql-connector-odbc-3.51.12-1.2.1 > mysql-devel-5.0.18-2.1 > > > > ------------------------------------------------------------------------ > Yahoo! Messenger with Voice. Make PC-to-Phone Calls > > to the US (and 30+ countries) for 2ï¿½/min or less. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serweb-users mailing list > Serweb-users(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serweb-users --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

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[Serusers] MS LCS 2005 / SER / Asterisk Integration
by harrygaillac-sip＠yahoo.fr 11 Sep '06

11 Sep '06

Hi to all, I read http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Int… Is it possible to use ser as a presence server instead of LCS 2005 ? Harry ___________________________________________________________________________ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com

1 0

[Users] 'To' header field and redirect
by Michel Zenone 11 Sep '06

11 Sep '06

Hi! My sip phone has troubles to understand which sip address to contact when it receives a redirection from my openser: In fact 'To' header field contains the new uri and the last one... So now I try to remove the last one from the field but I don't know how to do this! Can somebody help me? Thank you! Michel

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[Serusers] About the accounting problem
by AR. 11 Sep '06

11 Sep '06

Dear All I don't know that is anyone have the same problem. The sipua calls to the pstn user through the cisco voice gateway, then if the pstn user ends the session, the radius log will not to log the acct_stop_time until the sipua handup could anyone help Best Regards AR.

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[Serusers] No vm.so in my SER modules.
by Meidiana Ten 11 Sep '06

11 Sep '06

Good afternoon guys...I'd like to ask you some questions based on my project in building SIP server.. I need to activate msilo so that it could store offline messages from clients. Is it common that in my SER modules that i have installed has no module named vm.so?? The following configuration is my ser.cfg. Please tell me what i have to add to activate msilo module: ---------------------------------------------------------------------------------------------------------------------------- debug=3 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=8 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") alias='pcr.ac.id' 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!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); 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" && 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(); }; if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } #if (!allow_trusted() && nat_uac_test("19")) { # setflag(6); #} lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; } if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.254:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); #if (!allow_trusted() && isflagset(6)) { if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } 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(); }; } -------------------------------------------------------------- Then, i'd like to know why when i run my server, it comes with warning about RTP Proxy? What do i need to setup a right RTP Proxy? Thanx for the attention. --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

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[Serusers] Re: [Serweb-users] how to add users via serweb
by Andrey Kouprianov 10 Sep '06

10 Sep '06

Not create accounts per se. Ask for an account.. And sometimes it doesnt work.. Anyway, there a "register" link on the login page for users. Start from there. On 9/9/06, G.Jacobsen <g_jacobsen(a)yahoo.co.uk> wrote: > > Users can create their own accounts - not the admin. > > Cheers > > Gerry > > > ----- Original Message ----- > *From:* Ranjeet Kumar <ranjeetk(a)aztecsoft.com> > *To:* serweb-users(a)lists.iptel.org > *Sent:* Saturday, September 09, 2006 2:56 PM > *Subject:* [Serweb-users] how to add users via serweb > > > > Hi, > > > > I can see user through serweb, but there is no option to create user via > serweb. > > Please help me > > > > Thanks, > > Ranjeet > > > > The information contained in, or attached to, this e-mail, contains > confidential information and is intended solely for the use of the > individual or entity to whom they are addressed and is subject to legal > privilege. If you have received this e-mail in error you should notify the > sender immediately by reply e-mail, delete the message from your system and > notify your system manager. Please do not copy it for any purpose, or > disclose its contents to any other person. The views or opinions presented > in this e-mail are solely those of the author and do not necessarily > represent those of the company. The recipient should check this e-mail and > any attachments for the presence of viruses. The company accepts no > liability for any damage caused, directly or indirectly, by any virus > transmitted in this email. > > www.aztecsoft.com > > ------------------------------ > > _______________________________________________ > Serweb-users mailing list > Serweb-users(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serweb-users > > > _______________________________________________ > Serweb-users mailing list > Serweb-users(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serweb-users > > >

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[Users] Can serweb-0.9.4 run in openser? Please help....
by Ferianto siregar 10 Sep '06

10 Sep '06

Dear all, Thanks to every body in this forum for help me.. Thank you. All, now I try to build serweb-0.9.4 in openser-1.1.0-tls. I have ever got a message from this forum that the serweb-0.9.4 can work sucessfully in openser-1.1.0. So, I tried to install it in openser-1.1.0. But, When I tried, I got the error message. This is the error message that I have got. I do hope anybody can give me a suggestion to solve this problem. Please... The error message is : [Sat Sep 09 13:02:56 2006] [notice] mod_python: Creating 4 session mutexes based on 256 max processes and 0 max threads. [Sat Sep 09 13:02:56 2006] [notice] Apache/2.2.0 (Fedora) configured -- resuming normal operations [Sat Sep 09 13:03:10 2006] [error] [client 202.95.149.251] PHP Warning: require_once(DB.php) [<a href='function.require-once'>function.require-once</a>]: failed to open stream: No such file or directory in /var/www/serweb-0.9.4/html/main_prepend.php on line 38 [Sat Sep 09 13:03:10 2006] [error] [client 202.95.149.251] PHP Fatal error: require_once() [<a href='function.require'>function.require</a>]: Failed opening required 'DB.php' (include_path='.:/usr/share/pear') in /var/www/serweb-0.9.4/html/main_prepend.php on line 38 [root@sip httpd]# I have tried to analyze the error. But, I can not understand what happen. Maybe because my lack skill in php programming. I do hope anybody can help me..Please... Thanks Regards, Ferianto Note : 1. I am using Fedora Core 5 2. The php and mysql packet are : php-pdo-5.1.2-5 php-soap-5.1.2-5 php-bcmath-5.1.2-5 php-ncurses-5.1.2-5 php-gd-5.1.2-5 php-mbstring-5.1.2-5 php-xmlrpc-5.1.2-5 php-ldap-5.1.2-5 php-mysql-5.1.2-5 php-devel-5.1.2-5 php-imap-5.1.2-5 php-pgsql-5.1.2-5 php-xml-5.1.2-5 php-5.1.2-5 php-pear-1.4.6-2 php-dba-5.1.2-5 php-odbc-5.1.2-5 php-snmp-5.1.2-5 libdbi-dbd-mysql-0.8.1a-1.2.1 mysql-5.0.18-2.1 mysql-server-5.0.18-2.1 mod_auth_mysql-3.0.0-3 php-mysql-5.1.2-5 mysql-connector-odbc-3.51.12-1.2.1 mysql-devel-5.0.18-2.1 --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

2 2

[Serusers] no sound
by jorain 09 Sep '06

09 Sep '06

Hi all, I have 2 UA, one is fixed IP and another is dynamic IP, when i call from PSTN to fixed IP, it running well(audio on both side). But when call from either PSTN or fixed IP phone to dynamic IP phone, there is no sound on both sides. And when i call from fixed IP phone to dynamic IP phone, there is no sound on dynamic IP phone. What could be the problems? Thanks, Best Regards, jorain

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[Serusers] Out-of-office-reply: Serusers Digest, Vol 41, Issue 9
by David.Monahan＠aepona.com 09 Sep '06

09 Sep '06

I will be out of the office 8th - 18th September and will not be checking email during this time.

1 0

[Serusers] How to setup SIP Proxy seperatley
by Ranjeet Kumar 09 Sep '06

09 Sep '06

Hi, I want to setup the sip server in below mentioned way : - 1 server act like - proxy 1 server act like - registrar 1 server act like - sip domain Is it possible to setup like this? If you could you please give me some idea how to do it. Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com

1 0

[Users] Openser+mediaproxy... call still on hold
by ram 09 Sep '06

09 Sep '06

Hi iam testing media proxy with openser openser version: openser 1.0.1-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.7 2005/10/28 14:55:23 klaus_darilion Exp $ main.c compiled on 14:52:07 Jun 21 2006 with gcc 3.4.3 Media proxy 1.7.2 even though the call ended still it shows hold almost all 431 seconds and still its going on even thouhg i have logged out from users both the side UA1 ----- | ------NAT router----SER (public IP)---internet UA2------ here the session shows Every 2.0s: /usr/local/mediaproxy/sessions.py Sat Sep 9 21:12:32 2006 Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------------------------------------------------------ UA2IP:8002 - openserip:60030 - UA1IP:41182 hold (431) 7'31" G711u Audio 21.15k/35.31k/14.17k Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.7.2 how can i reset this kind of connections ? Ram

1 0

[Serusers] LCR question
by Steve Blair 08 Sep '06

08 Sep '06

I would like to use LCR so that outbound calls from my domain may try one of several gateways in sequence. The callers (from uri) will not be authenticated or registered to the proxy doing the LCR. Will LCR work in this type of environment? A few test calls have failed so far. Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

2 4

[Serusers] New polycom firmware / presence
by harrygaillac-sip＠yahoo.fr 08 Sep '06

08 Sep '06

Hello, I look at the new sip firmware however i don't undanstand the presence features. I don't use LCS but SER as presence server this one is able to provide a ressource list server and xcap server for sip buddies lists . Does polycom phones can suscribe to a sip:lists@domain.com for example to watch buddies status ? Regards ___________________________________________________________________________ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com

2 1

[Users] Please help for configuration TLS in Fedora Core5
by Ferianto siregar 08 Sep '06

08 Sep '06

Dear all, I would like to say thanks to all of you for your kind-hearted read my message. Thank you very much. All, I need you help now.Please.. Now, I tried to build openser with TLS in Fedora Core 5. Before, I use Redhat 9 and the openser installation can be build successfully.( I can make a call with TLS support) But, Why when I tried to install it in Fedora Core 5, the openser can not run?(there is no pid for openser) But, When I erase the TLS configuration (TLS script for enable the TLS) in openser.cfg, the openser can run successfully.(There is a pid). So, in my opinion, the TLS Script in openser.cfg is error. But, I need this script to make secure openser server (openser with TLS support). What should I do? I do hope anybody can give a suggestion. Please help me...Please.. Thank you very much Regards, Ferianto Note: 1. This is the error message that I got when execute, # openser, command: 0(8715) DEBUG:socket2str: <udp:202.95.149.251:5060> 0(8715) DEBUG:socket2str: <tcp:202.95.149.251:5060> 0(8715) DEBUG:socket2str: <tls:202.95.149.251:5061> Listening on udp: 202.95.149.251 [202.95.149.251]:5060 tcp: 202.95.149.251 [202.95.149.251]:5060 tls: 202.95.149.251 [202.95.149.251]:5061 Aliases: tls: sip:5061 tls: sip.pcr.ac.id:5061 tcp: sip:5060 tcp: sip.pcr.ac.id:5060 udp: sip:5060 udp: sip.pcr.ac.id:5060 0(8715) fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(8715) fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560 0(8715) fm_malloc_init(0xb5fbb000, 33554432), start=0xb5fbb000 0(8715) shm_mem_init: success 0(8715) init_tcp: using epoll_lt as the io watch method (auto detected) 0(8715) init_tls: Entered 0(8715) WARNING:init_tls: disabling compression due ZLIB problems 0(8715) ERRRO:init_tls: null openssl compression methods 0(8715) could not initialize tls, exiting... 0(8715) DEBUG: tm_shutdown : start 0(8715) DEBUG: tm_shutdown : emptying hash table 0(8715) DEBUG: tm_shutdown : releasing timers 0(8715) DEBUG: tm_shutdown : removing semaphores 0(8715) DEBUG: tm_shutdown : destroying tmcb lists 0(8715) DEBUG: tm_shutdown : done 0(8715) destroy_tls: Entered 0(8715) shm_mem_destroy 0(8715) destroying the shared memory lock 2. This is the contain part of openser.cfg listen=202.95.149.250 port=5060 children=4 dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" fifo_mode=0666 alias="pcr.ac.id" # uncomment the following lines for TLS support disable_tls = 0 listen = tls:202.95.149.250:5061 tls_verify_client = on tls_require_client_certificate = on tls_verify_server=on tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

4 6

[Users] Register confirmation error in serweb. Please help...
by cv aik 08 Sep '06

08 Sep '06

Dear all, Thank you very much. Now, I tried to build serweb in my openser server. I user openser-1.1.0-tls and for serweb, I use serweb-2004-07-27. I can install it successfully. I mean that from the serweb browser, I can login as admin and I can login with username that I have added from "openserctl add" command before. But, I have a problem when I add a new user from serweb browser. When I try to add a new user, I always get the error message in register confirmation. The error message says that : "SQL query error in line 43". I have tried to analize it. But I can not. Maybe because I have lack skill in this. I do hope anybody can give me some suggestions. Please help me....Please... Thank you very much Regards, Aldi Note: This the contain of confirmation.php file from line 41 to 45 // get the max alias number if (!$alias=get_alias_number($errors)) break; $q="insert into ".$config->table_subscriber." select * from ".$config->table_pending." where confirmation='$nr'"; $res=mySQL_query($q); if (!$res) {$errors[]="error in SQL query, line: ".__LINE__; break;} __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] [openser] 900/976 dialing advice
by Tracy Lofton 08 Sep '06

08 Sep '06

Hi all, I recently added a function to openser.cfg as shown below: if (uri=~"^sip:919(00|76)[0-9]{7}@") { acc_db_request("403 - 900/976 Disabled", "acc"); xlog("L_INFO", "Cancel attempt to call 9xx number (not 911), db recorded\n"); exit; }; but when a user calls the 900/976 prefix, it records in the database and the caller is restricted access, however, the caller waits too long before finally hearing a busy signal. Is there a way to speed up a return on the busy signal (some type of timer) or even possibly transfer the call to Asterisk IVR to hear a recording like, call cannot be completed or something, or some type of faster 486 busy. Need advice and how should i implement. Thanks, Tracy _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

2 1

[Users] Openser and which hardware?
by Ronald Wiplinger 08 Sep '06

08 Sep '06

I want to install a new computer with OpenSer. In the past I had some troubles with the right OS and right hardware. The system I want to build is for testing only, but if it is working it should be ported to other machines easy. I have the choice of: AMD or Intel I have the choice of SuSE 10.1 Debian Ubuntu I would like to use (in this order): AMD & SuSE 10.1 AMD & Ubuntu 6.06 AMD & Debian Intel & SuSE 10.1 Intel & Ubuntu 6.06 Intel & Debian What do you use? Did you have any problems with compiling and setting up with one of the above configuration? bye Ronald

2 1

[Users] Presence
by Bryan Hertz 07 Sep '06

07 Sep '06

Anyone who has experience with OpenSER and the Presence module please contact me if you are interested in some freelance work, or just want to help :-) Thank you Bryan

3 2

RE: [SPAM header] - Re: [Serusers] CPU % - Email found in subject
by Teran Jesus 07 Sep '06

07 Sep '06

I think 4 proces are block, and take a lot of % of CPU, yesterday I restart the SER and now loo like this: root 29921 0.0 0.3 37548 3712 ? S Sep06 0:00 /usr/sbin/ser root 29923 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29925 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29928 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29934 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29945 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29946 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29949 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29953 0.0 0.3 37548 3708 ? S Sep06 0:00 /usr/sbin/ser root 29958 0.0 0.4 37548 4396 ? S Sep06 0:00 /usr/sbin/ser root 29959 0.0 0.4 37548 4376 ? S Sep06 0:00 /usr/sbin/ser root 29960 0.0 0.4 37548 4388 ? S Sep06 0:00 /usr/sbin/ser root 29961 0.0 0.4 37548 4400 ? S Sep06 0:00 /usr/sbin/ser root 29962 0.0 0.4 37548 4368 ? S Sep06 0:00 /usr/sbin/ser root 29963 0.0 0.4 37548 4404 ? S Sep06 0:00 /usr/sbin/ser root 29964 0.0 0.4 37548 4412 ? S Sep06 0:00 /usr/sbin/ser root 29965 0.0 0.4 37548 4376 ? S Sep06 0:00 /usr/sbin/ser root 29966 0.0 0.5 38560 5712 ? S Sep06 0:06 /usr/sbin/ser and the vmstat like this: procs memory swap io system cpu r b swpd free buff cache si so bi bo in cs us sy id wa 0 0 17188 20012 169196 623072 0 0 1 2 1 1 0 0 0 0 0 0 17188 20012 169196 623072 0 0 0 36 124 76 0 0 100 0 0 0 17188 20008 169196 623072 0 0 0 20 121 68 0 0 100 0 0 0 17188 20004 169196 623072 0 0 0 0 108 34 0 0 100 0 0 0 17188 20004 169196 623072 0 0 0 0 114 56 0 0 100 0 0 0 17188 20004 169196 623072 0 0 0 24 128 137 0 0 100 0 0 0 17188 20004 169196 623072 0 0 0 20 133 110 0 0 100 0 0 0 17188 20004 169196 623072 0 0 0 0 116 93 0 0 100 0 0 0 17188 20004 169196 623072 0 0 0 0 136 178 0 0 100 0 0 0 17188 20004 169196 623072 0 0 0 0 134 192 0 0 99 0 We can see all is normal only one percent of the Server is used. -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Jueves, 07 de Septiembre de 2006 02:17 a.m. To: Teran Jesus Cc: Serusers(a)lists.iptel.org Subject: [SPAM header] - Re: [Serusers] CPU % - Email found in subject Well, 0.8.14 is the latest in the 0.8 series, while the latest stable release is the 0.9.x series. Maybe you want to consider upgrading? Apart from that, I see a lot of numbers, but you don't say what you think is wrong?! g-) Teran Jesus wrote: > Hello i am using the ser 0.8.12, it works fine, right now i run the linux > comand ps -aux , to see the list of proces that are runing and i see that i > have SER proces like the next, can you said me if it is correct or i have a > problem with my ser or is a problem with the version of the SER, because > sometimes the SER needs to reset because don't respond, if you see the last > SER reset tht i need to made was August 25. > > > USER PID %CPU %MEM VSZ RSS TTY STAT START > TIME COMMAND > root 24416 32.4 0.7 38668 7456 ? R Aug25 5547:04 > /usr/sbin/ser > root 24421 32.4 0.7 38684 7468 ? R Aug25 5537:54 > /usr/sbin/ser > root 24426 32.4 0.7 38664 7484 ? R Aug25 5541:14 > /usr/sbin/ser > root 24431 32.4 0.7 38652 7468 ? R Aug25 5543:54 > /usr/sbin/ser > root 24437 0.0 0.8 38676 9056 ? S Aug25 5:56 /usr/sbin/ser > root 24439 0.0 0.8 38756 9064 ? S Aug25 6:10 /usr/sbin/ser > root 24444 32.4 0.7 38684 7456 ? R Aug25 5551:00 > /usr/sbin/ser > root 24450 32.4 0.7 38664 7472 ? R Aug25 5539:39 > /usr/sbin/ser > > I ran the vmstat 1 10 too and i believe is not normall the processing, can > you help me?. > > r b swpd free buff cache si so bi bo in cs us sy id > wa > 6 0 27440 17824 168972 649488 0 0 1 2 0 2 0 0 1 > 0 > 6 0 27440 17844 168972 649488 0 0 0 40 130 480364 12 87 1 > 0 > 6 0 27440 17844 168972 649488 0 0 0 0 118 475900 11 89 0 > 0 > 6 0 27440 17840 168972 649488 0 0 0 0 120 481403 11 87 2 > 0 > 6 0 27440 17840 168972 649488 0 0 0 0 114 483715 12 86 2 > 0 > 6 0 27440 17836 168972 649488 0 0 0 36 134 476488 12 87 1 > 0 > 6 0 27440 17840 168972 649488 0 0 0 64 137 482903 14 84 1 > 0 > 6 0 27440 17840 168972 649488 0 0 0 68 148 473984 14 86 0 > 0 > 6 0 27440 17840 168972 649488 0 0 0 64 143 476887 12 88 0 > 0 > 6 0 27440 17840 168972 649488 0 0 0 52 147 476060 12 88 0 > 0 > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

2 1

Re: [Serusers] ACC 900/976 Dialing Advice
by Tracy Lofton 07 Sep '06

07 Sep '06

Hi all, I recently added a function to openser.cfg as shown below: if (uri=~"^sip:919(00|76)[0-9]{7}@") { acc_db_request("403 - 900/976 Disabled", "acc"); xlog("L_INFO", "Cancel attempt to call 9xx number (not 911), db recorded\n"); exit; }; but when a user calls the 900/976 prefix, it records in the database and the caller is restricted access, however, the caller waits too long before finally hearing a busy signal. Is there a way to speed up a return on the busy signal (some type of timer) or even possibly transfer the call to Asterisk IVR to hear a recording like, call cannot be completed or something, or some type of faster 486 busy (failure_route or record_route?). Need advice and how should i implement. Thanks, Tracy _________________________________________________________________ Got something to buy, sell or swap? Try Windows Live Expo ttp://clk.atdmt.com/MSN/go/msnnkwex0010000001msn/direct/01/?href=http://exp…

1 0

Re: [Serusers] ACC 900/976 Dialing Advice
by Tracy Lofton 07 Sep '06

07 Sep '06

Hi all, I recently added a function to openser.cfg as shown below: if (uri=~"^sip:919(00|76)[0-9]{7}@") { acc_db_request("403 - 900/976 Disabled", "acc"); xlog("L_INFO", "Cancel attempt to call 9xx number (not 911), db recorded\n"); exit; }; but when a user calls the 900/976 prefix, it records in the database and the caller is restricted access, however, the caller waits too long before finally hearing a busy signal. Is there a way to speed up a return on the busy signal (some type of timer) or even possibly transfer the call to Asterisk IVR to hear a recording like, call cannot be completed or something, or some type of faster 486 busy. Need advice and how should i implement. Thanks, Tracy _________________________________________________________________ Windows Live Spaces is here! Its easy to create your own personal Web site. http://spaces.live.com/signup.aspx

1 0

Re: [Serusers] SER & Asterisk (IVR problem)
by Roberto Lopes 07 Sep '06

07 Sep '06

Hi, Well I don't think I could skip the authorization process because there are analogical devices which need to be authenticated, so it would create problems. How do I add the Asterisk Server to the trust tables? I'm actually thinking of adding Asterisk as a RADIUS client to the RADIUS server (freeRadius) What do you think? Thanks in advance, Roberto So... if I understand this correctly, you're having problems with diverted calls FROM the Asterisk IVR not being able to authenticate on SER? Could you just add the Asterisk server to the trust tables and skip the auth process for calls coming back or would that create problems? N. * On Tue, 5 Sep 2006 21:16:45 +0100, Roberto Lopes wrote* > Hi to all! > I've a big problem with SER & Asterisk > I've a network where i put them working together. > Asterisk is the media gateway with PSTN as well as being the server which is making all the services available (IVR, voicemail, conference calls, etc). > SER just routes calls either to the softphones, VoIP phones or to Asterisk who will just forward the calls to PSTN (via Asterisk). > Till now everything works well except the service IVR which is working on Asterisk. > > Details: > *MailScanner has detected a possible fraud attempt from "192.168.226.13" claiming to be* *MailScanner warning: numerical links are often malicious: 192.168.226.13 - IP SER > **MailScanner has detected a possible fraud attempt from "192.168.226.45" claiming to be* *MailScanner warning: numerical links are often malicious: 192.168.226.45 - IP Asterisk * *** > > The problem is only after the user (roberto(a)192.168.226.13 - authenticated and authorized in RADIUS) calls IVR. > Till then the user calls, IVR gives some information to the user and everything goes well. > Problem is when he presses a number which will have to redirect the call to another user. > And why is there a problem. * *> Because the initially roberto(a)192.168.226.13 to be able to call IVR had to rewrite host & port **MailScanner has detected a possible fraud attempt from "192.168.226.45" claiming to be* *MailScanner warning: numerical links are often malicious: 192.168.226.45 so that the call would reach Asterisk who would then call to the IVR. * *** > That rewriting turned roberto(a)192.168.226.13 into roberto(a)192.168.226.45. That is, I can say, that it became a "new user". > While it's on the IVR there's no problem, but when call is diverted it has to give its credentials... * *> Credentials that don't exist in RADIUS cause it only accepts usernames that end in @ **MailScanner has detected a possible fraud attempt from " 192.168.226.13" claiming to be* *MailScanner warning: numerical links are often malicious: 192.168.226.13 (IP of SER) * *** > > Here is the following code, both of extensions file in Asterisk and configure file of SER > I hope somebody knows how to solve it or at least find a way how to. > > extensions.conf - Asterisk > [sip] exten => 74001,1,Answer exten => 74001,2,SetMusicOnHold(default) exten => 74001,3,Set(TIMEOUT(digit)=5) exten => 74001,4,Set(TIMEOUT(response)=10) exten => 74001,5,Background(menu) > exten => 1,1,Dial(SIP/roberto@192.168.226.13:5060,,r) ... exten => 7,1,Goto(sip,74001,1) exten => 8,1,Hangup > > ser.cfg - SER > > ... > if ((uri=~"^sip:7[0-9]{4}@.*")) { > route(5); > break; > }; ... > route[5] { # Redirecting to Asterisk > # MsgLog > xlog("L_ALERT","(WARNING) Sending to Asterisk request of %fu to %tu\n"); * ***> rewritehostport("**MailScanner has detected a possible fraud attempt from " 192.168.226.45:5060" claiming to be* *MailScanner warning: numerical links are often malicious: 192.168.226.45:5060"); * * > route(1); > } > > > Thanks for your help, > Roberto Lopes *

1 0

[Serusers] avp_delete problem
by Luis Silva 07 Sep '06

07 Sep '06

Hi all!! I'm having a big problem when trying to delete an avp. First of all, I do $avp_name= @db.fetch[1].field[1]; an retreive some information from the db. After using it I do avp_delete("$avp_name/g"); . Then I print the avps with the avp_print() an I watch that the value is still there. What might be the problem? thanks a lot Regards, Luís Silva

2 5

Re: [Users] Error in installing Ruby...Please help...Please
by Ferianto siregar 07 Sep '06

07 Sep '06

Dear Sunkara, Thank you very much. Maybe I am too stupid to understand you explanation. I am sorry because I ask you a question anymore? Can I ? Please.... :-) From your explanation, that`s mean I just have to install all packet from FC5 then untar the openseradmin packet and it will work successfully? Do I need Ruby (as mentioned in openseradmin installation note)? Because FC5 has had Ruby packet in it. Thanks you very much for your explanation. Thanks Regards, Ferianto raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: Hi Ferianto , I Too using the FC5 , install the all packages in FC5 from add or remove programs, And then process the openser addminstroator, Now the Ruby is work, On 9/7/06, Ferianto siregar <ferianto_voip(a)yahoo.com> wrote: Dear Sunkara, Thank you very much. I use FedoraCore 5 raviprakash sunkara < sunkara.raviprakash.feb14(a)gmail.com> wrote: Hello Siregar, I need urs Which OS U r using... On 9/6/06, Ferianto siregar <ferianto_voip(a)yahoo.com > wrote: Dear all, Thank you very much for your help. A few days ago, I have sent a message that to ask about openser admin installation. Thank you very much for your help. Thanks Now, I have tried to install it again. But, When I compile the rubygerms-0.9.0.tar.gz, I got error message. This is the task that I have done before: 1. Untar the rubygerms packet # tar -zxvf rubygems-0.9.0.tar.gz # cd rubygems-0.9.0 # ruby setup.rb # gem install --include-dependencies When I type this command, "gem install --include-dependencies", I got error message. This is the error message: Attempting local installation of 'rails' Local gem file not found: rails*.gem Attempting remote installation of 'rails' Updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Gem::RemoteSourceException) Error fetching remote gem cache: getaddrinfo: Temporary failure in name esolution What`s wrong? I do hope anybody can help me to solve this problem. Please help me...Please help because of my lack skills..Please Note: 1. I do not have connection through the internet, so for rails packet, I have downloaded the rails-1.1.6.tar.gz. I have untar it in root directory(same as rubygems) and type command rails # script/server But, Nothing happens. Why? Please... Thank you very much Regards, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail. _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535 --------------------------------- How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call rates. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535 --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

1 0

[Users] Problema with PDT and DBTEXT
by Andrew And 07 Sep '06

07 Sep '06

I am using PDT and I try to use DB type DBTEXT but I have problem, COuld some help me*? Below I wrote debug screen and configuration.* 0(0) PDT: initializing... 0(0) DEBUG: register_fifo_cmd: new command (pdt_add) registered 0(0) DEBUG: register_fifo_cmd: new command (pdt_delete) registered 0(0) DEBUG: register_fifo_cmd: new command (pdt_list) registered 0(0) unixsock_register_cmd: New command (get_domainprefix) registered 0(0) find_mod_export: found <db_use_table> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_init> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_close> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_query> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_raw_query> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_free_result> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_insert> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_delete> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: found <db_update> in module dbtext [/lib/openser/modules/dbtext.so] 0(0) find_mod_export: <db_replace> in module dbtext not found 0(0) DBT:dbt_cache_get_db: looking for db /etc/openser/dbtext! 0(0) DBT:dbt_cache_get_db: database [/etc/openser/dbtext] does not exists! 0(0) DBT:dbt_init: cannot get the link to database 0(0) PDT: mod_init: Error while connecting to database 0(0) init_mod(): Error while initializing module pdt ERROR: error while initializing modules - Configuration: loadmodule "/lib/openser/modules/dbtext.so" loadmodule "/lib/openser/modules/pdt.so" modparam("pdt", "db_url", "dbtext:///etc/openser/dbtext") #modparam("pdt", "db_table", "pdt") modparam("pdt", "prefix_column", "code") modparam("pdt", "domain_column", "hostname") #modparam("pdt", "prefix", "00") modparam("pdt", "hsize_2pow", 4) modparam("pdt", "sync_time", 300) modparam("pdt", "clean_time", 600) - /etc/openser/dbtext: hostname(str) code(str) groupX:2121 groupY.voip.nce.ufrj.br:3131

2 3

[Serusers] log (history of request)
by Legos Engineering 07 Sep '06

07 Sep '06

Hi I wish know if a tools or a log exist in order to know, when an user's boxe is connected and send a request of identification. I want to do on the webcare, stats and know who switch off his boxe ( and know too the uptime). If it's possible, thank to tell me where is the tools to do this.

2 1

[Serusers] New polycom firmware / presence
by harrygaillac-sip＠yahoo.fr 07 Sep '06

07 Sep '06

Hello, I look at the new sip firmware however i don't undanstand the presence features. I don't use LCS but SER as presence server this one is able to provide a ressource list server and xcap server for sip buddies lists . Does polycom phones can suscribe to a sip:lists@domain.com for example to watch buddies status ? Regards Harry ___________________________________________________________________________ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com

1 0

[Serusers] How to integrate SER with Jabber IM.
by srinivas sriram 07 Sep '06

07 Sep '06

Hello, Can you please suggest me how to intergrate SER with Jabber IM. I am using SER 0.9.6 version of SER.Aiming to send a sip messages from the SER client to the Jabber Client. What are the changes to ser.cfg file and any other setting like Jabber.xml or MySql Database for SER intergation with Jabber IM.I am attaching my configuration file ser.cfg. Thank you, Regards, Sriram Srinivas.

1 0

[Serusers] How to integrate SER with Jabber IM.
by srinivas sriram 07 Sep '06

07 Sep '06

Hello, Can you please suggest me how to intergrate SER with Jabber IM. I am using SER 0.9.6 version of SER.Aiming to send a sip messages from the SER client to the Jabber Client. What are the changes to ser.cfg file and any other setting like Jabber.xml or MySql Database for SER intergation with Jabber IM.I am attaching my configuration file ser.cfg. Thank you, Regards, Sriram Srinivas.

1 0

[Serusers] VoN Boston
by Jiri Kuthan 07 Sep '06

07 Sep '06

If there are folks who are in VoN Boston next week and would like to meet, drop me an email. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Users] acc_rad_request problem
by Cseke Tamas 07 Sep '06

07 Sep '06

Hi, Could anybody tell me why is Acct-Status-Type radius attribute always 'Failed', when i send a radius accounting packet with acc_rad_request? I 'd like to send a packet, when an INVITE is received. And i also would like to disable somehow the accounting packet sent for the INVITE message, and sending just for ACK and BYE Because the INVITE packet is delivered late. and i need this radius packet in the same time when INVITE is received for billing. Thanks, Tamas

2 2

[Serusers] Re: Sip Proxy behind Nat.
by ravi reddy 07 Sep '06

07 Sep '06

Hi Greger, I found the exact problem which have been paining me. 1) the nated Sip-proxy; its sending the route headers as if it forgets it staying behind the Nat.andsending the call as some thing like this invite 99106883(a)81.21.33.35 ----this is my SER - server record<192.168.1.100>------- this is nated sip-proxy address contact<192.168.1.100>-----contact to this address only SER-server is frowarding the call to pstn Gateway as it recieves and just rewriting the host name. and the other problem(I think) 2) PSTN Gateway. even when call connects it is adding the strict route headers as it recieves from SER-server and when it sends 200 0k after connecting the call it is not getting back or SER-server is also not getting back the "ACK" from the nated proxy because the nated proxy itself add the contact header as (private-ip ) ofcourse it is following strictly the SIP-PROTOCOL so,these are the two problems i found , Some providers are providing calls to this nated sip-proxy -------> How ???? Why the SER-server is also not following the path where "INVITE" message came ????? How to stop the Nated Sip-proxy to stop sending the headers in "INVITE"message not including the record route or contact headers in the message --- ?????? Or is there any way to do in SER-server itself ( majorly prefered) not to contact private ip-address and to follow the next address which is NAT-IP ---??????? (I think its foolish question )---> Or is there any way to get registered another sip-proxy in location database.----??????? Please suggest me some thing to get out of this problem. Thank You. Regards, Ravi.

2 1

[Serusers] About the failure route problem
by AR. 07 Sep '06

07 Sep '06

Hi all I have a problme about the failure route My scenario is that: If voice gateway 1 is busy then forward the call to the voice gateway 2 My config is : route[5] { rewritehostport("voice_gateway_1:5060"); forward(uri:host, uri:port); t_on_failure("2"); if(isflagset(1)) append_branch(); setflag(1); t_relay(); break; } failure_route[2] { route(6); } route[6] { rewritehostport("voice_gateway_2:5060"); forward(uri:host, uri:port); break; } But it still doesn't work anyone help? Best Regards AR.

3 4

[Serusers] Cancel 183 progress (result of forked invite)?
by Vamsi Pottangi 07 Sep '06

07 Sep '06

Hi, In my test setup, I have SER as proxy and SEMS as announcement server. In SER config I'm sending forking invite to SEMS using append_branch for every call made. For example when User1 dials User2, SER send INVITES to both SEMS and User2, while the User2 phone rings, announcement is played from SEMS to User1 using the early media 183 session progress message. Once the User2 answers the call, a CANCEL message is sent to SEMS and normal conversation happens between User1 and User2 which is expected. The problem which I'm facing is that when User2 doesn't answer the call or if rejects the call, SER never sends CANCEL to SEMS, so the annoucement is being played continously to User1 and User1 is unaware of the remote party status. How can I configure the SER script to handle such unanswered or call reject cases and send CANCEL to SEMS and inform User1 about the unanswered or cancel status? Does it cancel 183 progress only in case of 200 OK from other invite session? how can we handle others like 408, 486 etc? I'm not pasting my ser.cfg here because it is the basic sample script with just append_branch("SEMS") in case of INVITE. Did anybody tried out such scenarios? Any help would be appreciated. Thanks, ~Vamsi

2 1

[Serusers] SER as a system service
by vijay tiwari 07 Sep '06

07 Sep '06

hi all i want to start SER as the system Service Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

2 1

[Serusers] CPU %
by Teran Jesus 07 Sep '06

07 Sep '06

Hello i am using the ser 0.8.12, it works fine, right now i run the linux comand ps -aux , to see the list of proces that are runing and i see that i have SER proces like the next, can you said me if it is correct or i have a problem with my ser or is a problem with the version of the SER, because sometimes the SER needs to reset because don't respond, if you see the last SER reset tht i need to made was August 25. USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 24416 32.4 0.7 38668 7456 ? R Aug25 5547:04 /usr/sbin/ser root 24421 32.4 0.7 38684 7468 ? R Aug25 5537:54 /usr/sbin/ser root 24426 32.4 0.7 38664 7484 ? R Aug25 5541:14 /usr/sbin/ser root 24431 32.4 0.7 38652 7468 ? R Aug25 5543:54 /usr/sbin/ser root 24437 0.0 0.8 38676 9056 ? S Aug25 5:56 /usr/sbin/ser root 24439 0.0 0.8 38756 9064 ? S Aug25 6:10 /usr/sbin/ser root 24444 32.4 0.7 38684 7456 ? R Aug25 5551:00 /usr/sbin/ser root 24450 32.4 0.7 38664 7472 ? R Aug25 5539:39 /usr/sbin/ser I ran the vmstat 1 10 too and i believe is not normall the processing, can you help me?. r b swpd free buff cache si so bi bo in cs us sy id wa 6 0 27440 17824 168972 649488 0 0 1 2 0 2 0 0 1 0 6 0 27440 17844 168972 649488 0 0 0 40 130 480364 12 87 1 0 6 0 27440 17844 168972 649488 0 0 0 0 118 475900 11 89 0 0 6 0 27440 17840 168972 649488 0 0 0 0 120 481403 11 87 2 0 6 0 27440 17840 168972 649488 0 0 0 0 114 483715 12 86 2 0 6 0 27440 17836 168972 649488 0 0 0 36 134 476488 12 87 1 0 6 0 27440 17840 168972 649488 0 0 0 64 137 482903 14 84 1 0 6 0 27440 17840 168972 649488 0 0 0 68 148 473984 14 86 0 0 6 0 27440 17840 168972 649488 0 0 0 64 143 476887 12 88 0 0 6 0 27440 17840 168972 649488 0 0 0 52 147 476060 12 88 0 0

2 1

[Serusers] 415 unsopported document format
by Javier Ramirez 07 Sep '06

07 Sep '06

Hi, I have a problem, when I send a PUBLISH with personal status, if have accent, the ser send me 415 unsopported document format. How to encode for ser accept my PUBLISH ? best regards Javier

2 5

[Serusers] alias_db module exists?
by Ricardo Carvalho 07 Sep '06

07 Sep '06

Is Ser capable to implement the same offered by alias_db module of OpenSer? Thanks, Ricardo.

2 1

[Users] acc database support
by Tracy Lofton 07 Sep '06

07 Sep '06

Hi all, I am new to OpenSER and I am trying to save acc data in mysql database but I keep getting an error whenever I dial the 976 prefix number. What do I need to do, to configure the database as there is a table for acc, and i am not quite sure of what to do now. Here is a snippet of openser.cfg file relating to the area in question: if (uri=~"^sip:919(00|76)[0-9]{7}@") { acc_db_request("403 - 900/976 Disabled", "acc"); xlog("L_INFO", "Cancel attempt to call 9xx number (not 911), db recorded\n"); exit; }; Here is the .cfg error displayed on the terminal when calling 9-1 and then the 976 prefix: 2(12906) ERROR:acc:w_acc_db_request: DB support not configured 2(12906) Cancel attempt to call 9xx number (not 911), db recorded 3(12908) ERROR:acc:w_acc_db_request: DB support not configured Any help greatly appreciated. Thanks, Tracy _________________________________________________________________ Get real-time traffic reports with Windows Live Local Search http://local.live.com/default.aspx?v=2&cp=42.336065~-109.392273&style=r&lvl…

2 1

[Serusers] acc database support
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi all, I am new to OpenSER/SER and I am trying to save acc data in mysql database but I keep getting an error whenever I dial the 976 prefix number. What do I need to do, to configure the database as there is a table for acc, and i am not quite sure of what to do now. Here is a snippet of my openser.cfg file relating to the area in question: if (uri=~"^sip:919(00|76)[0-9]{7}@") { acc_db_request("403 - 900/976 Disabled", "acc"); xlog("L_INFO", "Cancel attempt to call 9xx number (not 911), db recorded\n"); exit; }; Here is the .cfg error displayed on the terminal when calling 9-1 and then the 976 prefix: 2(12906) ERROR:acc:w_acc_db_request: DB support not configured 2(12906) Cancel attempt to call 9xx number (not 911), db recorded 3(12908) ERROR:acc:w_acc_db_request: DB support not configured Any help greatly appreciated. Thanks, Tracy _________________________________________________________________ Got something to buy, sell or swap? Try Windows Live Expo ttp://clk.atdmt.com/MSN/go/msnnkwex0010000001msn/direct/01/?href=http://exp…

1 2

[Serusers] how to stop SER
by Henry Duan 06 Sep '06

06 Sep '06

Hi, Sorry for my ignorance. I tried to stop SER, which is installed at RedHat Linux 9, however there is no ser.pid file(in /proc/ or /var/run/ directories). Can anybody suggest me a good way to stop SER so that I don't need to restart the machine to stop it? thanks!

2 1

[Users] Cancel 183 progress branch (result of forked invite)
by Vamsi Pottangi 06 Sep '06

06 Sep '06

Hi, In my test setup, I have Openser as proxy and SEMS as announcement server. In Openser config I'm sending forking invite to SEMS using append_branch for every call made. For example when User1 dials User2, Openser send INVITES to both SEMS and User2, while the User2 phone rings, announcement is played from SEMS to User1 using the early media 183 session progress message. Once the User2 answers the call, a CANCEL message is sent to SEMS and normal conversation happens between User1 and User2 which is expected. The problem which I'm facing is that when User2 doesn't answer the call or if rejects the call, Openser never sends CANCEL to SEMS, so the annoucement is being played continously to User1 and User1 is unaware of the remote party status. How can I configure the Openser script to handle such unanswered or call reject cases and send CANCEL to SEMS and inform User1 about the unanswered or cancel status? Does it cancel 183 progress only in case of 200 OK from other invite session? how can we handle others like 408, 486 etc? I'm not pasting my Openser.cfg here because it is the basic sample script with just append_branch("SEMS") in case of INVITE. Did anybody try out such scenarios? Any help would be appreciated. Thanks, ~Vamsi

3 2

[Serusers] CPU %
by Teran Jesus 06 Sep '06

06 Sep '06

> Hello i am using the ser 0.8.12, it works fine, right now i run the linux > comand ps -aux , to see the list of proces that are runing and i see that > i have SER proces like the next, can you said me if it is correct or i > have a problem with my ser or is a problem with the version of the SER, > because sometimes the SER needs to reset because don't respond, if you see > the last SER reset tht i need to made was August 25. > > > USER PID %CPU %MEM VSZ RSS TTY STAT START > TIME COMMAND > root 24416 32.4 0.7 38668 7456 ? R Aug25 5547:04 > /usr/sbin/ser > root 24421 32.4 0.7 38684 7468 ? R Aug25 5537:54 > /usr/sbin/ser > root 24426 32.4 0.7 38664 7484 ? R Aug25 5541:14 > /usr/sbin/ser > root 24431 32.4 0.7 38652 7468 ? R Aug25 5543:54 > /usr/sbin/ser > root 24437 0.0 0.8 38676 9056 ? S Aug25 5:56 > /usr/sbin/ser > root 24439 0.0 0.8 38756 9064 ? S Aug25 6:10 > /usr/sbin/ser > root 24444 32.4 0.7 38684 7456 ? R Aug25 5551:00 > /usr/sbin/ser > root 24450 32.4 0.7 38664 7472 ? R Aug25 5539:39 > /usr/sbin/ser > > I ran the vmstat 1 10 too and i believe is not normall the processing, > can you help me?. > > r b swpd free buff cache si so bi bo in cs us sy > id wa > 6 0 27440 17824 168972 649488 0 0 1 2 0 2 0 0 > 1 0 > 6 0 27440 17844 168972 649488 0 0 0 40 130 480364 12 87 > 1 0 > 6 0 27440 17844 168972 649488 0 0 0 0 118 475900 11 89 > 0 0 > 6 0 27440 17840 168972 649488 0 0 0 0 120 481403 11 87 > 2 0 > 6 0 27440 17840 168972 649488 0 0 0 0 114 483715 12 86 > 2 0 > 6 0 27440 17836 168972 649488 0 0 0 36 134 476488 12 87 > 1 0 > 6 0 27440 17840 168972 649488 0 0 0 64 137 482903 14 84 > 1 0 > 6 0 27440 17840 168972 649488 0 0 0 68 148 473984 14 86 > 0 0 > 6 0 27440 17840 168972 649488 0 0 0 64 143 476887 12 88 > 0 0 > 6 0 27440 17840 168972 649488 0 0 0 52 147 476060 12 88 > 0 0 > >

1 0

[Users] Aggregating SIP devices
by Kristian Kielhofner 06 Sep '06

06 Sep '06

Hello everyone, I have multiple Asterisk systems registered to an OpenSER server. Each of them have several different SIP accessible extensions. Using alias_db (or something), I need to be able to create contacts that are reachable from the OpenSER server to individual extensions on each Asterisk system. For instance: SIP Phone -> Asterisk System -> OpenSER The SIP phone is registered to the Asterisk system and has an extension (100). The Asterisk System is registered to OpenSER and has a usrloc entry (asterisk). I would like to setup an alias like this: asterisk-100 This alias should be able to dynamically get the location of the Asterisk system from usrloc and reach that extension directly, basically forming a SIP uri like this: sip://100@[asterisk server ip address - loaded dynamically from usrloc] That way, if I were to do this: External SIP Phone -> OpenSER -> Asterisk System -> SIP Phone (ext 100) I could "dial" asterisk-100 from that External SIP phone and get connected to the Asterisk System at extension 100. This would be similar to the following (in Asterisk land): [sip-in] exten => asterisk-100,1,Dial(SIP/asterisk/100,20) exten => asterisk-100,2,Congestion(10) The most obvious way to do this is create multiple subscriber entries for each Asterisk system and one register => line for each entry in Asterisk. I would rather not have to do that!!! Is this clear enough? What do you think? Thanks! -- Kristian Kielhofner

2 1

[Users] prevent a dialled username without dialling the domain to ring a phone registered with that username
by Ricardo Carvalho 06 Sep '06

06 Sep '06

How can I prevent a dialled username (without dialling the domain) to ring a phone registered with that username? I pretend to use a multi-domain architecture where apart from aliases of users, only dialling with username@domain will ring registered phones. Thanks, Ricardo.

2 1

[Users] Automatic INVITE when fone goes off-hook
by Mahesh Paolini-Subramanya 06 Sep '06

06 Sep '06

I know, not *directly* openser related, but hey, it is tangentially related I've got a scenario where some of my users want to 'hard-wire' their fones so that when their fone goes off-hook, it automatically sends an INVITE to a different (pre-specified) fone. And yes, they don't want to have to press a key on their fone (don't ask. i *know* its dumb. but hey, its what they want...) Has anyone seen *anything* along these lines? Any fones that do something similar? cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

2 1

Re: [Users] OpenSER Summit
by Rodrigo P. Telles 06 Sep '06

06 Sep '06

Hi Bogdan, I think its a very good idea, there are many people using OpenSER for their SIP solutions around the world including us here in Brazil. Unfortunately won't be possible for me to get VoN Berlin but I'm going to VON Boston on next 9-11 (uhhhh), perhaps I could meet the OpenSER members that are visiting or passing by Boston there or in another place. Does some one is going to VoN Boston and wants to have a beer with others OpenSER members? Best regards, -- ============================================ Rodrigo P. Telles <telles(a)devel.it> IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 Ourinhos: +55 14 3324-1200 São Paulo: +55 11 2124-7474 ============================================ > > Hello everybody, > > It's more than one year since the OpenSER project started. A large community of users and developers has been formed around OpenSER. It's a very active community and one of the most important goals of the OpenSER project is to keep the users community as synchronized as possible with the developers and management groups. > > During past time we met a lot of people that were happy to see us or we were happy to see them. So, we came to the idea of organizing a OpenSER Summit where OpenSER fans (all types of fans) should meet to share their experience, to discuss about future ways of developing the project or just to personally meet people from the mailing lists. The idea is to have people talking about how they are using OpenSER in different contexts and which were/might be their needs, development people talking about possible solutions and about their ideas of enhancing OpenSER, etc... > > We were thinking of taking advantage of the upcoming VoN in Berlin in the beginning of November and to organize the Summit within the same time frame - it will be more convenient for people to travel for two events at the same time. > > We would like to have some feedback from the community - how many people are interested in such a meeting, what will be their primary interests. > > Any other OpenSER-related ideas are most welcome! > > Best regards, > Bogdan

2 1

[Users] DNS Server
by Tracy Lofton 06 Sep '06

06 Sep '06

Hello All, I am new to OpenSER/Ser and have learned a lot and received a lot of assistance from all of you. Anyway, I would like to express my gratitude by extending Free Dynamic DNS services for anyone interested in utilizing a dynamic IP address in a static capacity while learning and using your OpenSER/SER sip router and/or Asterisk server. If interested go to: http://www.t4tm.net. While visiting the site review the details regarding the FREE Dynamic DNS Service, signup, add my name servers where you registered your domain name, download the dynamic IP updater client and that's it. I hope you all enjoy this service. Tracy _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

1 0

[Serusers] SIP client with video???
by Joao Pereira 06 Sep '06

06 Sep '06

Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software..... Does someone knows about a better software? Regards Joao Pereira

2 2

[Serusers] log ( history of action for one user)
by Legos Engineering 06 Sep '06

06 Sep '06

HI. I wonder, if there is some log (like messages or another) which permit to know the history of one user ( his last calls, when he was called ...). I want to know that in order to add, an history of each user in the webcare. Therefore, if someone can say me, if this log exist and where it is. Thank you.

2 1

[Users] How to redirect based on from_uri and uri
by Darren DeLitizia 06 Sep '06

06 Sep '06

Here is what I want to do. Domestic routing defining interlata and intralata based off of ani and dnis (from uri and uri). Now let's not dive too much into the whole domestic routing thing or we'll be here for the next century. Bottom line is, this what I want to do with openser. 1. Look at ani (from_uri) (NPA - first 3 digits) 2. Look at dnis (uri) - (NPA - first 3 digits) 3. If the NPA (first 3 digits) in ani is in same as the DNIS NPA then redirect 555dnis@ip_address 4. If ANI NPA is in same as DNIS NPA then redirect 999dnbis@ip_address I am redirecting so I can be and stay stateless I am able to redirect but unable to figure out correct syntax for it to look at ani and dnis then redirect appropriately. Thanks all

2 1

[Users] crash
by Helge Waastad 06 Sep '06

06 Sep '06

Hi, in my 1.1.0 version I syddenly got a core dump. Does any one know why? (gdb) bt #0 0x000000304448660b in strftime_l () from /lib64/tls/libc.so.6 #1 0x0000002a95b29de4 in time2mysql (_time=Variable "_time" is not available. ) at utils.c:54 #2 0x0000002a95b2a731 in val2str (_c=Variable "_c" is not available. ) at val.c:147 #3 0x0000002a95b26b13 in print_values (_c=0x643648, _b=0x2a95db5269 "'SIP/2.0 100 Trying\\r\\nVia: SIP/2.0/UDP 217.17.222.4;branch=z9hG4bK1957.c42a1af6.0\\r\\nVia: SIP/2.0/UDP 10.1.93.186:10001;rport=44323;received=217.17.222.6;branch=z9hG4bK2eb74daf34599095\\r\\nFrom: \\\"Har"..., _l=65431, _v=0x7fbfffe990, _n=10) at dbase.c:144 #4 0x0000002a95b27aa4 in db_insert (_h=0x643e08, _k=Variable "_k" is not available. ) at dbase.c:495 #5 0x0000002a97c26bdc in trace_sl_onreply_out (req=Variable "req" is not available. ) at siptrace.c:1078 #6 0x0000002a956b30ad in run_sl_callbacks (req=0x643f28, buffer=Variable "buffer" is not available. ) at sl_cb.c:88 #7 0x0000002a956b3608 in sl_send_reply (msg=0x643f28, code=100, text=0x608ee8 "Trying") at sl_funcs.c:149 #8 0x0000000000409c38 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:701 #9 0x000000000040a87e in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #10 0x000000000040b43d in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #11 0x000000000040b496 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #12 0x000000000040b762 in run_top_route (a=Variable "a" is not available. ) at action.c:89 #13 0x000000000043188c in receive_msg (buf=0x2a97e4d2f8 "\026", len=822, rcv_info=0x7fbffff810) at receive.c:155 #14 0x000000000044fc74 in udp_rcv_loop () at udp_server.c:465 #15 0x000000000041f7f4 in main_loop () at main.c:925 #16 0x0000000000420446 in main (argc=Variable "argc" is not available. ) at main.c:1477 -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 5

[Users] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

I'm using multi-domain and I can only call users by their aliases if I'm dialling from a phone registered with the same domain of the called user or if I'm dialling alias@domain_of_the_callee from a phone registered with other domain. How can I make Ser allow that every user can contact other users even if they are from other domain by just dialling their alias (without dialling also their domain when the callee is from other domain than the caller)? Regards, Ricardo.

2 3

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi users, This is ngrep report regarding my problem; -----------------<when a sip proxy behind NAT and the user registerd to that sip-proxy calls>------------------------- U 82.102.69.105:39871 -> 81.21.33.35:5060 INVITE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 361. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 82.102.69.105.---------------------------------------------> this is the NAT for the sip-proxy , t=0 0. m=audio 5004 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. -----------------------------<when my SIP-SERVER standing on public ip recieved the "INV" message from the above"----------------------------------- U 81.21.33.35:5060 -> pstngw:5060 INVITE sip:99106883@pstngw:5060 SIP/2.0. Record-Route: <sip:99106883@81.21.33.35 :5060;nat=yes;ftag=c86b66ad8b9187c8;lr=on>. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK0ab.9522bc25.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport=39871. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:82.102.69.105:39871>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 360. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 81.21.33.35.--------------------------------------------->This is my SIP-SERVER standing on public ip (and is sending to pstngw) t=0 0. m=audio 60516 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. ---------------------------------------------<and the pstngw is making call>----------------------------------------------- U 81.21.33.35:5060 -> pstngw:39871 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105;branch=z9hG4bK-d87543-ac3b034487ba16641188e4c9e5ad0664-1--d87543-;rport=39871,SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK01c64e769ba6c176. From: "12345"<sip:12345@81.21.33.35:5060>;tag=9ca5c1fe3b43aad6. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E6010EC-1764. Date: Wed, 06 Sep 2006 10:27:40 GMT. Call-ID: 72d0e4e8adca2ab2(a)192.168.1.102. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 32352 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 237. . v=0. o=CiscoSystemsSIP-GW-UserAgent 54 9902 IN IP4 pstngw s=SIP Call. c=IN IP4 81.21.33.35.-------------------------------------<in pstngw it noticed the contact header of SIP-SERVER>--------------------- t=0 0. m=audio 60518 RTP/AVP 18 100. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. a=ptime:20. ---------------------------------------------<after talking some time if any of UA hung the phone (here i am showing from pstn hung up) the result goes like this---------------- U pstngw:52991 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060;nat=yes;ftag=1bab08cdaad1d6e8;lr=on SIP/2.0. Via: SIP/2.0/UDP 81.21.38.15:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 6. Route: <sip:192.168.1.100:5060>, <sip:82.102.69.105:39871>.------------------------> What is happening here ??????? Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. . # U 81.21.33.35:5060 -> 192.168.1.100:5060--------------------------------------------------->here unexpected problem arises for me it have to use 82.102.69.105 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: <sip:81.21.33.35;ftag=7E66E3D8-1292;lr=on>. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bKb0f1.3fe32691.0. Via: SIP/2.0/UDP pstngw:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 5. Route: <sip:82.102.69.105:39871>. Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. P-hint: rr-enforced. So the call never ends up from SIP SERVER to other party Make some comments on above and assist me where am I going wrong: and i am using www.openser.org pstn default script as ser.cfg Hope to get some help Thanks, Ravi.

1 0

[Serusers] SER & Asterisk (IVR problem)
by Roberto Lopes 06 Sep '06

06 Sep '06

Hi to all! I've a big problem with SER & Asterisk I've a network where i put them working together. Asterisk is the media gateway with PSTN as well as being the server which is making all the services available (IVR, voicemail, conference calls, etc). SER just routes calls either to the softphones, VoIP phones or to Asterisk who will just forward the calls to PSTN (via Asterisk). Till now everything works well except the service IVR which is working on Asterisk. Details: 192.168.226.13 - IP SER 192.168.226.45 - IP Asterisk The problem is only after the user (roberto(a)192.168.226.13 - authenticated and authorized in RADIUS) calls IVR. Till then the user calls, IVR gives some information to the user and everything goes well. Problem is when he presses a number which will have to redirect the call to another user. And why is there a problem. Because the initially roberto(a)192.168.226.13 to be able to call IVR had to rewrite host & port 192.168.226.45 so that the call would reach Asterisk who would then call to the IVR. That rewriting turned roberto(a)192.168.226.13 into roberto(a)192.168.226.45. That is, I can say, that it became a "new user". While it's on the IVR there's no problem, but when call is diverted it has to give its credentials... Credentials that don't exist in RADIUS cause it only accepts usernames that end in @192.168.226.13 (IP of SER) Here is the following code, both of extensions file in Asterisk and configure file of SER I hope somebody knows how to solve it or at least find a way how to. extensions.conf - Asterisk [sip] exten => 74001,1,Answer exten => 74001,2,SetMusicOnHold(default) exten => 74001,3,Set(TIMEOUT(digit)=5) exten => 74001,4,Set(TIMEOUT(response)=10) exten => 74001,5,Background(menu) exten => 1,1,Dial(SIP/roberto@192.168.226.13:5060,,r) ... exten => 7,1,Goto(sip,74001,1) exten => 8,1,Hangup ser.cfg - SER ... if ((uri=~"^sip:7[0-9]{4}@.*")) { route(5); break; }; ... route[5] { # Redirecting to Asterisk # MsgLog xlog("L_ALERT","(WARNING) Sending to Asterisk request of %fu to %tu\n"); rewritehostport("192.168.226.45:5060"); route(1); } Thanks for your help, Roberto Lopes

2 1

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi Users, I am using SER-0.9.6 on public domain . and I am using another Sip-proxy behind NAT (ip-address=192.168.1.100:5060) i have some clients registered to that nated sip-proxy and here that nated sip-proxy forwards PSTN calls to SER on public domain. when call make from client (registered to nated sip-proxy) the session establishing suscessfully , but when the client hung the phone the SER on public domain is taking the BYE message and is not sending the BYE message to the other end and it is sending directly to the Sip-proxy which is a private-ip . So, here SER is not getting ACK and its keep on sending the BYE requests to the privateip instead of sending to the NAT address . In some docs which jiri wrote on record route he mentioned the same problem below is the link ; but he dint mention how to solve that record route problem, If SER send messages back as it recieving path (from NAT)there is no problem , but its changing the route to send BYE , For the request "CANCEL","ACK","INVITE","200 OK" every thing is happening in the same Route but it is not happening only for the "BYE" request . Hope some body can help me. Thank You. here is the link which i told;-http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/…<http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin…> Regards, Ravi.

1 0

[Serusers] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

I'm using multi-domain and I can only call users by their aliases if I'm dialling from a phone registered with the same domain of the called user or if I'm dialling alias@domain_of_the_callee from a phone registered with other domain. How can I make Ser allow that every user can contact other users even if they are from other domain by just dialling their alias (without dialling also their domain when the callee is from other domain than the caller)? Regards, Ricardo.

2 3

Re: [Users] more on accounting
by tele 06 Sep '06

06 Sep '06

Glenn Dalgliesh wrote: >This is from previous list > >"i Glenn, > >For this feature to work, you need to be sure that you do record_route() on >the INVITE and that "append_fromtag" RR module param is not disable. > >regards, >bogdan " > > Yes i have read that thread. i always record_route all messages before do things. if (!method=="REGISTER") record_route(); by the sample openser.cfg and for be sure i have also forced: modparam("rr","append_fromtag",1) >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf >Of tele >Sent: Wednesday, June 28, 2006 11:05 AM >To: users(a)openser.org >Subject: [Users] more on accounting > >Hi all, > >I'm using OpenSER v1.0.1 and i have done an accounting compatible with >my billing system. >Now i have problem with detection of flow direction for billing reason. > >i'm talking about openser 1.0.x: > >the problem is when the BYE come from callee. >i know the problem is solved in openser1.1.x with the > >modparam("acc", "detect_direction", 1) > >i have read in modules documentation of 1.0.x that i can use >is_direction() of RR module for detect call flow direction but it says >that this must be called after loose_route() > >so for the accounting i must call setflag before loose_route() but what >can i do if i want detect call flow >before of exec setflag? > >Hmmm i think that i must install OpenSER 1.1.x and test with it > >:tele > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

5 21

[Serusers] Media Proxy and 2 interfaces (Private and Public)
by Kyriakos 06 Sep '06

06 Sep '06

Hello, Is there a way to have a machine that hosts media proxy have its one interface in a private network and the other on a public? >From what I read rtpproxy supports bridging which is basically what I want to do but I have not found anything for mediaproxy. Thanks KM

2 1

[Serusers] Sip Proxy behind Nat.
by ravi reddy 06 Sep '06

06 Sep '06

Hi Users, I am Using SER-0.9.6 on a public ip and using another sip-proxy(not SER) behind the NAT. and when the users who registerd to Sip-proxy which is behind NAT,, calls to the users who registered to public domain SER.. call is being established . but when hunging the phone the SER is not sending "BYE" messages to the NAT device it is sending to the Sip-proxy (not SER) that is for a private IP . so even when i hung the call from the Nated Sip proxy ,SER is getting messages through NAT but again SER is not sending "BYE" messages to other party its sending back to the nated sip-proxy having private IP , so the messages are going in a wrong direction and is being wasted . It looks like this is a BUG in SER , i would take this oppurtunity to report this. If this is not a bug please send me the suggestions ..... To solve my problem. thank you. regards, Ravi.

2 1

[Serusers] server error message " 500 I'm terribly sorry, server error occurred (1/SL)".
by ravi reddy 06 Sep '06

06 Sep '06

Hi users , I am proceeding in this way i have a SER-0.9.6 running fine :-) and I had given a username and password to a call-shop and this callshop owner with his username and password he connects to another 6 phones, Actually he bills to his 6 phones and I will bill him ."o.k overall Scenario is well and good". I made a little changes in ser.cfg and when the call made from the call shop up to call connecting is O.K but when we hung the phone the SER is not generating "BYE" messages to other party , so the call is on.. and i am not getting "Acct Stop" packet also SO How I can solve my problem :-( any suggestions will be appreciated . below is the message i am getting from SER when I hung the phone on one side -------------------<this message is came from callshop Nat address>---------<"it sends bye to my SER "------------------------------------------------------ U 82.102.69.105:32768 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Route: <sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a>. Route: <sip:192.168.1.100:5060>. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Proxy-Authorization: Digest username="12345", realm="81.21.33.35", algorithm=MD5, uri="sip:192.168.1.100:5060", nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde", response="4001812db780010a557b40683efd2f9e". Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 0. . ----------------<here it is as soon as SER recieve Bye message it has to send Bye to other party >------<But it is sending to the hung up phone itself>--------- # U 81.21.33.35:5060 -> 192.168.1.100:5060 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: <sip:81.21.33.35;ftag=837e5b2ff0b4cf4a;lr=on>. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK936f.7ff7572.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Proxy-Authorization: Digest username="12345", realm="81.21.33.35", algorithm=MD5, uri="sip:192.168.1.100:5060", nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde", response="4001812db780010a557b40683efd2f9e". Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 0. . -----------------<And here I go iam getting this message and the call is not being stopped >---------------------------------- # U 81.21.33.35:5060 -> 82.102.69.105:32768 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768;received= 82.102.69.105. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27773 req_src_ip= 82.102.69.105 req_src_port=32768 in_uri=sip:99106883@81.21.33.35:5060 out_uri=sip:192.168.1.100:5060 via_cnt==2". . any suggestions will be appreciated: Thank You.

2 1

[Serusers] Record Route problem :-(
by ravi reddy 06 Sep '06

06 Sep '06

Hi SER Users, I am using SER-0.9.6 on public domain . and I am using another Sip-proxy behind NAT (ip-address=192.168.1.100:5060) i have some clients registered to that nated sip-proxy and here that nated sip-proxy forwards PSTN calls to SER on public domain. when call make from client (registered to nated sip-proxy) the session establishing suscessfully , but when the client hung the phone the SER on public domain is taking the BYE message and is not sending the BYE message to the other end and it is sending directly to the Sip-proxy which is a private-ip . So, here SER is not getting ACK and its keep on sending the BYE requests to the privateip instead of sending to the NAT address . In some docs which jiri wrote on record route he mentioned the same problem below is the link ; but he dint mention how to solve that record route problem, If SER send messages back as it recieving path there is no problem , but its changing the route to send BYE , For the request "CANCEL","ACK","INVITE","200 OK" every thing is happening in the same Route but it is not happening only for the "BYE" request . Hope some body can help me. Thank You. here is the link which i told;- http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin… Regards, Ravi.

1 0

[Users] t_forward() functionality with t_relay()?
by Greg Fausak 06 Sep '06

06 Sep '06

I've got a dumb question. Can I get t_relay() functionality (transactions and whatnot) without getting the Via and potentially Record-Route tags in the header??? I warned you it was dumb.... -g -- Greg Fausak greg(a)thursday.com

3 4

[Users] Error in installing Ruby...Please help...Please
by Ferianto siregar 06 Sep '06

06 Sep '06

Dear all, Thank you very much for your help. A few days ago, I have sent a message that to ask about openser admin installation. Thank you very much for your help. Thanks Now, I have tried to install it again. But, When I compile the rubygerms-0.9.0.tar.gz, I got error message. This is the task that I have done before: 1. Untar the rubygerms packet # tar -zxvf rubygems-0.9.0.tar.gz # cd rubygems-0.9.0 # ruby setup.rb # gem install --include-dependencies When I type this command, "gem install --include-dependencies", I got error message. This is the error message: Attempting local installation of 'rails' Local gem file not found: rails*.gem Attempting remote installation of 'rails' Updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Gem::RemoteSourceException) Error fetching remote gem cache: getaddrinfo: Temporary failure in name esolution What`s wrong? I do hope anybody can help me to solve this problem. Please help me...Please help because of my lack skills..Please Note: 1. I do not have connection through the internet, so for rails packet, I have downloaded the rails-1.1.6.tar.gz. I have untar it in root directory(same as rubygems) and type command rails # script/server But, Nothing happens. Why? Please... Thank you very much Regards, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail.

2 1

[Serusers] NAT Solution
by Asep Saifudin 06 Sep '06

06 Sep '06

Hi, Does anyone can help me to choose best NAT solution for SER/OpenSER ?NatHelper or Mediaproxy ? And if they compare with STUN, TURN, ICE technology Nathelper or Mediaproxy better than them ? My target is I want my SIP Server can pass all NAT/Firewall just like SkyPE which hard to be block by firewall and can pass all NAT system. ~Asep

2 1

[Serusers] SER not responding
by Henry Duan 06 Sep '06

06 Sep '06

hi, I'm new to SER. I downloaded SER0.9.6 source to my local Fedora Core 5. I run make all and make prefix=/ install, and then set SIPDOMAIN, and run ser. However it didn't respond with incomging Registration(it showed incoming Registration from Ethereal). (I used default ser.cfg). Can anybody tell me what's the problem, or how to debug? Thanks so much! Henry

3 2

[Users] OSS on openSER and Redriect .
by raviprakash sunkara 06 Sep '06

06 Sep '06

Hi Users, Can any one tell about OSS ( Operation Support System) on openSER, Why we need ? But i think to reduce the load balance on openSER server, Suppose is I use 3 OSS's on SIP server. Sip users should also insert in to 3 OSS sip. And other doubt is ? Suppose if we using the redirect , that forwording into other openser system, When openSER redirect to other openSER, the user should also insert, But i think that .... OSS and redirect on openSER are the same. Please for me in English, I know its naughty questioin, please excuse me ! -- -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Serusers] Free DNS Service
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi All, I am new to OpenSER/SER but have been receiving quite a bit of help and would like to express my gratitude by letting you all know of the FREE DNS Service that I am offering to the Internet Community. Go to http://www.t4tm.net to find out more, signup and enjoy the benefits. Tracy _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

1 0

[Serusers] Preregistering a sip address?
by Ignacio Mas Ivars 06 Sep '06

06 Sep '06

Hi all! I have just started using the SIP Express Router and I am looking for a way to have a user preregistered when the router starts. Is there any way to configure that user1 should be registered by default at an IP address without needing to send the registration? I guess it will have to do with messing a bit with the routing engine, but I don't dare to touch it so far and I don't know how to do it. Any help would be very much appreciated! Thanks very much for any help! /Nacho --- ,,, (o o) --ooO--(_)---Ooo----------- Three may keep a secret, if two of them are dead.- --oo0--------0oo-----------

2 1

[Serusers] Free DNS Service
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi All, I am new to SER/OpenSER and I have received much help and in return, I am offering free Dynamic DNS services to whoever is interested. To find out more go to http://www.t4tm.net, signup and enjoy the benefits. Tracy _________________________________________________________________ Got something to buy, sell or swap? Try Windows Live Expo ttp://clk.atdmt.com/MSN/go/msnnkwex0010000001msn/direct/01/?href=http://exp…

1 0

[Serusers] delay to write things in DB when using serctl
by Ricardo Carvalho 06 Sep '06

06 Sep '06

When I add with serctl some alias to the DB, only after some time it appears written in the aliases table of DB. How can I force that to be simultaneously written in DB? Any parameter I have to tune in ser.cfg? Regards, Ricardo.

2 1

[Users] Re: Request for discussion: t_relay() internal error processing
by Mark Kent 05 Sep '06

05 Sep '06

>> 2) t_relay will return several error codes: >> -1 = no transaction created -> need to use sl_* functions >> -2 = transaction created, relay failed -> may destroy If I understand this correctly, the difference between these two return codes really has to do with who it is that cares about it... -2 relates to this exact call, at this exact moment, chances are the very next call the openser box handles will be OK. It looks to me like -1 means serious trouble, for the entire call switching platform. It sounds like it could happen because of a resource shortage (memory), possibly because of a bug in openser, or a bug in the openser.cfg. If you get -1 once, then the chances that you'll get the same code on the next call are very high. Let's say we've run out of memory. Then, what tools are available at the openser.cfg script level that can handle this? Is there a way to force some internal garbage collection? Or to do a warm-restart (e.g., trash all transactions, but not registrations)? My point here is this: if I'm right about -1 meaning serious trouble that is likely to mean serious problems from now on, then this is a more general problem than just t_relay, and much bigger than just handling this one call or a call to one particular neighbour. It's true that the application may halt and catch fire, but there are plenty of errors that are catchable by openser and the larger question seems to be: what do we do then? If there were script-level tools that could be used to try to manage the failure, possibly clean up (in a large way) and try to keep going, then maybe we should have openser_failure route blocks that the user can register with openser to handle various errors. Thanks, -mark

2 1

[Users] Request for discussion: t_relay() internal error processing
by Bogdan-Andrei Iancu 05 Sep '06

05 Sep '06

Hi all, there are some topics largely discussed lately on the mailing list; in order to fix them problems first we need to find a solution for logical point of view - and here I think that several opinions are better than one :) In order to have failover based on serial forking, it is mandatory to be able to detect from script if the initial t_relay() fails. The problem is that now t_relay() masks any error by internally generating an error SIP reply. Of course, it can be internal errors (like lack of memory), msg errors (parse errors) or transport errors (DNS failed, send/connect failed, etc). Now, disabling the auto replying in case of error is not a problem. The problem is how to handle the errors from the script; two make distinction between the error classes a listed above will not be the purpose of this email ( I will open another one for this ;) ). What is important here is to be able to properly resume from the point were the error popped up. What I mean is: the t_relay() function builds the transaction and sends the requests. If the error occurred during transaction creation, from script, you will had to use only stateless functions to send err. replies. If the transaction was already created and the error popped up during relaying stage, you will need from the script to take care of the transaction and either destroy it, either reply it. Based on this, here is what I suggest: 1) have a new param for disabling the auto reply in t_relay() 2) t_relay will return several error codes: -1 = no transaction created -> need to use sl_* functions -2 = transaction created, relay failed -> may destroy transaction, reply via TM with error, trigger failover. any comments on this ? regards, bogdan

4 4

[Users] mediaproxy error
by Richard Z 05 Sep '06

05 Sep '06

Hi, I tried the lateset mediaproxy. When starting, got these errors and dispatcher won't start, /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/proxydispatcher.py", line 65, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 39, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/accounting.py", line 14, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/mysql.py", line 8, in ? #ls /usr/local/mediaproxy/modules/ accounting.py daemon.py dispatcher.py mysql.py request.py serialize.py version.pyc accounting.pyc daemon.pyc dispatcher.pyc mysql.pyc request.pyc utilities.py configuration.py datatypes.py external options.py rtphandler.py utilities.pyc configuration.pyc datatypes.pyc formatting.py options.pyc rtphandler.pyc version.py Any suggestion? Thanks, Richard

1 0

[Serusers] No calls shown in serweb under accounting tab
by Chahn John Kim 05 Sep '06

05 Sep '06

Sorry for double posting. I forgot to put subject line in previous posting. Hi, all. I just installed serweb and it is working however there is are no calls shown under "accounting" tab, saying "no calls". Mysql table has call records with matching flag set in ser.cfg. Records seem to be correct for numerous test calls made. It just that serweb doesn't seem to pull those records out of mysql at all. Any idea where I should look at? Thanks in advance. -- Chahn John Kim

1 0

[Serusers] (no subject)
by Chahn John Kim 05 Sep '06

05 Sep '06

Hi, all. I just installed serweb and it is working however there is are no calls shown under "accounting" tab, saying "no calls". Mysql table has call records with matching flag set in ser.cfg. Records seem to be correct for numerous test calls made. It just that serweb doesn't seem to pull those records out of mysql at all. Any idea where I should look at? Thanks in advance. -- Chahn John Kim

1 0

[Serusers] prevent a dialled username without dialling the domain to ring a phone registered with that username
by Ricardo Carvalho 05 Sep '06

05 Sep '06

How can I prevent a dialled username (without dialling the domain) to ring a phone registered with that username? I pretend to use a multi-domain architecture where apart from aliases of users, only dialling with username@domain will ring registered phones. Thanks, Ricardo.

2 1

[Users] Proxy to Proxy Communication
by shiva sankar padhy 05 Sep '06

05 Sep '06

Dear Members! I am very new to This VOIP Technology. I have just installed the Openser. I am now able to Do the Communication between the users who are resisterd in the proxy but how do I establish the calls to another proxy.I mean Lets consider I have two Proxies and I want a user of Proxy 1 to coomunicate with a user of Proxy2.How will I edit the configuration files so that I can achieve that. And One more thing.How do I achieve the Nat Traversal in openser.Let us consider I am running my Server in 24.10.65.90(Public and static).and I have a Lan connection In the client side, When the two phones from the same LAN want to talk with each other by using the proxy at 24.10.65.90, Its not possible.I got Disconnected from the internet connection and also when I tried this.If I am calling to a Phone other than this LAN using proxy at 24.10.65.90, I am able to establish.What may be the problem to the previous. I hope I have explained my problems in a user Understandable way. I am expecting a sooner reply. Thanking you In Advance, Shiva

2 1

[Users] Freeradius dictionary additions..
by nick 05 Sep '06

05 Sep '06

When I add my dictionary.openser to Freradius v 1.0.1-3 on Centos 4.3, and after following all the steps outlined on the webpage: http://www.openser.org/docs/openser-radius-1.0.x.html When I get to testing the configuration, radiusd -X i get this: read_config_files: reading dictionary Errors reading dictionary: dict_init: No ATTRIBUTE "Sip-Method" defined for VALUE "Other" Errors reading radiusd.conf Unfortunately freeradius.org is now a parked domain (seems like a rather large oversight on their part...) so I can't ask them.. Has anyone here ever seen an issue like this? BTW, I'm using openser 1.1.0-notls compiled from source. Not that seems to apply here though.

6 10

[Serusers] DNS SRV Query's
by Ricardo Martinez 05 Sep '06

05 Sep '06

Hello list. I was doing some debugs on my SER machine and i found this these DNS query's: 9.381051 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 9.390312 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 9.390559 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 9.399772 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.337343 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _mediaproxy._tcp.sipproxy.mydomain.net 11.339614 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.345067 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 11.349127 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.349584 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 11.358696 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 17.247771 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 17.253948 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 17.254042 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 17.263298 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 18.988320 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 18.998219 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 18.998329 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 19.008071 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name What modules is generating this querys' and why ? Thanks Ricardo

2 1

[Users] Mediaproxy with t38
by Leonid Fainshtein 05 Sep '06

05 Sep '06

Pavel, I think you have to check if your VoIP gateways can send UDPTL "no-signal" messages. Sometimes the caller gateway doesn't send any messages until it receives CED from callee. Because of that neither MediaProxy nor RTPProxy can't complete the ip/port binding for the media session. Also, check which ports are used for UDPTL session. If they are differ from the ports that were used during the previous voice (RTP) session than you probably have to re-establish mediaproxy session. At least it is true for RTP Proxy that I use. For example, I call unforce_rtp_proxy() and then force_rtp_proxy() on each re-INVITE. To be honest, I have never tried the MediaProxy but I suppose that it works in the similar way like the RTP Proxy. Best regards, Leonid Fainshtein Product Manager IP Gear Ltd., a division of Qualmax "Get your VoIP network "In Gear" with Claro gateways from IP Gear" e-mail: LFAIN(a)ipgear.net Tel: +972-4-9907-525 Fax: +972-4-9990-334 www.ipgear.net

1 0

[Users] Not Hung uping...
by raviprakash sunkara 05 Sep '06

05 Sep '06

Good Morning, In local network openser is working fine...... In Behind the NAt the voice is also fine.... But problem is Not hung. upping.... I think problem in Record router header.... Actually it should reply as " Record Router < sip: public.ip:...... > " , But for me it reply as " Record Router <sip: private.ip.... > " That why i think Bye request is not reaching to openser server and its not hung upping.... and my guess is router(firewall) causing this problem , But voice is fine.... Below one is log file.... sip_domain= public_ip It s not hung upping....... openser server system ip 192.168.2.75 And UAC are outside the NAT The media is fine..... but not hung upping at a same time..... I think the record router is not correct... which is red color marked one...... 59.93.74.129:41010 -> 192.168.2.75:5060 INVITE sip:9001@public_ip SIP/2.0..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@6 1.17.248.68>;tag=587704647..To: <sip:9001@public_ip>..Contact: < sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2. 34..CSeq: 35636 INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 3 5180906 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=00..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a =rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2.34 ..CSeq: 35636 INV ITE..Server: OpenSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.75:5060 "Noisy feedback tells: pid=4700 req_src_ip=59.93.74. 129 req_src_port=41010 in_uri=sip:9001@public_ip out_uri= sip:9001@59.93.74.129:5060 via_cnt==1".... # U 192.168.2.75:5060 -> 59.93.74.129:5060 INVITE sip:9001@59.93.74.129:5060 SIP/2.0..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b 487.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public _ip>;tag=58770464 7..To: <sip:9001@public_ip>..Contact: <sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..M ax-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 35180906 IN IP4 59.93.74.1 29..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtp map:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;br anch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: < sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2.34 ..CSeq: 3 5636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;b ranch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: < sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip >;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647 >..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>.. Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129 :5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75 ;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>. .Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129 :5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] 415 unsopported document format.
by Javier Ramirez 04 Sep '06

04 Sep '06

Hi, I have a problem, when I send a PUBLISH with personal status, if have accent, the openser send me 415 unsopported document format. How to encode for openser accept my PUBLISH ? best regards Javier

2 1

[Users] How to configure multi-domain?
by Ricardo Carvalho 04 Sep '06

04 Sep '06

Besides from adding the following lines in configuration file and adding 2 diferent domains to the domain table, what else should I do to implement multi-domain? modparam("usrloc", "use_domain", 1) modparam("registrar", "use_domain", 1) My phones still register, but none of them can call any other, even if I call someone only by username or by username@domain. Regards, Ricardo.

2 1

[Users] run_failure_handlers warning message
by Federico Giannici 04 Sep '06

04 Sep '06

Since we upgraded to 1.1.0 from time to time the following warning is logged: WARNING:tm:run_failure_handlers: no UAC or CANCEL support (1, 2) Is it "normal" or does it indicate some problem some were? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 3

[Users] redirect and proxy functionality
by Mark Kent 04 Sep '06

04 Sep '06

I have a clash of concepts in my head, and it's messing with my openser config. 1) On one hand I think that if a phone is configured to use an outbound proxy then it should use it. Period. 2) On the other hand, I now do a "300 Redirect" and the phone sends back, to the same box, the same Invite but with the redirected r-uri. This is consistent with 1), but in this case I don't want this behaviour... If things are supposed to work like 2) then why wouldn't I just rewritehostport() and t_relay it without bothering with the redirect? In any case, I don't want to do that (rewritehostport), I want the redirect to be "try over there, and don't bug me again." I don't want to relay the dialog. So, in the real world, what should I hope to get out of a redirect (3xx) code? Thanks, -mark

2 1

[Users] uac_redirect out of memory?
by Greg Fausak 04 Sep '06

04 Sep '06

We've been doing a little bit of stress testing on the auc_redirect module. Throwing 60 calls per second at it. After quite a while I start getting error messages like: Sep 1 15:33:43 www openser[74528]: ERROR: sip_msg_cloner: cannot allocate memory Sep 1 15:33:43 www openser[74499]: redirect.g2.sbsqa2.com-LOOSE: time_t=1157142823 ^LINE=47 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=5723-28564(a)10.88.129.253 ^cseq=3 ^contact=<sip:18001002132@10.88.129.253> ^from=sip:+18001002132@10.88.129.253 ^fromtag=18001002132 ^to=sip:+16661012126@10.88.128.116 ^totag=5724 ^method=BYE ^ruri=sip:+116661 012126@10.88.129.251:5060 ^messageid=263458^remark=Doing it Sep 1 15:33:43 www openser[74528]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74592]: redirect.g2.sbsqa2.com-LOOSE: time_t=1157142823 ^LINE=47 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=5724-28564(a)10.88.129.253 ^cseq=3 ^contact=<sip:18001002133@10.88.129.253> ^from=sip:+18001002133@10.88.129.253 ^fromtag=18001002133 ^to=sip:+16661012127@10.88.128.116 ^totag=5725 ^method=BYE ^ruri=sip:+116661 012127@10.88.129.251:5060 ^messageid=263192^remark=Doing it Sep 1 15:33:43 www openser[74528]: ERROR: t_newtran: new_t failed Sep 1 15:33:43 www openser[74499]: ERROR: sip_msg_cloner: cannot allocate memory Sep 1 15:33:43 www openser[74592]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74499]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74592]: ERROR: t_newtran: new_t failed in the log file. I have seen similar problems with regular t_relay when we approach 200 calls per second. Is this a shared memory problem? How much shared memory to I need to set up? Have a great weekend! ---greg -- Greg Fausak greg(a)thursday.com

3 3

[Users] Not Hung uping
by raviprakash sunkara 04 Sep '06

04 Sep '06

Hi.... please forgive me in egnlish...... Below one is log file.... sip_domain= public_ip It s not hung upping....... openser server system ip 192.168.2.75 And UAC are outside the NAT The media is fine..... but not hung upping at a same time..... I think the record router is not correct... which is red color marked one...... 59.93.74.129:41010 -> 192.168.2.75:5060 INVITE sip:9001@public_ip SIP/2.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@6 1.17.248.68>;tag=587704647..To: <sip:9001@public_ip>..Contact: < sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2. 34..CSeq: 35636 INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 3 5180906 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=00..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a =rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INV ITE..Server: OpenSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.75:5060 "Noisy feedback tells: pid=4700 req_src_ip=59.93.74. 129 req_src_port=41010 in_uri=sip:9001@public_ip out_uri= sip:9001@59.93.74.129:5060 via_cnt==1".... # U 192.168.2.75:5060 -> 59.93.74.129:5060 INVITE sip:9001@59.93.74.129:5060 SIP/2.0..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b 487.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=58770464 7..To: <sip:9001@public_ip>..Contact: <sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..M ax-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 35180906 IN IP4 59.93.74.1 29..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtp map:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;br anch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 3 5636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;b ranch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip >;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647 >..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] another permissions documentation issue
by Juha Heinanen 04 Sep '06

04 Sep '06

http://www.openser.org/docs/modules/1.1.x/permissions.html does not include allow_refer_to(), but does include allow_uri(), which is not part of release 1.1.0. -- juha

2 1

[Users] ENUM
by Andrew And 04 Sep '06

04 Sep '06

I have problem with ENUM, below I am showing my DNS configuration (Bind version 9): $ORIGIN 1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server2.com!" . When I use DIG "1.8.8.5.2.1.2.3.3.e164.arpa" or " 1.8.8.5.3.1.2.3.3.e164.arpa", I don´t have problem. But When I try "1.8.8.3.2.1.2.3.3.e164.arpa" DIG don´t obtain resolution. I see that I need to add in DNS the tree between 1.2.3.3 and 8.5.2.1.2.3.3: $ORIGIN 2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . I have a lot of servers and this is not scalable, then I can´t use the tree between 1.2.3.3 and 8.5.2.1.2.3.3. Is there another configuration that I don´t need to use the tree between 1.2.3.3 and 8.5.2.1.2.3.3?

2 1

[Users] Record router problem in NAT RTPproxy.
by raviprakash sunkara 04 Sep '06

04 Sep '06

Hi Users , Good Morning, In local network openser is working fine...... In Behind the NAt the voice is also fine.... But problem is Not hung. upping.... I think problem in Record router header.... Actually it should reply as " Record Router < sip: public.ip:...... > " , But for me it reply as " Record Router <sip:private.ip.... > " That why i think Bye request is not reaching to openser server and its not hung upping.... and my guess is router(firewall) causing this problem , But voice is fine.... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Serusers] SER ,asterisk registration
by Siqhamo Sifo 04 Sep '06

04 Sep '06

How do I let clients registered on SER make calls through asterisk and virse versa

2 1

[Users] Mediaproxy with t38
by Pavel D. Kuzin 04 Sep '06

04 Sep '06

Hello! Somebody using mediaproxy with t38? Can you provide me some configuration examples? Thank you. -- Pavel D.Kuzin System Administrator Nodex ISP St. Petersburg, Russia pk(a)nodex.ru http://nodex.ru

5 7

[Serusers] UA, SER, RTPPROXY in local net, some external clients with public ip
by Rustam Yusupov 04 Sep '06

04 Sep '06

Hello, serusers. I am newbie for SIP. I installed ser, rtpproxy and read manual very many times :-), but... Ser server and some clients in local network : ser - 192.168.100.1 UA1 - 192.168.100.2 UA2 - 192.168.100.3 Some clients have public ip address, not behind firewall... All work fine between local clients, but not work with external clients. I think that local clients register with local ip in Contact field from local network and cannot be fix with fix_nated_ function, and so can't access from external... Simple config or config for rtpproxy from gettingstart not help me... Can anybody help me with simple config for this network configuration? -- Rustam mailto:ru@aspec.ru

2 1

[Serusers] Final ACK problem
by Abdul Qadir 04 Sep '06

04 Sep '06

Hi I am trying to setup SER with nathelper module and portaone proxy. All other messages are sent on the nated IP of both user agents, except final ACK. Which is sent on private IP of called party. Here is the sequence .... I have also pasted my ser.cfg file My ser is running on public IP and both user agents are behind firewall. UA1 SER UA2 INVITE ===============> INVITE ===============> TRYING <=============== Ringing <=============== Ringing <=============== OK <=============== OK <=============== ACK ===============> ACK (sent on UA2's Private IP) ===============> debug=7 fork=yes log_stderror=yes listen=some_public_ip port=5060 check_via=no dns=no rev_dns=no server_signature=no sip_warning=no fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/xlog.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/exec.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("nathelper","rtpproxy_sock", "udp:some_public_ip:port") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 0) modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 30) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # ------------------------- request routing logic ------------------- # main routing logic route { if(!mf_process_maxfwd_header("10")) { log(1, "too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if(msg:len >= max_len ) { log(1, "message too big\n"); sl_send_reply("513", "Message too big"); break; }; if(loose_route()) { t_relay(); break; }; if(uri==myself) { if(method=="REGISTER") { fix_nated_register(); force_rport(); if(!save("location")) { log(1, "* * * * * < < < could not save location > > > * * * * *\n"); sl_reply_error(); }; break; }; if(method=="INVITE") { fix_nated_contact(); record_route(); if(!lookup("location")) { sl_send_reply("404", "subscriber is offline. Pls try later"); break; } force_rport(); force_rtp_proxy("l"); fix_nated_sdp("1"); } }; if(!t_relay()) { log(1, "relay error\n"); sl_reply_error(); break; }; } --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

2 1

Re: [Serusers] Bandwidth parameter modification (b:) from ser.cfg
by Greger V. Teigre 04 Sep '06

04 Sep '06

Ah! Try: http://www.ietf.org/rfc/rfc3556.txt?number=3556 for description of b-parameter in SDP. g-) M. Victoria Peláez wrote: > Hi! > I will re-try with nathelper, mediaproxy and mangle modules ...... But > till now I haven't found anything about b parameter. About bandwidth > types I also don't now what that mean, I have written it down because > I have read it in some articles ... > > My problem is that I have one application that exchange a b parameter > (b=CT 140) during INVITE and isn't able to work with any other SIP > based application. And the only suitable explanation we have found is > that the b parameter is disturbing our communication. > > So thank you very much! We'll try something else! > > Best regards, > Victoria > > Greger V. Teigre escribió: >> Victoria, >> SIP and SER do not really look at the SDP payload (ideal world). >> However, nathelper, mediaproxy and the mangle (have a look at it) >> modules allow you to do tricks with SDP fooling the user agents >> involved in the dialog in some way (for example by making both talk >> to a rtp proxy server). >> As for bandwidth types (not sure what you mean here?), maybe somebody >> on the list knows of something, but it's slightly off topic on this >> list. >> g-) >> >> M. Victoria Peláez wrote: >>> Hi! >>> >>> I am a beginner with SIP/SER systems. I have been having a look into >>> ser.cfg, in order to modify some SDP parameters, for instance, >>> fix_nated_sdp to change connection IP, and I have found enough >>> information to do it. But, now, I am traying to change b parameter, >>> but cannot find where to do it (Which module manage it?). I am also >>> looking for a document with a description of all bandwidth types >>> (CT, AS). Could you tell me something about this stuff, please? >>> >>> Thank you very much in advance for your help. >>> >>> I am very sorry if this is a very silly question! >>> >>> Best regards, >>> Victoria >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >

1 0

[Serusers] RTP Proxy errors
by sip 04 Sep '06

04 Sep '06

So... I followed the instructions on OnSip on how to set up RTPProxy. However, I made a modification because I'm running it on a different machine... changing the modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpptoxy.sock") to modparam("nathelper", "rtpproxy_sock", "udp:XX.XX.XX.XX:22222") I've run RTPproxy on the remote machine as such: ./rtpproxy -f -l XX.XX.XX.XX -s udp:* 22222 And it's running. I start the ser server, and I see a slew of commands hit the RTPProxy console: received command "26742_0 V" sending reply "26742_0 20040107 " received command "26745_0 V" sending reply "26745_0 20040107 " received command "26791_0 V" sending reply "26791_0 20040107 " received command "26797_0 V" sending reply "26797_0 20040107 " received command "26748_0 V" sending reply "26748_0 20040107 " received command "26750_0 V" sending reply "26750_0 20040107 " received command "26751_0 V" sending reply "26751_0 20040107 " received command "26752_0 V" sending reply "26752_0 20040107 " received command "26784_0 V" sending reply "26784_0 20040107 etc, etc ad infinitum After a moment, the SER console spits back: ug 30 14:07:48 death ser[26742]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:48 death ser[26742]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:48 death ser[26742]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:48 death ser[26745]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:48 death ser[26745]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:48 death ser[26745]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26791]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26791]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26791]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26797]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26797]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26797]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26748]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26748]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26748]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26750]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26751]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26752]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26784]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26789]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26792]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy etc, etc, ad infinitum I've done a search on this on Google and found a couple of people with the same questions and NO people responding with answers. Any ideas what's going on? N.

4 12

Re: [Users] Error when installing openseradmin.Please help..Please
by Ferianto siregar 04 Sep '06

04 Sep '06

Dear Nick, Thank you very much for your kind-hearted to help me. Thank you. Nick, If I can make a wish for some help to you, would you mind help me in these questions: 1. What do you mean by installing mysql? because when I run # rpm -qa|grep mysql, I got the version of mysql are : mysql-3.23.54a-11 mysql-server-3.23.54a-11 php-mysql-4.2.2-17 Do I need install mysql again? If I have to, How? Please 2. Is there any configuration in websever? or I just have to restart the webserver by using "# /etc/rc.d/init.d/httpd restart"? If I have to configure the webserver, Would you mind share the file..Please.. Please Help me Nick..Please...because my skill is too lack. Please... I am so sorry for bothered and waste your time Nick. I am Sorry. Thank you very much Regards, Ferianto nick <nick(a)mobilia.it> wrote: Ferianto siregar wrote: > Dear all, > > Thank you very much to all of you for your kind-hearted to reply my > questions. > Thank you very much. > > All, Now I try to build the openser admin in my openser server. I have > tried to build it for many times, but never successfully. I know that my > skill is too low. So I try to ask it to you. Please help me..Please. > > I have tried to build the openser admin packet by using README .But, I > am confused. Maybe it is because of my lack skill to understand it. > Here is the task that I have done: > 1. I have download Ruby and Rails packet. The versions that I have got > were rubygems-0.9.0.tar.gz and rails-1.1.6.tar.gz > > a. Untar the Rubygems packet > # tar -zxvf rubygems-0.9.0.tar.gz > # cd rubygems > # ruby setup.rb No errors on the install? > b. Untar the Rails packet > # tar -zxvf rails-1.1.6.tar.gz > # cd rails > # ruby script/server If you have gems installed you can just gem install rails --include-dependencies > > c. Untar the openseradmin packet > # tar -xjvf openseradmin-0.3.tar.bz2 > # cd openseradmin-0.3 > That's correct, but without rails installed it probably won't work.. > > d.Logging into the database server > #mysql -u root -p > > Connect openser; > > GRANT ALL ON openser.* TO 'openser'@'localhost' IDENTIFIED > BY 'openserrw'; > > USE openser; > > SOURCE /root/openseradmin-0.3/db/database_tables.sql; > > EXIT > > Here is the confusing problem that I got : > 1. When I try to run "gem install mysql", I got the error message. So, I > can not execute it. What`s wrong? Try gem install mysql --with-mysql-config=/path/to/mysql/lib > 2. When I try to access it by using web browser by typing > "http://localhost:3000", I get nothing. What`s wrong? You need to start the web server, but without mysql, it won't work.. > 3. Would you mind tell me if all the task that I have done correct or > not? I need some suggestion or adviseable. Please.... Install rails, then install mysql, then start the server, it should work fine... > > > I do hope anybody can help me. Please... > > > Regards, > > > > Ferianto > > ------------------------------------------------------------------------ > Yahoo! Messenger with Voice. Make PC-to-Phone Calls > > to the US (and 30+ countries) for 2¢/min or less. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

[Serusers] Help for sn2400 call
by pokerino76 04 Sep '06

04 Sep '06

The mediagateway it is installed in a webfarm of BT (British Telecom) , while the sn2400 it is a sipgateway. I do not succeed to carry out calls of the net outside, while I succeed to receive them. What I must make? thanks. ___________________________________________________________________ Prestiti da 15 istituti. Tan dal 5,90% per richieste entro il 31 Agosto. Servizio gratuito. http://click.libero.it/mutuionline1

1 0

[Serusers] Setting Accounting Flags:-(
by ravi reddy 04 Sep '06

04 Sep '06

Hi users , I am using ser.cfg from onsip.org and because i am not a programmer (means i dont understand the language)I am using the config file as i downloaded , before i gave a lot messages regarding this getting of multiple records in radius database as well as in SER mysql(ac) database also ; i am attaching the ser.cfg below please let me know where to set flag so that i can get single record per call ;-) thank you. debug=4 fork=yes log_stderror=yes listen=My Ip address # INSERT YOUR IP ADDRESS HERE port=5060 children=5 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=766 fifo_db_url="mysql://ser:heslo@localhost/ser" sip_warning=yes server_signature=no loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|permissions|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) modparam("xlog", "buf_size", 8192) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "radius_flag", 11 ) modparam("acc", "radius_missed_flag",12) modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "db_flag",13) modparam("acc", "db_table_acc", "acc") 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("81.21.33.35:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- setflag(11); if (loose_route() ){ ------------------------------------------------------>>>>>>>>>>here i set flag for radius accounting ; # mark routing logic in request #append_hf("P-hint: rr-enforced\r\n"); t_relay(); route(1); break; }; # ----------------------------------------------------------------- # 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"); }; break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP 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()) { xlog("L_WARN", "[%Tf]: possible uri spoofing attempt!; method=%rm; IP=%is; from=%fu; to=%tu\n"); sl_send_reply("403", "Forbidden"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ---------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------- 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")) { setflag(7); force_rport(); fix_nated_contact(); }; if (uri=~"^sip:1[0-9]{8}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:00[0-9]*@") { route(4); route(5); break; }; if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { if (!avp_check("s:fwdbusy", "eq/$ruri/i")) { setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("pstn gateway ip"); avp_write("i:30", "inv_timeout"); t_on_failure("1"); route(4); route(1); } route[6] { # ---------------------------------------------------------- # Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ---------------------------------------------------------- if (uri=~"^sip:1[0-9]{8}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; if (uri=~"^sip:00[0-9]*@") { route(4); route(5); break; } if (uri=~"^sip:+[0-9]{3}[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { #local pstn route(4); route(1); break; }; sl_send_reply("404", "User Not Found"); }; route(4); route(1); } 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(); }; } .........................)

1 0

[Users] Error when installing openseradmin.Please help..Please
by Ferianto siregar 04 Sep '06

04 Sep '06

Dear all, Thank you very much to all of you for your kind-hearted to reply my questions. Thank you very much. All, Now I try to build the openser admin in my openser server. I have tried to build it for many times, but never successfully. I know that my skill is too low. So I try to ask it to you. Please help me..Please. I have tried to build the openser admin packet by using README .But, I am confused. Maybe it is because of my lack skill to understand it. Here is the task that I have done: 1. I have download Ruby and Rails packet. The versions that I have got were rubygems-0.9.0.tar.gz and rails-1.1.6.tar.gz a. Untar the Rubygems packet # tar -zxvf rubygems-0.9.0.tar.gz # cd rubygems # ruby setup.rb b. Untar the Rails packet # tar -zxvf rails-1.1.6.tar.gz # cd rails # ruby script/server c. Untar the openseradmin packet # tar -xjvf openseradmin-0.3.tar.bz2 # cd openseradmin-0.3 d.Logging into the database server #mysql -u root -p > Connect openser; > GRANT ALL ON openser.* TO 'openser'@'localhost' IDENTIFIED BY 'openserrw'; > USE openser; > SOURCE /root/openseradmin-0.3/db/database_tables.sql; > EXIT Here is the confusing problem that I got : 1. When I try to run "gem install mysql", I got the error message. So, I can not execute it. What`s wrong? 2. When I try to access it by using web browser by typing "http://localhost:3000", I get nothing. What`s wrong? 3. Would you mind tell me if all the task that I have done correct or not? I need some suggestion or adviseable. Please.... I do hope anybody can help me. Please... Regards, Ferianto --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

2 1

[Serusers] Proxy and Presence Agent support
by Henry Duan 04 Sep '06

04 Sep '06

hi, I'm new to SER. I need to use SER as a proxy and a PA as well. I'm wondering if I could just download the package unstable version of SER with presence<http://ftp.iptel.org/pub/ser/presence/> from iptel.org website. BTW, I'm using Linux Fedora 5, not debian. I'm wondering if there is any problem for this. Thanks a lot! henry

3 2

[Users] Problem with ENUM
by Andrew And 03 Sep '06

03 Sep '06

Below I m showing my DNS configuration (Bind version 9): $ORIGIN 1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server2.com!" . When I use DIG "1.8.8.5.2.1.2.3.3.e164.arpa" or " 1.8.8.5.3.1.2.3.3.e164.arpa", I don´t have problem. But When I try "1.8.8.3.2.1.2.3.3.e164.arpa" DIG don´t obtain resolution. I see that I need to add in DNS the tree between 1.2.3.3 and 8.5.2.1.2.3.3: $ORIGIN 2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@ server1.com!" . $ORIGIN 8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . I have a lot of servers and this is not scalable, then I can´t use the tree between 1.2.3.3 and 8.5.2.1.2.3.3. Is there another configuration that I don´t need to use the tree between 1.2.3.3 and 8.5.2.1.2.3.3?

1 0

[Serusers] SER+Asterisk
by Siqhamo Sifo 03 Sep '06

03 Sep '06

Where can I find docs on asterisk and ser integration.

1 0

[Users] Need help for openser admin installation
by Ferianto siregar 03 Sep '06

03 Sep '06

Dear all, Thanks to every body in this forum. Thank you very much I tried to build openser admin in my openser server. But, I am confused when i try to read the installation note. I do hope anybody can help me to understand it. These are my questions about it: 1. As I read, the openser admin need ruby packet. For installation of ruby, after we untar the packet, does we just type "gem mysql install" command?Because as i read from the installation openser admin, there are only this command. 2. When I try to build the openser admin, i got the error message. Here is the message : ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, domain, attribute, type AS att' at line 1 As I read from the message in this mailinglist before, this is because of the mysql version. I use mysql 3. Do i need use mysql 5? Where I can get the suitable varsion? That all my question. I do hope any body can help me. Thank you very much. Regards, Ferianto --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

1 0

[Serusers] Help for sn2400 call
by pokerino76 02 Sep '06

02 Sep '06

I have shaped the sn2400 for the ser, succeed to make telephone calls of the net inside ,the calls phone inside working, but I do not succeed to calls phone from the outside, which it must be the configuration of ser.cfg or the other? . Thaks for help.

2 1

[Serusers] SER+Asterisk
by Siqhamo Sifo 02 Sep '06

02 Sep '06

I have ser sitting on my iptables nat box and my asterisk box on the lan . Ser does forwarding so that any requests (register,invite,ack,...) to the nat box at 5060 r sent to my asterisk box on the lan .I can register from outside to my asterisk box but there is only one way audio , reason being that when the asterisk box send a sip packet whith session description the sdp part of the sip packet is not natted . I have tried : fix_nated_contact(); fix_nated_sdp("3");

3 2

[Serusers] Presence modules and LCS
by harrygaillac-sip＠yahoo.fr 02 Sep '06

02 Sep '06

Hello, Does ser presence modules are able to work with LCS micosoft ? http://www.microsoft.com/office/livecomm/prodinfo/compare.mspx#ECD anybody is experienced with LCS to answer ? Regards Harry p4.vert.ukl.yahoo.com uncompressed/chunked Fri Sep 1 12:13:36 GMT 2006 ___________________________________________________________________________ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com

1 0

[Users] RTPProxy non working?
by Federico Giannici 02 Sep '06

02 Sep '06

I'm trying to install rtpproxy. It usually works, but there are some cases that don't work. For example, when a Cisco PSTN gateway is called, RTP traffic is proxied in one direction only. I think the problem is in the handling of the reply received from the gateway: it have two "c" lines but only one is modified. Here it is the body of the original reply: v=0. o=CiscoSystemsSIP-GW-UserAgent 2199 2489 IN IP4 83.211.2.216. s=SIP Call. c=IN IP4 83.211.2.217. t=0 0. m=audio 18850 RTP/AVP 8 13 101. c=IN IP4 83.211.2.217. a=rtpmap:8 PCMA/8000. a=rtpmap:13 CN/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=direction:passive. And here it is the body modified by rtpproxy: v=0. o=CiscoSystemsSIP-GW-UserAgent 2199 2489 IN IP4 83.211.2.216. s=SIP Call. c=IN IP4 83.211.2.217. t=0 0. m=audio 35086 RTP/AVP 8 13 101. c=IN IP4 195.120.250.49. a=rtpmap:8 PCMA/8000. a=rtpmap:13 CN/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=direction:passive. a=nortpproxy:yes. Only the secondo "c" line contain the IP of our rtpproxy. I don't know RTP enough to understand if the problem is in rtpproxy or in the gateway. Thanks for any info. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 4

[Users] openser admin question..
by nick 01 Sep '06

01 Sep '06

Is an error like this normal when doing the db install? ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, domain, attribute, type AS att' at line 1 By the way, while ruby on rails might be a great development platform, it's a right pain in the behind to install.

2 5

[Users] codecs problem...
by raviprakash sunkara 01 Sep '06

01 Sep '06

Hi Users...... A doubt For me in Codec.... I'm Using openSER and RTPProxy openser didn't take the Media tranmmision , it transmmit to RTPproxy or mediaproxy. Media(voice) taking place in RTP proxy... UA1 using the G729 codec -------> openser server -------------------> UA2 using G711u for above case it will not generate the porper handshanking............ But RTP Proxy may has transmiits the media , because it has codecs in ( media proxy or rtpproxy) For a realtime producttion, UAs may having the differnet codecs. plz resolve my doubt..... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] Integration of Openser with CDRTool: Installation errors
by Alberto Pacifico 01 Sep '06

01 Sep '06

Hi all, I'm trying to install CDRTool (http://www.ag-projects.com/CDRTool.html) and some errors happened. When I try to access the CDRTool webpage, I'm able log in the page, but when I try to enter in the "call search page" some warnings appears: "Warning: main(SOAP/Client.php): failed to open stream: Arquivo ou diretÃ³rio nÃ£o encontrado in /var/www/CDRTool/rating_lib.phtml on line 4" "Fatal error: main(): Failed opening required 'SOAP/Client.php' (include_path='.:/opt/lampp/lib/php') in /var/www/CDRTool/rating_lib.phtml on line 4" In the file "rating_lib.phtml", the line 4 is: "require_once("SOAP/Client.php");" The file Client.php does not exist... But a file named "client.php" exists..and into it, theres an instruction calling "Client.php": "include("SOAP/Client.php");" An additional information is that I'm using the XAMPP Server Apache. Please, Can you help me? Thanks in advance! Alberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] openser+asterisk+voicemail password
by ram 01 Sep '06

01 Sep '06

Hi iam integrating with asterisk voice mail with Openser when we are creating openser views with asterisk the secret going to be Binary(0);, how can i change to numeric ? why i need is, take example If the user Phone logged in and the user have person voicemessage box he only can access his voice messages, when he dial secret If not any one can come and access voice message by dialing access number. with the config seen in the examples, person can dial *98 and hear all the messages and delete too. so for the security reason, how can i change the secret, iam trying to alter the table, but its say;s in view mode. any suggestions. Ra,

1 0

[Users] Question about voip-info.org's OpenSER/RTPProxy configuration example
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was exploring the example configuration file in the following URL: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy The following block bewilders me: % failure_route[1] { % if (isflagset(6) || isflagset(5)) { % unforce_rtp_proxy(); % } % } How a flag happens to be turned on there ? 6 is the registrar's nat_flag but no lookup() is performed. Furthermore, the tm module doesn't even depend on the registrar module. Flag 5 is set in route[2] which is obviously not called from failure_route[1]. Is there some black magic around here ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 3

[Users] RTPProxy failover... load-balancing ?
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was wondering whether I could use multiple rtpproxy daemons along one OpenSER instance, in order to both provide a failover mechanism as well as load-balacing. Google immediately pointed me out the following thread: http://www.openser.org/pipermail/users/2005-October/001071.html It happens that openser/modules/nathelper/nathelper.c contains the following comment: % 2005-02-24 Added support for using more than one rtp proxy, in which % case traffic will be distributed evenly among them. In addition, % each such proxy can be assigned a weight, which will specify % which share of the traffic should be placed to this particular % proxy. % % Introduce failover mechanism, so that if SER detects that one % of many proxies is no longer available it temporarily decreases % its weight to 0, so that no traffic will be assigned to it. % Such "disabled" proxies are periodically checked to see if they % are back to normal in which case respective weight is restored % resulting in traffic being sent to that proxy again. % % Those features can be enabled by specifying more than one "URI" % in the rtpproxy_sock parameter, optionally followed by the weight, % which if absent is assumed to be 1, for example: % % rtpproxy_sock="unix:/foo/bar=4 udp:1.2.3.4:3456=3 udp:5.6.7.8:5432=1" At first glance, I would say the aforementioned comment states that nathelper achieves both failover and weighted load-balancing. Can anyone confirm this ? Ultimately, I would like to do use the following arch : ________+----------+ /________| RTPProxy | +---------+ // +----------+ | OpenSER |_____// +---------+ /\_________+----------+ / \ ______| RTPProxy | +---------+ / \/ +----------+ | OpenSER |__/_____/\ +---------+ \ \_____+----------+ \____________| RTPProxy | +----------+ If I understand correctly how it works, this should be possible to use N OpenSER instances along with M RTProxy instances. Am I right ? Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

1 0

[Users] Simple configuration
by Vladimir Orehov 01 Sep '06

01 Sep '06

Hello, I'm new to this server, and I ask for help! I don't undertand for now how to configure openSER. Can someone send me simple configuration file "openser.cfg" wich included: -- accounting loggoing to a specific file ( like "/home/agb/acc.log") -- authentication from simple text file (if possible), or using SQL -- routing numbers started with 00 to specific gateway with ip (123.123.123.123) Another question is: how to enable debugs of this server? Please reply ASAP and best regards, Vladimir Orehov SIA Velkoms Support engineer

2 1

[Users] Question about voip-info.org's OpenSER/RTPProxy configuration example
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was exploring the example configuration file in the following URL: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy The following block bewilders me: % failure_route[1] { % if (isflagset(6) || isflagset(5)) { % unforce_rtp_proxy(); % } % } How a flag happens to be turned on there ? 6 is the registrar's nat_flag but no lookup() is performed. Furthermore, the tm module doesn't even depend on the registrar module. Flag 5 is set in route[2] which is obviously not called from failure_route[1]. Is there some black magic around here ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] Error in running serweb. Please help me..Please
by Ferianto siregar 01 Sep '06

01 Sep '06

Dear all, First of all, I would like to say thanks to all of you for your help and suggestion to solve my problem untill now. Thank you very much. Now, I try to install serweb_2004-07-27 in my openser server. I have read the installation note that I got from the serweb packet carefully. So, I can build the serweb in my openser server successfully. It means that it can display a login screen in the browser. But, I have a problem when I try to login by using username "admin" with password "openserrw". (This is based on the installation note that I have read before) I got error message. This is the error message : Database error: Invalid SQL: select username, perms from subscriber where phplib_id = '721153ca8942ea1a08900f5b61b69a6c' MySQL Error: 1054 (Unknown column 'perms' in 'field list') Session halted. As I try to understand from the error message, it says that there is no "perms" column. So I try to check it in the subscriber table in mysql.But, I did not find "perms" column in the subscriber table. I just see these columns in the subscriber table: -phplib_id -last_name - sendnotification - rpid -username - phone - greeting -domn -domain - email_address - ha1 -uuid -password - datetime_created - hdb1 -first_name - datetime_modified - allow_find -confirmation - flag - timezone What should i do to solve this problem? What should I do if there is no "perms" column in the subscriber table? I do hope anybody can give me a suggestion because this is my first time to build the serweb and mysql. Please help me....Please... Note: I user openser1.1.0-tls and serweb_2004-07-27 As I see from the error message that the phplib_id is exactly matched with phplib_id where username "admin" with password "openserrw" (from the subscriber table). Thank you very much Regards, Ferianto --------------------------------- Get your email and more, right on the new Yahoo.com

3 2

[Users] Migrate from SER to open SER
by ram 01 Sep '06

01 Sep '06

Hi all I have VoIP CDrouter running But i have seen lot of limitation How can i migrate from SER to Open Ser with out lossing any data of users information and CDR Help will be appriciated Ram

2 1

[Users] Incorrect Route header
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I am running a NATing proxy. Everything works fine, except that some of my neighbour proxies send me BYE requests with an incorrect Route header containing their own IP address. The result is naturally that my proxy sends back the request and the call is never ended. I know this is a wrong behaviour and I have contacted the remote administrator to have him fix his proxy. But in the meantime I would like to find a workaround for my users. I tried to do the following, before calling loose_routing(). I am not sure if it's relevant or not nevertheless: % if (method == "BYE") { % # XXX Remove buggy Route header from BYE packets. % xlog("L_ALERT", "Removing buggy Route header from BYE request"); % remove_hf("Route"); % lookup("location"); % } However, I raised the verbosity level and found that the Route header is read as soon as the packet is received, therefore the above block is not really useful. Thank you for your ideas. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 2

[Users] VIA Parameter?
by Ricardo Martinez 01 Sep '06

01 Sep '06

Hello list. Sorry to ask this question here, but i can't find the answer in the RFC3261 Is this "VIA" permited according to the RFC? Via: SIP/2.0/UDP 69.20.6.205:15060, SIP/2.0/UDP 69.1.229.69:4081;branch=z9hG4bK-58e9ccfcef0f4c99915b8007cb9713e1-1156793814 It seems to be two VIA's in ONE line... Thanks Regards, Ricardo.-

3 2

[Users] SIP Redirection in Openser
by Padmaja RV 01 Sep '06

01 Sep '06

Hi all, I have tried implementing SIP redirection as mentioned in the example script below: route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # rewrite current URI, which is always part of destination ser rewriteuri("sip:parallel@iptel.org:9"); # append one more URI to the destination ser append_branch("sip:redirect@iptel.org:9"); # redirect now sl_send_reply("300", "Redirect"); } But that does redirection only for that account as the account is included in the code. I need the code to be able to redirect any call. Can you tell me how I can generalize the code so that redirection works for any account I choose to redirect? Thanks in advance, Padmaja

2 1

[Users] Authentication in Trunk-Mode
by Martin Klisch 01 Sep '06

01 Sep '06

Hey, are there working ways for authentication with sip trunks? i've read cisco gateways (PGW) and cisco call manager can do this. anyone knows if openser+asterisk support it too?

3 4

[Users] About 'location' table
by Michel Zenone 01 Sep '06

01 Sep '06

Hi! I have registered, thanks to openserctl, an user :'sip:testeur@192.168.0.102'. But when I tyr to reach him, the transaction isn't considered as local processed one. In location table there is: +username + domain + contact + |testeur | | sip:testeur@192.168.0.120:5080 | +---------+--------+--------------------------------+ -192.168.0.120 is the IP address of the user's phone. -192.168.0.102 is the openser IP address. The openser.cfg is as the initial openser.cfg. Thanks, Michel

2 1

[Users] OpenSER AstLinux build posted for testing
by Kristian Kielhofner 01 Sep '06

01 Sep '06

Hello OpenSer Users, It seems that many of you are also Asterisk users or at least subscribe to asterisk-users. Some of you might recognize AstLinux... Just in case you don't: http://www.astlinux.org ---and--- http://www.sf.net/projects/astlinux [in transition] Anyways, I have long hoped to build a version of AstLinux that includes OpenSer. Basically, what I have done is taken astlinux-trunk, add openser, add rtpproxy, de-select the Asterisk packages, and build! What I got was 11mb (gzipped) disk images ready to be written to CF cards/hard disks. As usual, I have created two images: *net4801.img.gz is for the Soekris net4801 SBC *i586.img.gz is for just about any i586 and higher machine There is also an ISO image ready to boot on or install to most i586 or higher machines. You guessed it, it is the one with .iso in the filename. It includes both the net4801 and i586 images. These are 48mb gzipped disk images, with GRUB bootloader and ext2 root filesystem. If you want to write them to disk, try this: gunzip [image.gz] dd if=[output file.img] of=/dev/sdb # <--- VERIFY THIS DEVICE Where /dev/sdb is where your USB CF flash adapter is. If you want access to the ext2 filesystem, try this: gunzip [image.gz] mkdir /mnt/image mount -t ext2 -o loop,offset=32256 [image.gz] /mnt/image If you are curious about how these images were created, take a look at SVN on the astlinux project page. The OpenSer version used for these builds was openser-1.1.0-notls. I built every module that did not depend on an external library that I don't have yet (sorry, no mysql, jabber, or radius yet). I need help!!! For the time being, I don't have mysql. Also, because AstLinux is targeted for compact flash, mysql isn't a good idea anyways. I was not able to get a working configuration in the time alloted using DBTEXT. Does anyone have a sample config that I can put in this build? As of now, openser, rtpproxy, openser modules, and all of the astlinux-style init scripts are done. All I need is a good sample config file to make it "plug and play". I also need testing! My current to-do list: - add mysql (and the openser modules, of course) - add radius support (freeradius?) - jabber/xmpp (depending on size, complexity, etc) - fix astshape (iproute2-based traffic shaper) RTP TOS values to be the same that rtpproxy uses, and standardize this across all astlinux builds - COME UP WITH A NAME. I have no idea what to call this thing. Any suggestions? SERlinux (or something like it) is too obvious and boring! You can grab the goodies here: http://www.krisk.org/openser/ Regardless, I hope that someone finds this interesting or useful. Let me know what you think! -- Kristian Kielhofner

3 2

[Users] icmp port unreachable
by Helge Waastad 01 Sep '06

01 Sep '06

Hi, (just a little bit on the side-bar) I've had some discussion with the developer of twinkle (which is, BTW, a very nice linux softphone) regarding handeling icmp port unreachable and switching to a lower priority SIP router (by DNS SRV). It seems like the dist I'm using (Dapper, kernel 2.6.15-26) will not easily allow the applications to receive the icmp on an unconnected udp socket, hence failover will not work. However on an 2.6.11.4-21.11 kernel it works. Does anyone know if there are any kernel opts which should be added for having this kind of support? Just my thoughts this morning......

2 1

[Users] Openser + Asterisk for P2P calls
by Lucio Maciel 01 Sep '06

01 Sep '06

Hi, Hi, im using an Openser integrated with Asterisk to route PSTN calls, this is working great, but i also want to route P2P calls to Asterisk like this UA1 -> Openser -> Asterisk (calls DIAL(SIP/xxxxx@openser) -> Openser -> UA2, to use its mediaproxy, and account all calls. When UA1 calls UA2, everything goes fine, UA2 receive the INVITE and sends the OK to UA1, the only thing is that UA2 waits for an ACK from the OK that never comes. Have someone tried to do this? How can i configure openser and asterisk for this? Thanks Lucio Maciel

2 1

[Users] IP authentication
by Brian Fertig 01 Sep '06

01 Sep '06

greetings, I have digest and basic auth setup. I am trying to figure out how to setup IP auth. I have read a lot of documentation but I can't seem to find anything to help me understand how to accomplish this. Can someone point me in a direction to some information on how to set this up? Brian

6 5

[Users] 1.1.x/permissions.html is not up to date
by Juha Heinanen 01 Sep '06

01 Sep '06

for some reason web page http://www.openser.org/docs/modules/1.1.x/permissions.html is not up to date. for example, 1.3.12. tag_col (string) 1.3.13. peer_tag_avp (string) are missing. they do exist in rel_1_1_0 source. -- juha

2 1

[Serusers] extra things confusing auth_radius?
by Natambu Obleton 01 Sep '06

01 Sep '06

Ok. I have two clients setup. One is X-Lite and the other is a polycom 601. I am doing radius authentication, but for some reason it doesn't work with the polycom 601 phone. In the debug the only thing that is different about them is something in the uri. Aug 30 09:16:51 laplata ser[25968]: DEBUG: get_hdr_field: <To> [72]; uri=[sip:311@voip.fasttrackcomm.net] Works, but.. Aug 30 09:17:53 laplata ser[25970]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:311@voip.fasttrackcomm.net;user=phone] <mailto:%5bsip:311@voip.fasttrackcomm.net;user=phone%5d> Doesn't. Like it doesn't even send an authentication request to my radius server. I think it is the ';user=phone' part that is confusing it? So what I want to do is remove that part from the URI as a test. I just added a subst to remove it, but still isn't working with radius. It acts like the realm is no longer local. Here is my config file: http://www.natambu.com/serconf.html Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

2 3

[Serusers] SIP Contact header field issue incase of NAT
by Abdul Qadir 01 Sep '06

01 Sep '06

Hi All, I have problem connecting Nokia Phone with SER. The problem is When SER receives the registration request from the Nokia phone. The registration request contain the phone's private IP (192.xxx.xx.xx), since we use wifi for connection. Our proxy detects that since this is a private IP handles the NAT translation and determines the phone public IP address (e.g., 202.xxx.xx.xx, port xx). Our SIP proxy informs the phone that registration is successful but the contact information is changed from userid(a)192.xxx.xxx.xx to userid(a)202.xxx.xxx.xx, port xx. On recieving this changed contact info the Phone rejects 200 Ok. We tried to contact nokia people and they said that its part of RFC3261 to reject such contacts. After this I have also gone through RFC 3261 section "10.2.4" and it seems that phone is doing rite. Can any body tell me why SER is changing the contact address in case of NAT. It would be higly appreciated if you can give me some reference from some RFC or other document which I can show to nokia people. Best Regards, Abdul Qadir --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

3 2

[Serusers] Bandwidth parameter modification (b:) from ser.cfg
by "M. Victoria Peláez" 01 Sep '06

01 Sep '06

Hi! I am a beginner with SIP/SER systems. I have been having a look into ser.cfg, in order to modify some SDP parameters, for instance, fix_nated_sdp to change connection IP, and I have found enough information to do it. But, now, I am traying to change b parameter, but cannot find where to do it (Which module manage it?). I am also looking for a document with a description of all bandwidth types (CT, AS). Could you tell me something about this stuff, please? Thank you very much in advance for your help. I am very sorry if this is a very silly question! Best regards, Victoria

2 1

[Serusers] mediaproxy
by Javier Ramirez 01 Sep '06

01 Sep '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

2 1

[Users] openser routing lookup question
by Dave Logan 01 Sep '06

01 Sep '06

Hey all, I have what may or may not be a simple situation. I've read through dozens of forum entries, dozens more list mails, and a handful of "tutorials" for openser, and I'm still missing something I think. I'm not afraid of the documentation, but neither the "core cookbook" nor the modules documentation had enough detail that I could catch on to what's going on. Let me lay out my scenario, and maybe somebody can point to a particular document or solution I can follow up on. I have multiple asterisk boxes: four as media gateways with TDM trunks, one feature server (VM), and three registrars. So needless to say, I have a lot of dialplan in there just for calls that stay in my network. What I'd like is to introduce a single openser box in there, and for any call that any of those asterisk machines gets, it sends it to openser, which then just calls a simple script (which I've already more or less written) to know what to do. So for example, I own the 555-1000(-1099) DID block DID 555-1000 is registered to machine1 DID 555-1001 is registered to machine2 In openser, I'm envisioning a call like such (php style notation): my $number = 5551000 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551000@machine1 or my $number = 5551001 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551001@machine2 Then I can just route it in the normal openser way. So there, my lookupdestination script knows all the numbers in my central database and where they're located, and on an unknown number assumes it's outside our network so just sends it to the gateway as is (5552000@gw1) or so. Thanks for any pointers at all, dave -- Dave Logan http://www.digitalcoven.com/ "Do let's pretend that I'm a hungry hyena, and you're a bone!" - Alice

2 1

[Users] Help in dailplan in asterisk
by raviprakash sunkara 01 Sep '06

01 Sep '06

Hi Users, I'm new to Asterisk, and I'm working with openSER , For Call Routing I'm using the OpenSER and for PBX, Voicemail and Conferencing I"m using the Asterisk. Now i'm planing for Voicemail, openser listen on 192.168.2.75:5060 Asterisk listen on 192.168.2.76:5060. in Extension.conf [from-sip[ exten => 9001,1,Ringing exten => 9001,2,Voicemail(9001) Just I wnat to leave the voicemail in voice mail box, Can anybody help me......... below message What it is ?................ Executing Ringing("SIP/9001-0984c1e8", "") in new stack -- Executing VoiceMail("SIP/9001-0984c1e8", "9001") in new stack Aug 31 19:31:10 ERROR[7869]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. Aug 31 19:31:10 WARNING[7869]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '9001' Aug 31 19:31:10 WARNING[7869]: pbx.c:1700 pbx_extension_helper: No application 'Hungup' for extension (from-sip, 9001, 3) == Spawn extension (from-sip, 9001, 3) exited non-zero on 'SIP/9001-0984c1e8' -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" back from 192.168.2.75 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

1 0

[Users] avp_pushto() changed???
by Greg Fausak 31 Aug '06

31 Aug '06

This doesn't work for me any more: avp_pushto("$P-Asserted-Identity", "$avp(s:tmp_uri)"); I get this: Aug 31 13:42:00 www openser[21792]: ERROR:avpops:fixup_pushto_avp: unsupported destination "$P-Asserted-Identity"; expected $ru,$du,$br Has anything changed? -g -- Greg Fausak greg(a)thursday.com

3 2

[Users] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Users] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Serusers] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Users] OpenSER Summit
by Bogdan-Andrei Iancu 31 Aug '06

31 Aug '06

Hello everybody, It's more than one year since the OpenSER project started. A large community of users and developers has been formed around OpenSER. It's a very active community and one of the most important goals of the OpenSER project is to keep the users community as synchronized as possible with the developers and management groups. During past time we met a lot of people that were happy to see us or we were happy to see them. So, we came to the idea of organizing a OpenSER Summit where OpenSER fans (all types of fans) should meet to share their experience, to discuss about future ways of developing the project or just to personally meet people from the mailing lists. The idea is to have people talking about how they are using OpenSER in different contexts and which were/might be their needs, development people talking about possible solutions and about their ideas of enhancing OpenSER, etc... We were thinking of taking advantage of the upcoming VoN in Berlin in the beginning of November and to organize the Summit within the same time frame - it will be more convenient for people to travel for two events at the same time. We would like to have some feedback from the community - how many people are interested in such a meeting, what will be their primary interests. Any other OpenSER-related ideas are most welcome! Best regards, Bogdan

2 1

[Serusers] pre_auth(): Credentials with given realm not found
by ram 31 Aug '06

31 Aug '06

Hi all iam testing Voicemessages with Asterisk in the log i get frequently pre_auth(): Invalid nonce value received, very suspicious ! when i debug, i get the following message i have posted my config couple of day back 5(2213) get_hdr_field: cseq <CSeq>: <6> <REGISTER> 5(2213) DEBUG: is_maxfwd_present: value = 70 5(2213) parse_headers: flags=256 5(2213) DEBUG: get_hdr_body : content_length=0 5(2213) found end of header 5(2213) find_first_route(): No Route headers found 5(2213) loose_route(): There is no Route HF 5(2213) check_self - checking if host==us: 14==13 && [mydomain.com] == [myipaddress] 5(2213) check_self - checking if port 5060 matches port 5060 5(2213) parse_headers: flags=-1 5(2213) parse_headers: flags=-1 5(2213) check_via_address(some ip address, some ip address, 0) 5(2213) check_self - checking if host==us: 14==13 && [mydomain.com] == [myipaddress] 5(2213) check_self - checking if port 5060 matches port 5060 5(2213) parse_headers: flags=4096 5(2213) pre_auth(): Credentials with given realm not found 5(2213) build_auth_hf(): 'WWW-Authenticate: Digest realm="mydomain.com", nonce="44f6ee623e15edb81a22dcec25ebe8c58b6e056d" '

3 3

RE: [Users] openser admin question..
by peter.3.edwards＠bt.com 31 Aug '06

31 Aug '06

> mysql-4.1.20-1 > > There's no 5 built for centos yet, but this should be fairly > up to date... Well, there's always the option to compile from source ... which isn't quite as painful as it sounds. > The program works.. but I'm not sure what that error might > cause in way > of bugs, or whatever. me either ... sorry! > I had no end of problems installing mysql-ruby in the end, I ended up > running the extern.rb with some command line options > --with-mysql-config > and that got the makefile built. Yeah, same here ... it was quite tedious trying to get it to compile correctly. Ended up having to reinstall MySQL to get the libmysqlclient.so library which isn't bundled with the Solaris package. *sigh* > BTW, what version of mysql-ruby should I be running, I'm > using 2.5.1 at > the moment, 2.6 and 2.7 were also available I went for 2.7 - again, the latest version available - when I just built my Solaris 9 OpenSER Administrator installation. HTH, Peter. -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

1 0

[Serusers] LCR module documentation
by Graham Lewis 31 Aug '06

31 Aug '06

2 1

[Serusers] caller=callee, Not giving me Busy
by ram 31 Aug '06

31 Aug '06

Hi iam using old SER 0.814, since its deployed already in my Office we are in process of migrationg to openser soon mean time i need to fix some problems with existing server only that is the reason iam trying to modify the code and achive the following things 1. Voice mail 2. Caller =callee 3. if the user not in voicemail group ( get busy), if the user off line, Not found I have modified some code after reading all news group here iam attaching the config file which iam trying iam also getting the some errors in Log Aug 30 17:58:20 router ser[609]: ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) Aug 30 17:58:27 router ser[604]: ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) Aug 30 18:00:05 router ser[1564]: ERROR: tcp_init: bind(6, 0x80bd7c0, 16) on 127.0.0.1: Address already in use Aug 30 18:00:06 router ser[1571]: ERROR: tcp_init: bind(6, 0x80bd7c0, 16) on 127.0.0.1: Address already in use iam running a cron job to check the services #!/bin/sh mysqlcheck --password=cdrouter --auto-repair --all-databases sleep 4 ser sleep 4 chmod 777 /tmp/ser_fifo help will be appriciated Ram

2 1

[Serusers] How to Integrate SER Server with Jabber IM.
by srinivas sriram 31 Aug '06

31 Aug '06

Hello, I am look to send Missed Call Alert Instant Message(IM) to Jabber Client from SER Server. 1) I have Installed and Configured SER on IP(xyz.xyz.xyz.xx), and Added two(2) user in mysql i.e 1000, 2000 and also Configured SER Client(Kphone) with SER Server. 2) I have Installed and Configured Jabber Server i.e WildFire on IP(xyz.xyz.xyz.yy), and Added two(2) user, i.e 1000, 2000 ans also configured Jabber Client(Exodus) with Jabber Server. 3)My aim is to send Missed call alert, when a call is made from SER Client(K phone) User name:1000 to SER Client(Kphone) User name:2000, and User 2000 is not replyed. SER Server should send Instant Message(IM) to Jabber Client( User name: 2000) saying one missed call from user 1000. 4)Ser Server is running on xyz.xyz.xyz.xx Jabber Server is running on xyz.xyz.xyz.yy Came you please send me ser.cfg with give me this feature ( SER integration with Jabber IM ). And step to Integrate with Jabber IM. My configuration is attached as ser.cfg. Thanks & Regards ************************************ Sriram Srinivas ************************************

1 0

[Serusers] How to Integrate SER Server with Jabber IM.
by srinivas sriram 31 Aug '06

31 Aug '06

Hello, I am look to send Missed Call Alert Instant Message(IM) to Jabber Client from SER Server. 1) I have Installed and Configured SER on IP(xyz.xyz.xyz.xx), and Added two(2) user in mysql i.e 1000, 2000 and also Configured SER Client(Kphone) with SER Server. 2) I have Installed and Configured Jabber Server i.e WildFire on IP(xyz.xyz.xyz.yy), and Added two(2) user, i.e 1000, 2000 ans also configured Jabber Client(Exodus) with Jabber Server. 3)My aim is to send Missed call alert, when a call is made from SER Client(K phone) User name:1000 to SER Client(Kphone) User name:2000, and User 2000 is not replyed. SER Server should send Instant Message(IM) to Jabber Client( User name: 2000) saying one missed call from user 1000. 4)Ser Server is running on xyz.xyz.xyz.xx Jabber Server is running on xyz.xyz.xyz.yy Came you please send me ser.cfg with give me this feature ( SER integration with Jabber IM ). And step to Integrate with Jabber IM. My configuration is attached as ser.cfg. Thanks & Regards ************************************ Sriram Srinivas ************************************

1 0

RE: [Users] openser admin question..
by peter.3.edwards＠bt.com 31 Aug '06

31 Aug '06

> Is an error like this normal when doing the db install? > > ERROR 1064 (42000): You have an error in your SQL syntax; check the > manual that corresponds to your MySQL server version for the right > syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, > domain, attribute, type AS att' at line 1 Hi, Nick. Not that it helps you much - but, no, I don't think that's normal - I didn't get this when installing on MySQL 5.0.24 (latest stable version), compiled from source. You might get compatibility issues if your version of MySQL is out of date ... what are you running? I'm sure Mike Williams will be along to help you debug it ... > By the way, while ruby on rails might be a great development > platform, > it's a right pain in the behind to install. Completely agree - especially when you're in a network which requires the use of a proxy to reach the web ... Bleugh. Still, once it's up and running it should be easy to maintain, or so I'm led to believe! Peter. -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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[Serusers] Granstream AtA g729 codec --- help please
by ravi reddy 31 Aug '06

31 Aug '06

---------- Forwarded message ---------- From: ravi reddy <mravikreddy(a)gmail.com> Date: Aug 29, 2006 9:56 AM Subject: Re: [Serusers] Granstream AtA g729 codec --- help please To: "Greger V. Teigre" <greger(a)teigre.com> Thanks Greger for asking me, I want a small information regarding billing of sip calls (even though it is not related SER) please make a note to me :-) here in radacct i am getting more than one record for every call ; some folks told that i need some perl script to format all the database entries and write a fresh copy so that i can get one record for one call which is easy for billing. So, what i need a suggestion from you is 1) Do i need to learn perl language and write the script ?. 2) or is there any other way to control the overflow of start stop messages in to SER ...?. If you know please tell me . Thank You. Regards, Ravi. On 8/28/06, Greger V. Teigre <greger(a)teigre.com> wrote: > > Good to hear! > g-) > PS! No need to "sir"-me. Keep it informal, I'm Greger ;-) > > > ravi reddy wrote: > > Dear Sir, > > Thanks for your reply. i found the problem :-) > > the problem is itself in my pstn gateway that they have to configure to > allow my SER sending g729 codecs , > > i came to know from the folling grep message. > > v=0. > o=32331001 8000 8001 IN IP4 192.168.0.74. > s=SIP Call. > c=IN IP4 81.21.33.35. > t=0 0. > m=audio 60040 RTP/AVP 18 4 99 2. > a=sendrecv. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:99 iLBC/8000. > a=fmtp:99 mode=20. > a=rtpmap:2 G726-32/8000. > a=ptime:20. > > > # > U 81.21.33.35:5060 -> 81.21.34.34:5068 > SIP/2.0 501 Not Implemented. > Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received= 81.21.34.34 > ;branch=z9hG4bK8bcaffffe66dffff. > From: "ravi" <sip:32331001@81.21.33.35>;tag=502effff5ddeffff. > To: <sip:99106883@81.21.33.35>;tag=E067A27C-3ED. > Date: Mon, 28 Aug 2006 08:24:22 GMT. > Call-ID: 61d40000ddc8ffff(a)192.168.0.74. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 47822 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > > > because this 501 is internal server or gateway error so iam working on > that now. > > Thankyou. > > Regards, > Ravi. > > > > > On 8/28/06, Greger V. Teigre <greger(a)teigre.com > wrote: > > > > My gut feeling would tell me that the codec has nothing to do with it > > unless you find error messages in your ser log or var/log/messages. Sure > > nothing else has changed? > > g-) > > > > ravi reddy wrote: > > Hi SER users, > > > > Iam using SER-0.9.6 with mediaproxy-0.5 and every think > > works fine except with the codecs . G711 a & g711 ulaw works fine but when i > > tuned grandstream settings to use g729 codec for pstn calls the call is not > > done by the Mediaproxy server . > > > > but when i > > created a g729 fake rtp generator it looks fine in sessions.py . so in > > order to forward the g729 phone call in to the pstn world what i have to do > > ?. > > > > Some folks told that "integrate SER with Asterisk Works!" is it really > > works for me? > > > > my pstn provider supports all codecs . ofcourse iam working in that > > company itself. > > > > please suggest some thing , so that I can bye pass this problem. > > > > > > Thank You. > > > > > > Regards, > > Ravi. > > > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

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[Serusers] Any plans to implement UAC module?
by Andy Thomas 31 Aug '06

31 Aug '06

Probably one for Greger- Does anyone know if there are plans to implement the UAC module (as seen in Openser) within the next version of Ser? I am currently running 0.9.6, and when I dial out to the PSTN, the CLI is shown as the Ser username, not the DID number All incoming DID numbers are specified as aliases, so when the PSTN dials in, it gets forward to the correct user. However I need to rewrite the outgoing SIP message, changing the the from: URI so that the alias is displayed, not the ser username. The UAC module is probably the easiest way changing the from: URI, unless anyone knows of a better way??

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Re: [Users] openser admin
by Mike Williams 30 Aug '06

30 Aug '06

On Monday 28 August 2006 03:15, raviprakash sunkara wrote: > Hi Mike , > Thanks for replying the mail , > openser administrator is wrriten in ruby on rails .. which is MVC structure Correct. > in database interface in web interface.. > If I commited in the subscriber in ur list... > In grp table also have to change from id to phplib_id.. No, that shouldn't be the case unless I'm mistaken. In the subscriber table, there was originally a phplib_id. I made the mistake of replacing it with id, instead of using both. The grp table never had a phplib_id, so id wasn't replacing anything. > > Tell me is openser administrator in fully stable and compitable with > openser. I see no reason why it should not be. > > But i think it may be unstable or stable version .. > i'm sure openser administrator 0.2 is unstable version > I need to know about the 0.3 .... > please for give me , if any mistakens in my secenrio .... Hopefully, they'll all be stable versions. However, it is still a very new project, and I cannot say, that with so few regular users of it, all the bugs are out. Mike Williams > > On 8/26/06, Mike Williams <mike(a)mikebwilliams.com> wrote: > > Peter & Sunkara, > > > > Thanks for the good idea, I hadn't even thought of uninstall or upgrade > > situations. > > > > The phplib_id problem can be solved by readding that column. There is a > > commented-out SQL statement in the SQL script that you can use. It is > > listed > > below as well. > > > > ALTER TABLE subscriber ADD COLUMN phplib_id varchar(32) NOT NULL default > > '' > > FIRST; > > > > I will work on these things today. Also, please let me know of any > > feature requests you have, otherwise I will have nothing to work on. > > > > Thanks, > > > > Mike Williams > > > > On Friday 25 August 2006 06:58, peter.3.edwards(a)bt.com wrote: > > > Mike, > > > > > > What's the upgrade procedure to go from 0.2 -> 0.3? > > > > > > Presumably readd the phplib_id column as per your mail on 21 Aug ... > > > and, what - just replace the old openseradmin-0.2 directory with this > > > one (including creating a new database.yml file)? > > > > > > Also, is there a way to uninstall OpenSER Administrator without > > > effecting OpenSER ... Presumably, just delete the openseradmin-x > > > directory ... But is there an equivalent SQL script to uninstall it's > > > database component? > > > > > > I think it would be good if you could include upgrade and uninstall > > > instructions within the release notes / on the website. > > > > > > Cheers, > > > > > > Peter. > > > > > > > -----Original Message----- > > > > From: users-bounces(a)openser.org > > > > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > > > > Sent: 24 August 2006 21:02 > > > > To: users(a)openser.org > > > > Subject: [Users] openser admin > > > > > > > > All, > > > > > > > > OpenSER Admin version 0.3 has been released. This fixes the bug where > > > > compatibility with openserctl was broken, and adds the > > > > ability to add other > > > > management accounts to the interface. > > > > > > > > Let me know what you think. > > > > > > > > Thanks, > > > > > > > > Mike Williams > > > > > > > > _______________________________________________ > > > > 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 > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users

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RE: [Users] openser admin
by peter.3.edwards＠bt.com 30 Aug '06

30 Aug '06

Hi, Mike. I've noticed that you've added Upgrade and Uninstall options to the OpenSER Administrator website - great stuff. Although, I did notice that your upgrade instructions don't mention anything about upgrading the database tables - I presume if you need to do that at a later date, you'll supply an equivalent SQL script (like the database_tables.sql and delete_database_tables.sql scripts for install and uninstall, respectively)? Might be worth mentioning that. As for your request for ideas for new features ... I think the documentation within OpenSER admin could do with a bit more content - I'm quite new to OpenSER and OpenSER Administrator so I'm not entirely clear what VoIP Accounts, Domains, Groups, Group Membership etc. specifically refer to (well, I get accounts and domains but it gets a bit fuzzier after that). Perhaps a little paragraph at the top of each page with a brief but clear description? On top of that, I guess I come less from a user-centric world and lean more towards the administrator side of things so I'm less interested in letting users login to change their profiles and more interested in seeing options that will allow easier adminstration - such as a web interface to configuring parts of the openser.cfg file - being able to toggle the debug level at will, stop/start/restart OpenSER from the web, change the port OpenSER listens on, enable/disable modules on startup - and, generally, anything that can be web-ified from the .cfg file. Obviously, I'm not talking about a replacement to the .cfg file - but something that can modify some of the various options on the fly would be useful ... Anyway, just thought I'd make some suggestions. I finally got the logging feature working today (oh, yes, when the log file doesn't exist, OpenSER doesn't handle the error very gracefully - so perhaps some work required on the error handling?) so I think I can safely say everything's working ... Many thanks, Peter. > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > Sent: 26 August 2006 15:07 > To: users(a)openser.org > Subject: Re: [Users] openser admin > > > Peter & Sunkara, > > Thanks for the good idea, I hadn't even thought of uninstall > or upgrade > situations. > > The phplib_id problem can be solved by readding that column. > There is a > commented-out SQL statement in the SQL script that you can > use. It is listed > below as well. > > ALTER TABLE subscriber ADD COLUMN phplib_id varchar(32) NOT > NULL default '' > FIRST; > > I will work on these things today. Also, please let me know > of any feature > requests you have, otherwise I will have nothing to work on. > > Thanks, > > Mike Williams -- Peter Edwards This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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Re: [Serusers] Serweb: HTTP 403 Forbidden error
by Roberto Lopes 30 Aug '06

30 Aug '06

Hi to all, My apache errror log just showed me this: [Tue Aug 29 15:45:57 2006] [error] [client 193.137.239.11] (13)Permission denied: access to /serweb/ denied Roberto > > On 8/29/06, Karel Kozlik <karel(a)iptel.org> wrote: > > > > hi, > > what your apache error log say? > > I think there are no read rights for apache on serweb files. > > > > Karel > > > > Roberto Lopes napsal(a): > > > > > > Dear SER Users, > > > > > > > > > I've a problem with Serweb and I hope somebody could help me out > > > I know this problem was already talked about but i never saw a > > solution > > > to it, so I'll post it again hoping somebody will know how to solve > > this > > > mistake. > > > So the problem is that when i try to put serweb working in my firefox > > > browser the following error is shown: > > > > > > > > > Forbidden > > > > > > You don't have permission to access /serweb/index.html on this server. > > > > > > Additionally, a 403 Forbidden error was encountered while trying to > > use > > > an ErrorDocument to handle the request. > > > > > > > > ------------------------------------------------------------------------ > > > Apache/2.2.0 (Fedora) Server at xxx.xxx.xxx.xxx Port 80 > > > > > > > > > I think i installed everything correctly. > > > > > > I'll show some of the configurations I've made > > > > > > #1 - In "/etc/php.ini" I changed the memory limit from 8 to 12M > > because > > > of the Serweb extension classkit.so > > > … > > > memory_limit = 12M > > > > > > #2 - I used pear to install classkit.so > > > > > > pear install <path to classkit-0.4.tgz> > > > > > > #3 - In "php.ini" > > > … > > > ;;;;;;;;;;;;;;;;;;;;;; > > > ; Dynamic Extensions ; > > > ;;;;;;;;;;;;;;;;;;;;;; > > > … > > > extension=classkit.so > > > > > > #4 - In "/serweb-0.4/config/config_data_layer.php": > > > … > > > $config->data_sql->host[$i]['host']="<IP MySQL Server>"; > > > … > > > $config->data_sql->host[$i][ > > > 'user']="<DB User>"; > > > $config->data_sql->host[$i]['pass']="<User password>"; > > > .. > > > > > > #5 - In "/root/instalacao/serweb-0.9.4/config/config_paths.php" > > > > > > $config->root_path="/root/instalacao/serweb- 0.9.4/html/"; > > > > > > #6 - I created a log file > > > touch /var/log/serweb > > > chown apache /var/log/serweb > > > > > > #7 - I created a folder for SIP domain > > > mkdir /root/installation/serweb-0.4/html/domains/xxx.xxx.xxx.xxx(IP of > > SER) > > > > > > #8 - I copied the files and folder from the default folder > > > ("/root/installation/serweb-0.4/html/domains/_default/") to the one i > > > created. > > > > > > #9 - I specified the SIP domain in the file > > > "/root/installation/serweb-0.4/config/set_domain.php": > > > … > > > $config->domain = "xxx.xxx.xxx.xxx(IP of SER)"; > > > … > > > > > > #10 - I enabled logging in file > > > "/root/installation/serweb-0.4/config/config.php": > > > … > > > $config->enable_loging = true; > > > $config->log_file = "/var/log/serweb; > > > > > > #11 - In /etc/httpd/conf/httpd.conf I added an alias > > > … > > > > > > <Directory "/var/www/html"> > > > Options Indexes FollowSymLinks > > > AllowOverride None > > > Order deny,allow > > > Allow from all > > > </Directory> > > > > > > <IfModule mod_userdir.c> > > > UserDir disable root > > > #UserDir public_html > > > </IfModule> > > > > > > Alias /serweb/ "/root/instalacao/serweb-0.9.4/html" > > > <Directory "/root/instalacao/serweb-0.9.4/html"> > > > Options Indexes MultiViews > > > Options +FollowSymlinks > > > AllowOverride None > > > Order Deny,Allow > > > Allow from all > > > </Directory> > > > ... > > > > > > #12 - I then put some commands for Apache to begin when computer would > > > > > start > > > > > > chkconfig httpd on > > > service httpd start > > > > > > #13 - I introduced the following URL > > > > > > http://<IP of SER>/<alias i wrote in httpd.conf>/ > > > http://xxx.xxx.xxx.xxx/serweb/ > > > > > > > > > And then that HTTP 403 error shows up > > > > > > I really hope somebody can help me because I'm doing a VoIP project > > for > > > my school and I'd really like to have everything done in time. > > > > > > Thank you for your attention, > > > Roberto Lopes > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > >

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[Serusers] FW: [Serdev] variable header field value comparation
by Luis Silva 30 Aug '06

30 Aug '06

>From: "Luis Silva" <lfs12(a)hotmail.com> >To: serdev(a)iptel.org >Subject: [Serdev] variable header field value comparation >Date: Fri, 25 Aug 2006 13:44:53 +0000 > >Hi all!! > >I need to compare an header field value with an avp. I tried to use the >hf_value_exists("header_name","$value") but the header that I need to >compare isn't static. I replaced the header_name with an avp but it didn't >work. then I tried to use avp_check("$value", "eq/$hdr($header_name)")) >but again, it didn't work. am I doing something wrong? is there another >solution? > >regards, >Luís Silva > > >_______________________________________________ >Serdev mailing list >Serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev

3 3

[Serusers] Error with credentials
by Daniel Mirrha Santos 30 Aug '06

30 Aug '06

Does anybody know what is this error about "is_user_in(): No authorized credentials found (error in scripts)"? Thanks, Daniel

3 9

[Serusers] SIP soft client supporting G.728
by VIDOVIC Mato 30 Aug '06

30 Aug '06

Hi, I am looking for a Windows SIP soft client supporting G.728 codec for narrowband interworking with a PSTN gateway. Can someone help me? Best regards, Mato

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[Serusers] caller=callee, Not giving me Busy
by ram 29 Aug '06

29 Aug '06

Hi all I have added the Follwing Line in my ser.cfg if (!uri==myself) { t_relay(); break; }; But iam not getting Busy Tone Ram

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[Serusers] t_was_cancelled() equivalent function for Ser?
by Ricardo Carvalho 29 Aug '06

29 Aug '06

Hi, I'm using Ser and I think I need to use the same functionality that the function t_was_cancelled() from openser does. As much as I know, this functions still doesn't exist in Ser. Is there any other that does the same? (I need it to be able to skip sending a call to Asterisk voicemail every time that a call is hangup by the caller when it is still ringing on the callee. If I don't such kind of function, if the caller hungs up when still ringing the callee UA, Ser sends the call to voicemail erroneously, which sends a voicemail e-mail to the ringed person) Thanks, Ricardo.

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[Serusers] Via Format?
by Ricardo Martinez 29 Aug '06

29 Aug '06

Hello list. Sorry to ask this question here, but i can't find the answer in the RFC3261 Is this "VIA" permited according to the RFC? Via: SIP/2.0/UDP 69.20.6.205:15060, SIP/2.0/UDP 69.1.229.69:4081;branch=z9hG4bK-58e9ccfcef0f4c99915b8007cb9713e1-1156793814 It seems to be two VIA's in ONE line... Thanks Regards, Ricardo.-

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RE: [Users] Please help me in Openser Admin
by peter.3.edwards＠bt.com 29 Aug '06

29 Aug '06

> 1. Is openser adminstrator a web application? I mean, does it have > same function like "serweb" in ser? "OpenSER Administrator: Mission The mission is to provide a simple interface to OpenSER that can be quickly and easily modified and upgraded. It is not designed to provide an interface for the customer, but is to be used only for administration duties." http://openseradmin.sourceforge.net/ > 2. How to install the openser administrator in openser server? > I use openser1.1.0-tls and what packets does it need? OpenSER Administrator: Installation http://openseradmin.sourceforge.net/docs.php > 3.Is the openser administrator just used by the administrator to manage > his/her server? or The openser administrator can be used like a > serweb?(the client also can use it) because as I read from the reply > message in this forum, the serweb can not use for openser1.1.0-tls See answer to question 1. > 4. If it is not like serweb, can anybody tell me what is the suitable > web application for openser ? As far as I'm aware from reading the website and mailing list archives, the only web application designed for use with OpenSER is OpenSER Administrator. It sounds like it might be possible to modify SERweb but that would require going "under the hood" ... Ferianto, all of this information is easily found from the website - rather copying your mail to three mailing lists which each have a slightly different remit, you may find you'd get a better response if you do some initial research first and then direct any further questions you still have, to the most appropriate list instead of sending to all of them at once. Even Microsoft know ... How to ask a question http://support.microsoft.com/?kbid=555375 HTH, Peter. -- Peter Edwards This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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