sr-users December 2004

sr-users@lists.kamailio.org

184 participants
400 discussions

by Amozurrutia Jesus

Does anybody know a solution besides B2BUA for billing/prepaid?? Thanks Jesus -----Original Message----- From: Mike Tkachuk [mailto:mike@yes.net.ua] Sent: Lunes, 20 de Diciembre de 2004 02:41 a.m. To: Suresh; serusers(a)lists.iptel.org Subject: Re: [Serusers] SER and B2BUA Hello Suresh, Sunday, December 19, 2004, 9:07:10 PM, you wrote: S> Hello All, S> Is anyone successfully used SER and VOVIDA as prepaid platform ?. S> Could anyone point me to some resource that might be helpful with S> some examples ? S> Thanks in advance S> -skt Generally it works, but looks like Vovida's b2bua is died. :( I will not recommend it, no support, no community, no development. -- Best regards, ~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,. Mike Tkachuk, ph:380-3433-47067 YES ISP, fx:380-3433-47067 Valova 17, mike|a|yes.net.ua Kolomyia, www.yes.net.ua Ukraine 78200 FWD: 66518 20.12.2004 ICQ# 57698805 MSN: mike_tkachuk|a|hotmail.com ~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 10 months

[Serusers] Gateway Numbers Translation

by Andr�s Parra L.

Dear sirs: I almost have implemented the PSTN termination service in my Sip Proxy using a third-party Gateway provider in Miami, but im having some trouble translating the phone numbers in my sip Proxy to GW numbers that the provider gave me for each user of my system that wanted to have a PSTN Termination Service. I was talking to the GW people and they told me that they don't know if the Remote-Party-ID header field is supported by them. I've been testing registering to my SIP Proxy with a GW number (provided by the GW people) as username ("sip:gateway#@mydomain.com") and succesfully called to anywhere i wanted. But the real issue comes when i register with a username of my domain ("sip:username@mydomain.com") the GW send me to an IVR and ask me for a pin number. I�m sending the Remote-Party-ID with append_rpid_hf(); Also, the GW people told me that they check the From field of the SIP Header for the GW numbers they gave me. If i�m interested in leave alone my clients numeration (which i am!!) i should have to rewrite the From Header Field which will cause interop problems. Guys i need some help, i�m lost, so close and so far at the same time. Is there a way to translate the numbers of my domain to the numbers of the gw provider? Best Regards Andr�s Parra L andresparra(a)ipsofactum.com www.ipsofactum.com Ipsofactum LTD. --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.

19 years, 10 months

[Serusers] Windows Messenger 5.1

by Michael Shuler

Windows Messenger 5.1 attempts to register but SER decides that its got a bad packet. Any idea what I'm doing wrong? if(uri == myself) { if(method == "REGISTER") { # Make sure they are a valid user on our proxy if(!www_authorize("bwsys.net", "sipfriends")) { # xlog("L_INFO", "Req Auth For %ct, URI = %ru"); www_challenge("bwsys.net", "1"); break; }; # xlog("L_INFO", "Registered Contact %ct, URI = %ru"); save("location"); break; }; }; U 12.221.249.222:2235 -> 198.88.216.84:5060 REGISTER sip:198.88.216.84 SIP/2.0..Via: SIP/2.0/UDP 172.25.25.100:13184..Max-Forwards: 70..From: <sip:MikeLaptop@198.88.216.84 >;tag=bbad1c9410db4c55bc92b5f1ee653c1f;epid=0bde29f1fa..To: <sip:MikeLaptop@198.88.216.84>..Call-ID: c2c09c85a7894f259e6be7cdf1 8ddc7c..CSeq: 1 REGISTER..Contact: <sip:172.25.25.100:13184>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, N OTIFY, ACK, REFER, BENOTIFY"..User-Agent: RTC/1.3.5369 (Messenger 5.1.0639)..Event: registration..Allow-Events: presence..Conte nt-Length: 0.... # U 198.88.216.84:5060 -> 12.221.249.222:2235 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 172.25.25.100:13184;rport=2235;received=12.221.249.222..From: <sip:MikeLaptop@198.88 .216.84>;tag=bbad1c9410db4c55bc92b5f1ee653c1f;epid=0bde29f1fa..To: <sip:MikeLaptop@198.88.216.84>;tag=469318dd5d1a71e9927f4c3d0 4c855c1.3684..Call-ID: c2c09c85a7894f259e6be7cdf18ddc7c..CSeq: 1 REGISTER..WWW-Authenticate: Digest realm="bwsys.net", nonce="4 1c7137ddee9e9583c50f481de82ff43d682d7d2", qop="auth"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0.... ############################################################################ ########### U 12.221.249.222:2236 -> 198.88.216.84:5060 REGISTER sip:198.88.216.84 SIP/2.0..Via: SIP/2.0/UDP 172.25.25.100:13184..Max-Forwards: 70..From: <sip:MikeLaptop@198.88.216.84 >;tag=6fe4da518cec45079592bab51b24c9a6;epid=0bde29f1fa..To: <sip:MikeLaptop@198.88.216.84>..Call-ID: 8c2c5a2c23b246c9acae80f1f0 f819f0..CSeq: 1 REGISTER..Contact: <sip:172.25.25.100:13184>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, N OTIFY, ACK, REFER, BENOTIFY"..User-Agent: RTC/1.3.5369 (Messenger 5.1.0639)..Event: registration..Allow-Events: presence..Conte nt-Length: 0.... # U 198.88.216.84:5060 -> 12.221.249.222:2236 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 172.25.25.100:13184;rport=2236;received=12.221.249.222..From: <sip:MikeLaptop@198.88 .216.84>;tag=6fe4da518cec45079592bab51b24c9a6;epid=0bde29f1fa..To: <sip:MikeLaptop@198.88.216.84>;tag=469318dd5d1a71e9927f4c3d0 4c855c1.3684..Call-ID: 8c2c5a2c23b246c9acae80f1f0f819f0..CSeq: 1 REGISTER..WWW-Authenticate: Digest realm="bwsys.net", nonce="4 1c713a5f7cb83c6a875ededbbd8c05990dcfcfc", qop="auth"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0.... # U 12.221.249.222:2236 -> 198.88.216.84:5060 REGISTER sip:198.88.216.84 SIP/2.0..Via: SIP/2.0/UDP 172.25.25.100:13184..Max-Forwards: 70..From: <sip:MikeLaptop@198.88.216.84 >;tag=6fe4da518cec45079592bab51b24c9a6;epid=0bde29f1fa..To: <sip:MikeLaptop@198.88.216.84>..Call-ID: 8c2c5a2c23b246c9acae80f1f0 f819f0..CSeq: 2 REGISTER..Contact: <sip:172.25.25.100:13184>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, N OTIFY, ACK, REFER, BENOTIFY"..User-Agent: RTC/1.3.5369 (Messenger 5.1.0639)..Authorization: Digest username="MikeLaptop(a)198.88. 216.84", realm="bwsys.net", qop=auth, algorithm=md5, uri="sip:198.88.216.84", nonce="41c713a5f7cb83c6a875ededbbd8c05990dcfcfc", nc=00000001, cnonce="34434462344244301731442301567622", response="bffa95785672e35e97197aded919c830"..Event: registration..Allo w-Events: presence..Content-Length: 0.... # U 198.88.216.84:5060 -> 12.221.249.222:2236 SIP/2.0 400 Bad Request..Via: SIP/2.0/UDP 172.25.25.100:13184;rport=2236;received=12.221.249.222..From: <sip:MikeLaptop@198.88. 216.84>;tag=6fe4da518cec45079592bab51b24c9a6;epid=0bde29f1fa..To: <sip:MikeLaptop@198.88.216.84>;tag=469318dd5d1a71e9927f4c3d04 c855c1.3684..Call-ID: 8c2c5a2c23b246c9acae80f1f0f819f0..CSeq: 2 REGISTER..Server: Sip EXpress router (0.8.14 (i386/linux))..Con tent-Length: 0.... ---------------------------------------- Michael Shuler, C.E.O. BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP) 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike(a)bwsys.net Customer Service: (877) 976-0711

19 years, 10 months

[Serusers] I drop some filds from subscriber and i have a problem now

by Andr�s Parra L.

Hi, I drop some filds from subscriber and i have a problem now. I droped ha1,ha1b, greeting,confirmation,sendnotification,and de user can`t register in the SER. I want to know if there`s a problem with this action, i think that those wasn't neccesary becouse we have our owner web server a w have another fields an tables. Please help me. --------------------------------- Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more.

19 years, 10 months

[Serusers] NAT & Madia Proxy.

by Mohamed Omar

I'm currently run ser with media proxy and it working fine if I force all calls throught the the media proxy... I want only calls destine to PSTN gateway to go through the media proxy for billing purpose and all other calls, I just want to fix the NAT and setup the call. I dont want the media to go throught my server if it not destine to PSTN network. Let me know how my config should be...Am new with SER and am stuck on this. below is my configure for NAT. Thanks in Advance. # -- nat ping -- modparam("mediaproxy", "natping_interval", 10) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # ------------------------- request routing logic ------------------- # main routing logic record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # -- nat code -- if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (client_nat_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { fix_contact(); force_rport(); }; }; if (method=="INVITE") { use_media_proxy(); t_on_reply("1"); setflag(6); }; # 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") { save("location"); break; }; # setflag(1); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log (1, "********************** Rewriting the host:port ***********************************"); rewritehostport ("PSTN Gateway Ip:5060"); }; }; if (method == "INVITE"){ t_relay(); sl_send_reply ("180", "Ringing"); break; } if (!t_relay()) { sl_send_reply("404", "Not Found"); break; }; # -- nat code ---- } onreply_route[1] { if (status=~"(183)|2[0-9][0-9]") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; if (status=~"[3-4]0[0-9]") { end_media_session(); break; }; } --------------------------------- Post your free ad now! Yahoo! Canada Personals

19 years, 10 months

[Serusers] one way voice for calls to forwarded to PSTN provider

by Mohamed Omar

Am using SER 8.14 and media proxy. Call between my UA are fine without any issue but when I fordward the calls to the PSTN service provider gateway the voice is one way. PSTN user call hear me but I can't. Below is my complete configure with Ip modified. Thanks in Advance. # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode */ listen=233.x.x.x. port=5060 #debug=20 #fork=no #log_stderror=yes reply_to_via=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" #loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" #loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.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") # -- acc params -- #modparam("acc", "log_level", 7) #modparam("acc", "radius_flag", 1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nat ping -- modparam("mediaproxy", "natping_interval", 10) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # ------------------------- 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; }; # 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 # setflag(1); if (loose_route()) { t_relay(); break; }; # -- nat code -- if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (client_nat_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { fix_contact(); force_rport(); }; }; if (method=="INVITE") { use_media_proxy(); t_on_reply("1"); setflag(6); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!radius_www_authorize("")) { # www_challenge("", "0"); # break; # }; save("location"); break; }; # setflag(1); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log (1, "********************** Rewriting the host:port ***********************************"); rewritehostport ("234.y.y.y.:5060"); }; }; if (method == "INVITE"){ t_relay(); sl_send_reply ("180", "Ringing"); break; } if (!t_relay()) { sl_send_reply("404", "Not Found"); break; }; # -- nat code ---- } onreply_route[1] { if (status=~"(183)|2[0-9][0-9]") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; if (status=~"[3-4]0[0-9]") { end_media_session(); break; }; } [root@simu ser]# --------------------------------- Post your free ad now! Yahoo! Canada Personals

19 years, 10 months

[Serusers] Here Is A Complete ser.cfg Example File

by Chris

Aj & Paul, This example is just what I was looking for, however I get an error on serctl start: In syslog I get “fatal open data base /var/lib.mailman/data/aliases.db No such file or dir…” and on console I get “cat /var/run/ser.pid No such file or dir…” Can you point me I the right direction? Chris --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.809 / Virus Database: 551 - Release Date: 09/12/2004

19 years, 10 months

RE: [Serusers] Backup GW

by Amozurrutia Jesus

You can set the timeout with the timer parameters for the TM module: modparam("tm", "retr_timer1p1", 1) modparam("tm", "retr_timer1p2", 2) modparam("tm", "retr_timer1p3", 4) modparam("tm", "retr_timer2", 4) modparam("tm", "fr_timer", 18) modparam("tm", "fr_inv_timer", 60) check http://www.iptel.org/ser/doc/modules/html/tm.html <http://www.iptel.org/ser/doc/modules/html/tm.html> for details. Regards, Jesus -----Original Message----- From: Vitor Brasileiro [mailto:vitorbrasileiromail@yahoo.com.br] Sent: Sábado, 18 de Diciembre de 2004 12:58 p.m. To: Amozurrutia Jesus; serusers(a)lists.iptel.org Subject: Re: [Serusers] Backup GW Thank u Jesus, it worked. Now, I have another question. How I set the failure timeout. It's taking too much tike to send to the secondary GW. Vitor Brasileiro ----- Original Message ----- From: Amozurrutia Jesus <mailto:jamozurrutia@mcmtelecom.com.mx> To: Vitor Brasileiro <mailto:vitorbrasileiromail@yahoo.com.br> ; serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> Sent: Friday, December 17, 2004 10:38 PM Subject: RE: [Serusers] Backup GW I'm using something like this: ###################################################################### # Route to main Gateway ###################################################################### route[1] { rewritehost("Main.Gateway.IP.Address"); t_on_failure("1"); t_relay(); } ###################################################################### # Route to secondary Mex. PSTN Gateway ###################################################################### failure_route[1] { log(1,"Main Gw failed or is busy. Trying secondary.\n"); rewritehost("Second.Gateway.IP.Address"); append_branch(); t_relay(); } Call will be routed to the second gateway if the first gateway is busy or down. Does anybody know a method for load balancong? Regards, Jesus -----Original Message----- From: Vitor Brasileiro [mailto:vitorbrasileiromail@yahoo.com.br] Sent: Viernes, 17 de Diciembre de 2004 06:27 p.m. To: serusers(a)lists.iptel.org Subject: [Serusers] Backup GW Hi list, My SER sends to a specific GW all the external calls, but I intend to have a secondary (backup) GW to terminate those calls would't be completed for some reason. Does nbody knows how? Thanks, Vitor Brasileiro.

19 years, 10 months

[Serusers] PSTN Fowarding Problem

by jht2

Hi: I am using SER to fowrad calls to a Cisco 2600 Gateway to PSTN call.But strange problem occured. 1, When I use X-lite to dial out,everything is OK.But when I use SJphone and some other hardware phone to dial out,the call cann't be fowarded and get "400" message from Cisco,which I found different from X-lite's call is there is an error message during the invite:"ERROR: extract_mediaip: no `c=' in SDP"(no apper during the X-lite's calling) . And If I get rid of the "fix_nated_sdp("1")" in the ser.cfg,no more 400 message feedback,the Sjphone and Hardware phone can connect with the PSTN number but only single way audio,it seems the RTP stream is abnormal.while get rid of "fix_nated_sdp("1")",the internal call between 2 X-lites is OK. 2. And another problem is while I use Radius for accounting,even I use X-lite make a successful call to Cisco,it hasn't start message in Radius log detail file,only "408" is logged as "Failed" logged on Acct-Status-Type.While internal calls between X-lites,the Radius log is properly correct with Invite 200 starts and 200 stops.The detail file is also pasted below. Any Advice? Thanks. A. Debug Log of Hardwarephone(Invite part,then get 400 bad requrest) : 11(23018) Sending: INVITE sip:008613381786981@84.233.140.73:5060 SIP/2.0 Record-Route: <sip:008613381786981@62.164.130.1;ftag=qy9DjT5ubwqB6Ttp;lr=on> Via: SIP/2.0/UDP 62.164.130.1;branch=0 Via: SIP/2.0/UDP 192.168.1.88:5060;rport=60487;received=218.82.26.6;branch=z9hG4bKSaoOR5zbirM6Xbvg Max-Forwards: 69 User-Agent: PA168S From: "8888" <sip:8888@62.164.130.1 >;tag=qy9DjT5ubwqB6Ttp To: "008613381786981" <sip:008613381786981@62.164.130.1 > Call-ID: 2GRBd0SARGev9jNA(a)192.168.1.88 Contact: <sip:8888@218.82.26.6:60487> CSeq: 1 INVITE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 312 CC-Diversion:sip:008613381786981@62.164.130.1 v=0 o=8888 08882186 49218023 IN IP4 192.168.1.88 s=SIP CALL c=IN IP4 192.168.1.88 t=0 0 m=audio 8000 RTP/AVP 4 18 0 8 3 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active . 11(23018) orig. len=744, new_len=968, proto=1 11(23018) lookup(): '008613381786981' Not found in usrloc 11(23018) check_self - checking if host==us: 13==9 && [84.233.140.73] == [127.0.0.1] 11(23018) check_self - checking if port 5060 matches port 5060 11(23018) check_self - checking if host==us: 13==12 && [84.233.140.73] == [192.168.1.17] 11(23018) check_self - checking if port 5060 matches port 5060 11(23018) check_self - checking if host==us: 13==12 && [84.233.140.73] == [62.164.130.1] 11(23018) check_self - checking if port 5060 matches port 5060 11(23018) check_self: host != me 11(23018) parse_headers: flags=-1 11(23018) parse_headers: flags=-1 11(23018) DEBUG:check_content_type: type <application/sdp> found valid 11(23018) ERROR: extract_mediaip: no `c=' in SDP 11(23018) DEBUG: t_addifnew: msg id=26 , global msg id=24 , T on entrance=(nil) 11(23018) parse_headers: flags=-1 11(23018) parse_headers: flags=60 B. ser.cfg # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line:-d) 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/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" # !! Nathelper loadmodule "/usr/local/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") modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("uri_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("auth_radius","service_type",15) modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions" ,1) modparam("acc", "radius_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_missed_flag", 3) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT #xlog #modparam("xlog", "buf_size", 8192) #tm modparam("tm", "fr_inv_timer", 400) # ------------------------- 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; }; # !! Nathelper # 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 is used); also, # the received test should, if completed,should check all # vias for rpesence 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 enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". 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 }; }; setflag(1); setflag(2); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) # Uncomment this if you want to use digest authentication if (uri==myself) { if (method=="REGISTER"){ if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; if (!check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id in REGISTERs"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting(the same value as in log_flag!) */ }; if (method=="ACK") { if (uri=~"sip:0[1-9][0-9]+@.*") { log(1, "ACK\n"); setflag(1); /* set for accounting(the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting(the same value as in log_flag!) */ }; if ( method=="BYE" || method=="CANCEL" ) { log (1, "BYE or CANCEL\n"); setflag(1); }; record_route(); if (uri=~"sip:00[1-9][0-9]+@.*") { rewritehostport("84.xx.xx.xxx:5060"); append_urihf("CC-Diversion:","\r\n"); forward(84.xx.xx.xxx, 5060); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT,enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to alltransactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing);look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } C,Radius Detail File: Sun Dec 19 19:52:19 2004 Acct-Status-Type = Failed User-Service-Type = Sip-Session Sip-Response-Code = 408 Sip-Method = Invite User-Name = "491001(a)62.xx.xx.xxx" Caller-ID = "sip:491001@62.xx.xx.xxx" Client-Port-DNIS = "sip:003238777697@62.xx.xx.xxx" Sip-Translated-Req-ID = "sip:003238777697@84.xx.xx.xxx:5060" Acct-Session-Id = "FE2AF475-2066-475F-B960-D51E8FE7D051(a)212.202.103.93" Sip-To-Tag = "n/a" Sip-From-Tag = "1913233880" Sip-Cseq = "15841" Client-Id = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "2e94ece290cdd8ce" Timestamp = 1103457139 Sun Dec 19 19:59:42 2004 Acct-Status-Type = Stop User-Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 8 User-Name = "491001(a)62.xx.xx.xxx" Caller-ID = "sip:491001@62.xx.xx.xxx" Client-Port-DNIS = "sip:003238777697@62.xx.xx.xxx" Sip-Translated-Req-ID = "sip:003238777697@84.xx.xx.xxx:5060" Acct-Session-Id = "FE2AF475-2066-475F-B960-D51E8FE7D051(a)212.202.103.93" Sip-To-Tag = "3312445751-681955" Sip-From-Tag = "1913233880" Sip-Cseq = "15842" Client-Id = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "2e94ece290cdd8ce" Timestamp = 1103457582

19 years, 10 months

[Serusers] SER 0.8.12, Polycom 300, and ignored CANCEL messages

by Jamin W. Collins

Anyone else seeing problems with Polycom 300 phones ignoring CANCEL messages from a SER proxy? I've got a number of SER proxies (running 0.8.12) that are receiving CANCEL messages and passing them on to the Polycom phone, but all indications are that the phone is just ignoring them (as it continues to ring). Below is an example capture: U 2004/11/08 15:10:32.802663 172.16.22.25:5060 -> 130.110.72.16:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 130.110.72.16..From: "Interaction Cen ter" <sip:8475787000@GLCU.ORG;type=ICConnectionCall>;tag=8500..To: <sip:187 42@172.16.22.25:5060>;tag=585588F8-A212D515..CSeq: 1 INVITE..Call-ID: 198f7 c0ca5c28e7f59759c1f1c024617@130.110.72.16..Contact:<sip:18742@172.31.22.12> ..Record-Route: <sip:18742@172.16.22.25;ftag=8500;lr=on>..User-Agent: Polyc omSoundPointIP-SPIP_500-UA/1.1.0..Content-Length: 0.... U 2004/11/08 15:10:47.634541 130.110.72.16:1136 -> 172.16.22.25:5060 CANCEL sip:18742@172.16.22.25:5060 SIP/2.0..To: <sip:18742@172.16.22.25:506 0>..From: "Interaction Center" <sip:8475787000@GLCU.ORG;type=ICConnectionCa ll>;tag=8500..Via: SIP/2.0/UDP 130.110.72.16..CSeq: 1 CANCEL..Call-ID: 198f 7c0ca5c28e7f59759c1f1c024617@130.110.72.16..User-Agent: ININ-EICSRVR01-9987 9649..Content-Length: 0.... U 2004/11/08 15:10:47.634635 172.16.22.25:5060 -> 172.31.22.12:5060 CANCEL sip:18742@172.31.22.12 SIP/2.0..Max-Forwards: 10..Record-Route: <sip :18742@172.16.22.25;ftag=8500;lr=on>..To: <sip:18742@172.16.22.25:5060>..Fr om: "Interaction Center" <sip:8475787000@GLCU.ORG;type=ICConnectionCall>;ta g=8500..Via: SIP/2.0/UDP 172.16.22.25;branch=z9hG4bK7377.54324bd5.0..Via: S IP/2.0/UDP 130.110.72.16..CSeq: 1 CANCEL..Call-ID: 198f7c0ca5c28e7f59759c1f 1c024617@130.110.72.16..User-Agent: ININ-EICSRVR01-99879649..Content-Length : 0.... U 2004/11/08 15:10:47.986990 172.16.22.25:5060 -> 172.31.22.12:5060 CANCEL sip:18742@172.31.22.12 SIP/2.0..Max-Forwards: 10..Record-Route: <sip :18742@172.16.22.25;ftag=8500;lr=on>..To: <sip:18742@172.16.22.25:5060>..Fr om: "Interaction Center" <sip:8475787000@GLCU.ORG;type=ICConnectionCall>;ta g=8500..Via: SIP/2.0/UDP 172.16.22.25;branch=z9hG4bK7377.54324bd5.0..Via: S IP/2.0/UDP 130.110.72.16..CSeq: 1 CANCEL..Call-ID: 198f7c0ca5c28e7f59759c1f 1c024617@130.110.72.16..User-Agent: ININ-EICSRVR01-99879649..Content-Length : 0.... U 2004/11/08 15:10:48.124132 130.110.72.16:1136 -> 172.16.22.25:5060 CANCEL sip:18742@172.16.22.25:5060 SIP/2.0..To: <sip:18742@172.16.22.25:506 0>..From: "Interaction Center" <sip:8475787000@GLCU.ORG;type=ICConnectionCa ll>;tag=8500..Via: SIP/2.0/UDP 130.110.72.16..CSeq: 1 CANCEL..Call-ID: 198f 7c0ca5c28e7f59759c1f1c024617@130.110.72.16..User-Agent: ININ-EICSRVR01-9987 9649..Content-Length: 0.... -- Jamin W. Collins "Never underestimate the power of very stupid people in large groups." -- John Kenneth Galbraith

19 years, 10 months

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sr-users December 2004