sr-users June 2006

sr-users@lists.kamailio.org

203 participants
496 discussions

[Serusers] Routing users to other SIP server
by KaveH Aasaraai 03 Aug '06

03 Aug '06

Hi, I was wondering how I can route my SER users to other SIP servers, without need of authentication to other server. I mean this: My User -> Auth -> My SER My SER -> Auth -> Other SIP Server My User --------make call--------> My SER --------route call-------> Other SIP Server Thank you. Kaveh __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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RE: [Users] Privacy (*67)
by Pablo Delgado 28 Jul '06

28 Jul '06

Is there a module being develop for ser that give it b2bua capabilities or is there a b2bua that integrates well with openser? Pablo Delgado Digiport Miami LLC Phone: (305) 424-0016 Web: www.digiportmiami.com Email: pablo(a)digiportmiami.com -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Tuesday, June 20, 2006 4:01 PM To: Pablo Delgado Subject: RE: [Users] Privacy (*67) Pablo Delgado writes: > To other sip phones. then you need a b2bua that terminates the call and creates another one, where caller has been changed to anonymous. -- juha

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RE: [Users] Out of Memory
by Douglas Garstang 21 Jul '06

21 Jul '06

Bogdan, I upgraded to the current cvs version, and I think that has fixed it. With no change to openser.cfg, it's been running for about 3 hours now without any reported memory issues. Douglas. > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, June 26, 2006 10:20 AM > To: Douglas Garstang > Cc: Users(a)openser.org > Subject: Re: [Users] Out of Memory > > > Ok Douglas - please let me know if the latest changes in > resolver have > any effect on the leak you observed. > > regards, > bogdan > > Douglas Garstang wrote: > > >Bogdan, > > > >Don't appear to have ever received that message. > > > >To answer your questions: > >We are not using ENUM at all. We aren't using our own module > for DNS lookups. > >I will try the latest cvs. > > > >Doug > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Monday, June 26, 2006 8:33 AM > >>To: Douglas Garstang > >>Cc: Users(a)openser.org > >>Subject: Re: [Users] Out of Memory > >> > >> > >>See the list. I sent you a reply : > >> http://www.openser.org/pipermail/users/2006-June/005321.html > >> > >>regards, > >>bogdan > >> > >>Douglas Garstang wrote: > >> > >> > >> > >>>>-----Original Message----- > >>>>From: Douglas Garstang > >>>>Sent: Thursday, June 22, 2006 12:09 PM > >>>>To: Bogdan-Andrei Iancu > >>>>Cc: Users(a)openser.org > >>>>Subject: RE: [Users] Out of Memory > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>-----Original Message----- > >>>>>From: Douglas Garstang > >>>>>Sent: Wednesday, June 21, 2006 12:48 PM > >>>>>To: Bogdan-Andrei Iancu > >>>>>Cc: Users(a)openser.org > >>>>>Subject: RE: [Users] Out of Memory > >>>>> > >>>>> > >>>>>Bogdan, > >>>>> > >>>>>I finally managed to upload the memory dumps to pastebin. > >>>>> > >>>>> > >>Links are: > >> > >> > >>>>>Hopefully you can access them. > >>>>> > >>>>>Memory dump right after OpenSER was started: > >>>>>http://pastebin.com/723872 > >>>>>Memory dump after OpenSER running for 20min: > >>>>>http://pastebin.com/723890 > >>>>>Memory dump after OpenSER running for 50min: > >>>>>http://pastebin.com/723902 > >>>>> > >>>>>The problem started to occur between the 20 and 50 minute > >>>>>samples. About 20 calls had been processed in that time. > >>>>> > >>>>> > >>>>> > >>>>> > >>>>So... I'm wondering if anything has been determined with this... > >>>> > >>>> > >>>> > >>>> > >>>Hi Bogdan. Did you manage to find anything with this? > >>> > >>>Douglas. > >>> > >>> > >>> > >>> > > > > > > > >

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[Serusers] How to send instant message programmatically
by srinivas.sriram＠wipro.com 21 Jul '06

21 Jul '06

Hello, I am using SER-0.9.6. How to send instant message programmatically, how to build sip message and which function can by used(like t_uac, t_request). Can you help where and which to be changed in ser code. Thank you, Regards, Sriram Srinivas. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Users] Parallel ringing
by Andreas M. 19 Jul '06

19 Jul '06

Hello, how could it be done through AVP´s and database lookup, to do a parallel ringing. Following setup: If uri eg example(a)test.dom is called, this call should be forwarded in parallel to a pstn-gateway. That means, receive the AVP with the telephonnumber associated to uri example(a)test.dom, rewrite uri, rewrite host/port and forward in parallel to gateway. I tried "append_branch", but was not able to put in the AVP (phonenumber), rewrite host/port points directly to the gateway. I have no troubles to load the AVP, only to make is possible, that the SIP Client and PSTN Phone are ringing in parallel. Some suggestions ? regards, Andreas

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[Users] UAC behavior, specifically uac_replace_from
by TSS Support 14 Jul '06

14 Jul '06

Okay, I'm trying to use uac_replace_from to anonymize the From: in order to suppress ANI. I think I've got it, but it seems like the behavior isn't what I expect. First, the relevent config file section: if (search("Anonymous")) { uac_replace_from("","sip:anonymous@xxxxxxx.com"); append_rpid_hf("",";party=calling;id-type=subscriber;screen=yes;privacy=full"); log(1,"Made a call anon"); }; Now, the headers as they go in and come out: SIP MESSAGE 4 xx.7.96.185:5061() -> xx.7.96.82:5060() UDP Frame 4 30/Jun/06 08:30:57.4723 TimeFromPreviousSipFrame=0.0069 TimeFromStart=0.0204 INVITE sip:1360789xxxx@voip02.xxxxxxx.com SIP/2.0 Via: SIP/2.0/UDP xx.7.96.185:5061;branch=z9hG4bK-7fa6fcd6;rport From: Anonymous <sip:+136091xxxxxx@voip02.xxxxxxxx.com>;tag=8c07328919ade13ao1 To: <sip:1360789xxxx@voip02.xxxxxxxx.com> Call-ID: 51835bf5-ba1be6be@localhost CSeq: 102 INVITE Max-Forwards: 70 Contact: Anonymous <sip:+136091xxxxx@xx.7.96.185:5061> Expires: 240 User-Agent: Sipura/SPA2000-3.1.5 Content-Length: 311 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp SIP MESSAGE 6 xx.7.96.82:5060() -> xx.7.96.90:5060() UDP Frame 6 30/Jun/06 08:30:57.4791 TimeFromPreviousSipFrame=0.0007 TimeFromStart=0.0272 INVITE sip:360789xxxx@xx.7.96.90:5060 SIP/2.0 Record-Route: <sip:xx.7.96.82;ftag=8c07328919ade13ao1;lr=on> Via: SIP/2.0/UDP xx.7.96.82;branch=z9hG4bKe9d1.8c515a97.0 Via: SIP/2.0/UDP xx.7.96.185:5061;branch=z9hG4bK-7fa6fcd6;rport=5061 From: Anonymous <sip:36091xxxxx@voip02.xxxxxxxx.com>;tag=8c07328919ade13ao1 sip:anonymous@xxxxxxxx.com To: <sip:360789xxxx@xx.7.96.90:5060> Call-ID: 51835bf5-ba1be6be@localhost CSeq: 102 INVITE Max-Forwards: 69 Contact: Anonymous <sip:+136091xxxxx@xx.7.96.185:5061> Expires: 240 User-Agent: Sipura/SPA2000-3.1.5 Content-Length: 311 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Remote-Party-ID: "xxxxxxxx Customer" <sip:+136091xxxxx@xxxxxxxx.com>;party=calling;id-type=subscriber;screen=yes;privacy=full And finally what I'm seeing in the log: Jun 30 08:18:33 voip02 /usr/sbin/openser[4684]: DEBUG:uac:replace_from: removing display [Anonymous] Jun 30 08:18:33 voip02 /usr/sbin/openser[4684]: DEBUG:uac:replace_from: uri to replace [sip:+136091xxxxx@voip02.xxxxxxxx.com] Jun 30 08:18:33 voip02 /usr/sbin/openser[4684]: DEBUG:uac:replace_from: replacement uri is [sip:anonymous@xxxxxxxx.com] So what am I missing here? It looks like uac_replace_from is finding the right URI and replacing it, just not properly. I've already gone into CVS and I don't see any changes since 1.0.1 (which is what is running here), nor any relevent changes from SER compared to OpenSER. Any help? Please? -Keith

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Re: [Users] UA not receiving BYE from SPA 3000
by Kenny Chua 14 Jul '06

14 Jul '06

Hi Bogdan, Thanks for the reply. I'm still somewhat lost on the second point you told me. >Second, BYE is an within the dialog request so you need to do >loose_route() for it, Just relaying it based on RURI is not correct at >all. Does that mean in my loose route I have to have something like this: If(method=="BYE") { route(1); exit; } Here is my updated openser.cfg. Also, am I suppose to exit after a forbidden meesage or error messages? Or should I return? Please advise me. Thank you. debug=3 fork=yes log_stderror=no listen=192.168.1.5 # INSERT YOUR IP ADDRESS HERE alias=test.com port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" 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/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") 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", "/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") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); t_relay(); exit; } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- 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"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); xlog("Setflag cos behind NAT\n"); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:00[0-9]*@") { # International PSTN xlog("International PSTN\n"); strip(2); route(4); route(5); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); xlog("NAT Traversal Section\n"); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routing to PSTN gateway\n"); rewritehostport("psnt.com:5060"); avp_write("i:45", "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(); }; } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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RE: [Serusers] RTP proxy between two subnetworks with private @s
by Jose Soler 14 Jul '06

14 Jul '06

Hi Joao, No I was not able to solve the issue. It seems (this is my guess tough) that the Portaone RTP proxy assumes that it has one public IP adress, so the valid configuration to use it is Public Nt-Private Nt. I was not able to make it work in other configurations (neiher I got feedback from Portaone to do so). Nevertheless the code is available, so it could be modified...as long as you have the time and will to do so. I did not ;). Best regards, josé -----Original Message----- From: Joao Pereira [mailto:joao.pereira@fccn.pt] Sent: 19. oktober 2005 20:17 To: Jose Soler; serusers(a)lists.iptel.org Subject: Re: [Serusers] RTP proxy between two subnetworks with private @s Hello, did you made it to put the clients of networks A and B to call each other? I want to do the same, and tried a lot of SER/RTPproxy configurations, including the one in: /ser-0.9.0/modules/nathelper/examples/alg.cfg and also tried to run rtpproxy with the "-l 10.0.0.135/193.136.2.2" option. But I just was able to ring the phones (wen calling between networks), but the RTP doesnt pass... If you found the solution, please tell me. Thanks Joao Pereia www.fccn.pt Jose Soler wrote: > Hi, > > I am trying to figure out how to solve the follwoing problem. I have > two subnetworks, A and B, with different private ip adressing schemes > (IP@A <mailto:IP@A>) and (IP@B <mailto:IP@B>). > > SER is installed in a computer with network interfaces towards both > subnetworks. > SER's SIP signalling proxying operation works properly within the > subnetworks and when trying to set up a communication between users in > A and B. But in that last case, obviously there is no media at all > circulating among the subnetworks. > > Portaone's RTP proxy has been installed and configured in the computer > with interfaces towards both subnetworks where SER is installed. > > I am trying to configure SER so that, based on the nathelper module, > when communication between both subnetworks occurs, the RTP proxy is > involved and the communication (also media and not only signalling) is > possible. BUT I am making something wrong, becouse it does not work ... > > Can anyone give me a hand /hint? > Thanks a lot in advance / in any case. > > My SER config file is the following: > > > # > > # ----------- global configuration parameters ------------------------ > > /* Uncomment these lines to enter debugging mode > > debug=7 > > fork=no > > log_stderror=yes > > */ > > check_via=no # (cmd. line: -v) > > dns=no # (cmd. line: -r) > > rev_dns=no # (cmd. line: -R) > > fifo="/tmp/ser_fifo" > > fifo_mode=0662 > > alias=wirelessip.x.x.x > > alias=sip..x.x.x > > alias=x.x.x > > log_stderror=no > > debug=3 > > children=3 > > mhomed=1 > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > > loadmodule "/lib/ser/modules/mysql.so" > > loadmodule "/lib/ser/modules/sl.so" > > loadmodule "/lib/ser/modules/tm.so" > > loadmodule "/lib/ser/modules/rr.so" > > loadmodule "/lib/ser/modules/maxfwd.so" > > loadmodule "/lib/ser/modules/usrloc.so" > > loadmodule "/lib/ser/modules/textops.so" > > loadmodule "/lib/ser/modules/registrar.so" > > # Uncomment this if you want digest authentication > > # mysql.so must be loaded ! > > loadmodule "/lib/ser/modules/auth.so" > > loadmodule "/lib/ser/modules/auth_db.so" > > # For NAT support / media proxying > > loadmodule "/lib/ser/modules/nathelper.so" > > # ----------------- setting module-specific parameters --------------- > > # -- usrloc params -- > > #modparam("usrloc", "db_mode", 0) > > # Uncomment this if you want to use SQL database > > # for persistent storage and comment the previous line > > modparam("usrloc", "db_mode", 2) > > # -- auth params -- > > # Uncomment if you are using auth module > > modparam("auth_db", "calculate_ha1", yes) > > # If you set "calculate_ha1" parameter to yes (which true in this > config), > > # uncomment also the following parameter) > > modparam("auth_db", "password_column", "password") > > # -- rr params -- > > # add value to ;lr param to make some broken UAs happy > > modparam("rr", "enable_full_lr", 1) > > # For NAT > > # We will use flag 6 to mark NATed contacts > > modparam("registrar", "nat_flag", 6) > > # Enable NAT pinging > > modparam("nathelper", "natping_interval", 60) > > # Ping only contacts that are known to be > > # behind NAT > > modparam("nathelper", "ping_nated_only", 1) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > # initial sanity checks -- messages with > > # max_forwards==0, or excessively long requests > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops"); > > break; > > }; > > if ( msg:len > max_len ) { > > sl_send_reply("513", "Message too big"); > > break; > > }; > > # special handling for NATed clients; first, nat test is > > # executed: it looks for via!=received and RFC1918 addresses > > # in Contact (may fail if line-folding used); also, > > # the received test should, if complete, should check all > > # vias for presence of received > > if (nat_uac_test("3")) { > > # allow RR-ed requests, as these may indicate that > > # a NAT-enabled proxy takes care of it; unless it is > > # a REGISTER > > if (method == "REGISTER" || ! search("^Record-Route:")) { > > log("LOG: Someone trying to register from private IP, rewriting\n"); > > # This will work only for user agents that support symmetric > > # communication. We tested quite many of them and majority is > > # smart smart enough to be symmetric. In some phones, like > > # it takes a configuration option. With Cisco 7960, it is > > # called NAT_Enable=Yes, with kphone it is called > > # "symmetric media" and "symmetric signaling". (The latter > > # not part of public released yet.) > > fix_nated_contact(); # Rewrite contact with source IP of signalling > > if (method == "INVITE") { > > fix_nated_sdp("1"); # Add direction=active to SDP > > }; > > force_rport(); # Add rport parameter to topmost Via > > setflag(6); # Mark as NATed > > }; > > }; > > # we record-route all messages -- to make sure that > > # subsequent messages will go through our proxy; that's > > # particularly good if upstream and downstream entities > > # use different transport protocol > > record_route(); > > # loose-route processing > > if (loose_route()) { > > t_relay(); > > break; > > }; > > lookup("aliases"); > > # if the request is for other domain use UsrLoc > > # (in case, it does not work, use the following command > > # with proper names and addresses in it) > > if (uri==myself) { > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest authentication > > if (!www_authorize("com.dtu.dk", "subscriber")) { > > www_challenge("com.dtu.dk", "0"); > > break; > > }; > > save("location"); > > break; > > }; > > # native SIP destinations are handled using our USRLOC DB > > if (!lookup("location")) { > > sl_send_reply("404", "Not Found"); > > break; > > }; > > }; > > # forward to current uri now; use stateful forwarding; that > > # works reliably even if we forward from TCP to UDP > > if (!t_relay()) { > > sl_reply_error(); > > }; > > } > > # > > # Forcing media relay if necessary > > # > > route[1] { > > #if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){ > > # sl_send_reply("479", "We don't forward to private IP addresses"); > > # break; > > #}; > > #if (isflagset(6)) { > > force_rtp_proxy(); # I force everything through the proxy > > t_on_reply("1"); > > append_hf("P-Behind-NAT: Yes\r\n"); > > #}; > > if (!t_relay()) { > > sl_reply_error(); > > break; > > }; > > } > > onreply_route[1] { > > if (status =~ "(183)|2[0-9][0-9]") { > > fix_nated_contact(); > > force_rtp_proxy(); > > }; > > } > > > > > > > > >----------------------------------------------------------------------- >- > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] SER PRACK
by Razvan Nemesiu 12 Jul '06

12 Jul '06

Dear all, Does anyone know if SER (0.8.14) supports PRACK messages ? I want to know if it forwards them.I want to add in my configuration if(method==PRACK) {..} and I want to know if I can do that and if SER allows this configuration line. Best regards, Razvan

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[Users] remove_hf("Contact") after fix_nated_contact()
by Walter Schober 07 Jul '06

07 Jul '06

Hello! Just curious: I do remove_hf("Contact") right after fix_nated_contact() (for whatever reason). Then the remove_hf() removes anything but the new SIP URI from the message, leaving e.g. sip:user@received_ip:port in front of the headerfield following the Contact. e.g. Before: Some-Header: bla Contact: Name <sip:user@privateip:port> Authorization: bla Then: Some-Header: bla sip:user@public_ip:portAuthorization: bla How to remove that Contact: after having it nat_helped and then saved? br Walter

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sr-users June 2006