- sr-users - kamailio.org

[Users] Doubts about installation
by Javier Allende Astigarraga 13 Feb '07

13 Feb '07

i'm begining with openser , i've installed on ubuntu dapper but reading installation notes , always speak about openser-mysql-module, persistent, but i cant do apt.get install .... and if i seek i find files for http://packages.ubuntulinux.org/feisty/net/ only download and put con mysql/lib?? thanks for all any information about openserv config?? bye bye -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

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[Serusers] call forwarding issue - attr_destination() unwanted behavior - ottendorf
by tzieleniewski 13 Feb '07

13 Feb '07

Hi!! I am trying to set up the blind call forwarding with the usage of attr_destination() function. I do the following I load the user attributes from user_attrs and uri_attrs table and then according to the loaded parameters I do: if ($tu.call_forward == "blind" && $tr.forward_blind) { xlog("L_INFO", " route[CALL_FORWARD]: fwd \n"); attr_destination("$tr.forward_blind"); xlog("L_INFO", " route[CALL_FORWARD]: route(FORWARD) \n"); route(FORWARD); xlog("L_INFO", " route[CALL_FORWARD]: drop \n"); drop; } after the ser.cfg logic goes through this part I don't see any change in the request uri?? Please point me what do I miss? here is my database contents: mysql> select * from user_attrs where uid like 'hellboy%' and name like 'call_%'; +----------------------+--------------+-------+------+-------+ | uid | name | value | type | flags | +----------------------+--------------+-------+------+-------+ | hellboy(a)voip.touk.pl | call_forward | blind | 2 | 1 | +----------------------+--------------+-------+------+-------+ 1 row in set (0.00 sec) mysql> select * from uri_attrs where username like 'hellboy%'; +----------+--------------+---------------+----------------------+------+-------+--------+ | username | did | name | value | type | flags | scheme | +----------+--------------+---------------+----------------------+------+-------+--------+ | hellboy | voip.touk.pl | forward_blind | sip:tzl@voip.touk.pl | 2 | 1 | sip | +----------+--------------+---------------+----------------------+------+-------+--------+ 1 row in set (0.00 sec) Cheers -tomasz

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[Users] Presence server -Confused about the Notify
by H Quintana 13 Feb '07

13 Feb '07

Hello, I'm a newbie with the SIMPLE extensions. This week I started using the presence server module (with the configuration suggested in the wiki and 2 X-lite's) The softphones are sending the SUBSCRIBE and the PUBLISH messages. The NOTIFYs are received for the Event presence.winfo with some XML content: <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="0" state="full"> <watcher-list resource="sip:10092011@a.b.c.d" package="presence"/> </watcherinfo> But, when I add each softphone as a Contact for the other there is no XML content for the Event presence, and the contact's status never changes even if I do multiple changes. I noticed that in the table xcap_xml there is no data, Should I put some information manually? or the information should be inserted when receiving the PUBLISH? Am I missing something? Any ideas? Thanks for your help, Humberto Quintana ____________________________________________________________________________________ Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121

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[Users] several questions/concerns about branches
by Klaus Darilion 13 Feb '07

13 Feb '07

Hi! I hav the following questions about branch handling in openser 1.1.1: In route[1] I use the dispatcher to forward to the gateway 11.22.33.46. Thus, the DURI will be set. Further I use port 6060 to send calls to the GW. So far everything is ok. ... setting fr_timer to 2 seconds XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX route[1]: request's uri: $ru=sip:123456789@foobar.net route[1]: request's destination uri: $du=sip:11.22.33.46:5060 route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 route[1]: request's first branch: $br=<null> route[1]: request's all branches: $bR= route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net ... t_relay Then, if the GW does not reply within 2 seconds the failure route triggers: First question: Why is the DURI set to NULL now? ... entering failure route ERROR: no response from gateway or clients a1.net ... branches before ds_next_dst: $ru=sip:123456789@foobar.net, $du=, $bR= XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX failure_route[1]: request's uri: $ru=sip:123456789@foobar.net failure_route[1]: request's destination uri: $du= failure_route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 failure_route[1]: request's first branch: $br=<null> failure_route[1]: request's all branches: $bR= failure_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net ... ds_next_dst branches after ds_next_dst: $ru=sip:123456789@foobar.net, $du=, $bR=sip:123456789@foobar.net XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX failure_route[1]: request's uri: $ru=sip:123456789@foobar.net failure_route[1]: request's destination uri: $du= failure_route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 failure_route[1]: request's first branch: $br=sip:123456789@foobar.net failure_route[1]: request's all branches: $bR=sip:123456789@foobar.net failure_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net, sip:979004369911 ... activating branch_route ... t_relay Second question: In the branch route the send_socket is reported as NULL. AFAIK it will be copied from branches[0] to the new branch during ds_next_dst. Also the request is sent from port 6060 - thus it looks like the $fs is wrong in branch_route. ==================================================== branch_route[1]: request's uri: $ru=sip:123456789@foobar.net branch_route[1]: request's destination uri: $du=sip:11.22.33.45:5060 branch_route[1]: request's force_send_sock: $fs=<null> branch_route[1]: request's first branch: $br=sip:123456789@foobar.net branch_route[1]: request's all branches: $bR=sip:123456789@foobar.net branch_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net, sip:9790043699111 thanks klaus

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[Serusers] branch parameter
by Kapil Dhawan 12 Feb '07

12 Feb '07

Whats the importance of branch parameter. I am using RTC/1.2 library and its not generating branch parameter. _________________________________________________________________ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

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Re: [Users] Corrupted 'To' after uac_restore_from
by Mahesh Paolini-Subramanya 12 Feb '07

12 Feb '07

Bogdan I'm sorry - I commited the cardinal cut/paste sin (copying stuff incorrectly :-( ) Anyhow, I've *correctly* cut/paste the INVITE and BYE headers below, and it looks to me like the route headers *are* being rewritten correctly. So, it doesn't look like that was the issue - unless you are referring to something else? cheers --- Original Message ----- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: mahesh(a)aptela.com Cc: users(a)openser.org Sent: Thursday, February 8, 2007 3:29:26 AM GMT-0600 Subject: Re: [Users] Corrupted 'To' after uac_restore_from Hi Mahesh, I suspect you have a client that does not preserve the Route/Record-Route parameters. According to RFC326, the parameters must not be altered at al, but I saw UAC doing lowercase on the param's value. Most probably this is your case also - the uac module use B64 encoding of the old uri in RR param, so this is case sensitive. Get the full trace of the call and check the original RR param (inserted by openser in INVITE) against the one in put in the BYE. regards, bogdan ---- CORRECTED MESSAGE----- I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header. The weird thing is that it happens with only one end point (a Televantage SIP Trunk). With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems. The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway The ACKs and the 200 OK all have their From/To headers appropriately rewritten The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?). Any ideas? FYI, Openser 1.1.0 (latest CVS checkout), and modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","aptelavsf") --INVITE-- UAC->proxy INVITE sip:17038677006@69.25.47.136 SIP/2.0. Via: SIP/2.0/UDP 10.10.35.100;rport;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1. Allow-Events: message-summary. Allow-Events: presence. Allow-Events: refer. Max-Forwards: 70. Call-ID: 56796C1E(a)10.10.35.100. From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>. CSeq: 132610708 INVITE. Expires: 180. Supported: replaces. Supported: 100rel. Contact: <sip:6462174165@10.10.35.100>. Content-Type: application/sdp. Content-Length: 242. . User-Agent: TeleVantage-UA/7.50.4817. . v=0. o=TeleVantage 10671 0 IN IP4 10.10.35.100. s=phone-call. c=IN IP4 10.10.35.100. t=0 0. m=audio 6036 RTP/AVP 0 18 101. a=rtpmap:0 pcmu/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=ptime:20. ---INVITE--- proxy->gateway INVITE sip:17038677006@65.91.91.16:5060;transport=udp SIP/2.0. Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>. Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bKf67d.2d31dbb.0. Via: SIP/2.0/UDP 10.10.35.100;rport=5060;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1. Allow-Events: message-summary. Allow-Events: presence. Allow-Events: refer. Max-Forwards: 69. Remote-Party-ID: "Membersevice Queue" <sip:6462174165@flareon.aptela.com>;party=calling;id-type=subscriber;screen=yes;privacy=off. Supported: replaces. Call-ID: 56796C1E(a)10.10.35.100. From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>. CSeq: 132610708 INVITE. Expires: 180. Contact: <sip:6462174165@10.10.35.100:5060>. Content-Type: application/sdp. Content-Length: 262. User-Agent: TeleVantage-UA/7.50.4817. . v=0. o=TeleVantage 10671 0 IN IP4 10.10.35.100. s=phone-call. c=IN IP4 66.150.122.23. t=0 0. m=audio 59520 RTP/AVP 0 18 101. a=rtpmap:0 pcmu/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=ptime:20. a=nortpproxy:yes . ---BYE---gateway->proxy BYE sip:6462174165@10.10.35.100:5060 SIP/2.0. Via: SIP/2.0/UDP 66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1. To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Call-ID: 56796C1E(a)10.10.35.100. CSeq: 1 BYE. Content-Length: 0. Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>. Max-Forwards: 70. ---BYE--- proxy->UAC BYE sip:6462174165@10.10.35.100:5060 SIP/2.0. Record-Route: <sip:69.25.47.136;lr;ftag=1386496>. Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bK7791.48e0e2f5.0. Via: SIP/2.0/UDP 66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1. To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Call-ID: 56796C1E(a)10.10.35.100. CSeq: 1 BYE. Content-Length: 0. Max-Forwards: 69. -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] MAJOR FIXUP: avp access in reply route
by Bogdan-Andrei Iancu 12 Feb '07

12 Feb '07

Hi everybody, The ability of accessing/processing AVPs in reply route was a long debated and wanted functionality. Finally, OpenSER 1.2.x fixes this. The new code allows two types of behaviours: - use per message AVPs in reply route - no special synchronization is needed, but there is no persistence outside reply route. - use per transaction AVPS in reply route (as in failure route). the above behaviours apply to the reply routes triggered by TM. You can control this via the new TM parameter "onreply_avp_mode": http://www.openser.org/docs/modules/1.2.x/tm.html#AEN303 in default reply route only per message AVPs will be available. regards, bogdan

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[Users] invitation: SER User and Developer Meeting, collocated with the IETF, in Prague, March
by Jiri Kuthan 12 Feb '07

12 Feb '07

Ladies and gentlemen, boys and girls, be wholeheartly welcome to the SER user/developer meeting in Prague, March, colocated with the IETF meeting. For more information see http://www.iptel.org/ietf_meeting If interested in participation, just register yourself. Whereas this event is coming from the SER crowd, the content is going to be primarily technical and openser all in all draws greatly from SER. I believe this can be most useful for users(a)openser.org as well. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] ser log file
by zahra bahar 12 Feb '07

12 Feb '07

could we see the processes that ser has when receives each massage? is there any log file for ser in about this? --------------------------------- Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit.

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[Users] DB tables for presence module
by Reinhold Buchinger 12 Feb '07

12 Feb '07

Hello, Can you explain me the difference between the information stored in the "watchers" table and "active-watchers" table? When is a watcher insert in which table? Thanks a lot! Best regards, Reinhold --

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[Users] MEDIAPROXY - Cannot detect inactive rtp sessions, any other BYE fix method?
by Dan-Cristian Bogos 12 Feb '07

12 Feb '07

Hi Guys, I know that this is not the support channel of mediaproxy, but just hoping that somebody from ag-projects is monitoring this list as well. I found out that, in my case, the SIP-BYE issue is still persistent even by using mediaproxy . If I will stop sending rtp through mediaproxy (eg by killing my client from task manager, or loosing the internet), mediaproxy will still find the session active, like nothing happened and therefore the bill will be inaquarate. I am using the version 1.7.2, and as read in the change log, 1.8.0 didn't change anything on this chapter. Is there any other solution or workarround available? Thxs, Dan

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[Serusers] Patch for MySQL 5.0.19 and above in SER 0.9.x
by sip 12 Feb '07

12 Feb '07

I've attached a patch file for people to use (it really only adds 3 lines) to patch the my_con.c file in the mysql module for SER 0.9.x I've tested it for a week with inactivity, restarts of mysql, etc, and it holds up rather well, but of course I would be remiss if I didn't add a 'use at your own risk' disclaimer. ;) The basic problem is that, after MySQL 5.0.3, they changed the default behaviour of the mysql connection socket to NOT reconnect in the case of a timeout or a restart of the server. If you had SER up and running for a while, or if there were a glitch with your mysql, the connection would time out/disconnect, and SER would no longer be able to hit the database (requiring a restart of SER). The MySQL coders added an option to enable the reconnect ability of the mysql connection, but it didn't work until 5.0.19. SO... if you're using mysql 5.0.19 or above with SER 0.9.x (there may only be 3 of us doing this ;) ), here's a patch file for you to apply to my_con.c in your modules/mysql directory. I'd recommend backing up your my_con.c file before patching. N.

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[Users] ?opsr? - setup
by Michel Bensoussan 12 Feb '07

12 Feb '07

Hello I have some setup problems. I compiled and install as following: /prompt$ make mode=debug all prompt$ su prompt# make install /To run openSer I tried: /prompt# /usr/local/sbin/openser / and /prompt# /usr/local/sbin/openser D E / in both case there is the following output: / 127.0.0.1 [127.0.0.1]:5060 192.168.1.86 [192.168.1.86]:5060 192.169.4.86 [192.169.4.86]:5060 192.168.13.86 [192.168.13.86]:5060 127.0.0.1 [127.0.0.1]:5060 192.168.1.86 [192.168.1.86]:5060 192.169.4.86 [192.169.4.86]:5060 192.168.13.86 [192.168.13.86]:5060 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.86 [192.168.1.86]:5060 udp: 192.169.4.86 [192.169.4.86]:5060 udp: 192.168.13.86 [192.168.13.86]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.86 [192.168.1.86]:5060 tcp: 192.169.4.86 [192.169.4.86]:5060 tcp: 192.168.13.86 [192.168.13.86]:5060 Aliases: tcp: michel:5060 tcp: michel.local:5060 tcp: localhost:5060 udp: michel:5060 udp: michel.local:5060 udp: localhost:5060 //prompt# /Attached my config file ( /etc/openser/openser.cfg). On this config file, I set listen=192.168.13.86 so I expected the openSER to listen only 192.168.13.86. Actually I don't understand from where the application takes the openser.cfg file? I removed /etc/openser/openser.cfg but the application is still running. / /Regards, Michel. / / # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode */ fork=no log_stderror=yes /**/ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.13.86 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_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "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; }

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Re: [Users] A better way to count the length of a variable
by Tavis P 12 Feb '07

12 Feb '07

Ah efficient length counting, you're only a version away... Thanks Brandon Brandon Armstead wrote: > Tavis P wrote: >> I'm trying to find a fast way to count the length of a variable (and >> detect when the length exceeds a certain value) >> All i've come up with so far is a regex, does anyone know of a >> faster/better way to do this? >> >> >> Heres my current approach: >> !avp_check("$ci", "re/^.{1,128}$/g") >> >> >> tavis >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > Hello Tavis, > > Currently that is the only method available, however keep your eye > out for the release of OpenSER version 1.2.0, it was recently froze in > CVS and hopefully will be ready pretty soon, however here are some of > the methods that will be available: > http://openser.org/dokuwiki/doku.php/transformations:devel. > > Regards, > Brandon > >

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[Users] A better way to count the length of a variable
by Tavis P 12 Feb '07

12 Feb '07

I'm trying to find a fast way to count the length of a variable (and detect when the length exceeds a certain value) All i've come up with so far is a regex, does anyone know of a faster/better way to do this? Heres my current approach: !avp_check("$ci", "re/^.{1,128}$/g") tavis

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[Serusers] Problems with Rtpproxy traffic balancing in SER 0.10.x
by Angelo Taddeo 12 Feb '07

12 Feb '07

I'm using ser 0.10.x ( CVS version). About rtpproxy traffic balancing, I don't know where I have to put rtpproxy IP address and where I have to set $node variable value. How does SER obtain rtpproxy IP address ? What the relation between rtpproxy IP address and N0, N1.....etc ? What parameters have I to put in force_rtp_proxy() function? Only rtpproxy IP address? In my framework happens : when SER receives INVITE Request and 200 OK it rewrites the body of SIP message (SDP). The problem is that it doesn't put in connection information (c) rtpproxy IP address; it puts its IP address. So clients send their RTP packet to SER instead to rtpproxy. How I can force SER to rewrite correctly SDP so that clients receive rtpproxy IP address in INVITE and 200 OK SIP message? Thanks in advance Alekos --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail

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[Serusers] register with ser
by zahra bahar 12 Feb '07

12 Feb '07

Hi. I installed SER.when I want to register with xlite in ser.xlite phone sends INVITE to ser.but ser doesn't send ok or any reply. what is the wrong. wen I run ser I have: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.10.10.138 [10.10.10.138]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.10.10.138 [10.10.10.138]:5060 Aliases: tcp: Shahab2:5060 tcp: Shahab2.Sarv:5060 tcp: localhost:5060 udp: Shahab2:5060 udp: Shahab2.Sarv:5060 udp: localhost:5060 WARNING: no fork mode and more than one listen address found(will use only thethe first one) stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 112640 0(0) INFO: udp_init: SO_RCVBUF is finally 225280 0(0)WARNING: using only the first listen address (no fork) 2(6248) INFO: fifo process starting: 6248 2(6248) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(6248) WARNING: no fifo_db_url given - fifo DB commands disabled! --------------------------------- Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games.

3 2

RE: [Serusers] register with ser
by Su Kerry 10 Feb '07

10 Feb '07

1 0

[Users] NAT problem for PRACK replies
by Jeff Williams 09 Feb '07

09 Feb '07

Hi, I seem to have a problem replies to PRACKs from NATed UAs. The UA is behind the NAT on a private IP listening on port 5060. This port gets mapped to 1350 on the external IP by the NAT device. The INVITE goes out fine and gets the contact updated for NAT by: if (client_nat_test("3")) { setflag(7); force_rport(); fix_contact(); }; This then receives a 183 reply to which it sends a PRACK. However, the 200 response to the PRACK gets sent to port 5060 on the external IP => which of course doesn't get to the UA. Is there something I'm missing in my config? I have attached a trace of the call setup. Jeff No. Time Source Destination Protocol Info 19424 6.838985 203.24.54.17 202.74.164.44 SIP/SDP Request: INVITE sip:0893680949@proxy.sipone.com;user=phone, with session description Frame 19424 (941 bytes on wire, 941 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:0893680949@proxy.sipone.com;user=phone SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Max-Forwards: 70 Contact: Jeff <sip:610089001@192.168.10.30:5060;user=phone;transport=udp> User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 288 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 610089001 147590 147590 IN IP4 192.168.10.30 Session Name (s): Cisco 7912 SIP Call Connection Information (c): IN IP4 192.168.10.30 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 16384 RTP/AVP 0 18 8 101 Media Attribute (a): rtpmap:0 PCMU/8000/1 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:8 PCMA/8000/1 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 19428 6.842541 202.74.164.44 203.24.54.17 SIP Status: 100 trying -- your call is important to us Frame 19428 (655 bytes on wire, 655 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: 1350 (1350) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK26f185af4ccc0d42;rport=1350;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:0893680949@proxy.sipone.com;user=phone out_uri=sip:61893680949@202.74.164.72:5060;user=phone via_cnt==1" No. Time Source Destination Protocol Info 19429 6.842579 202.74.164.44 202.74.164.72 SIP/SDP Request: INVITE sip:61893680949@202.74.164.72:5060;user=phone, with session description Frame 19429 (1148 bytes on wire, 1148 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:61893680949@202.74.164.72:5060;user=phone SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Max-Forwards: 69 Contact: Jeff <sip:610089001@203.24.54.17:1350;user=phone;transport=udp> User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 288 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 610089001 147590 147590 IN IP4 192.168.10.30 Session Name (s): Cisco 7912 SIP Call Connection Information (c): IN IP4 202.74.164.44 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 63054 RTP/AVP 0 18 8 101 Media Attribute (a): rtpmap:0 PCMU/8000/1 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:8 PCMA/8000/1 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 19450 6.860058 202.74.164.72 202.74.164.44 SIP Status: 100 Trying Frame 19450 (445 bytes on wire, 445 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header From: "Jeff" <sip:0892209080@sipone.com:5060>;tag=1166274291 To: <sip:0893680949@202.74.164.72:5060;user=phone>;tag=669b638bbf9e0903fd8a2a66 CSeq: 1 INVITE Call-ID: 3478456911(a)192.168.10.30 Via: SIP/2.0/UDP 202.74.164.44:5060;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 Content-Length: 0 No. Time Source Destination Protocol Info 21001 8.339800 202.74.164.72 202.74.164.44 SIP/SDP Status: 183 Session Progress, with session description Frame 21001 (1064 bytes on wire, 1064 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:07 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Require: 100rel RSeq: 3067 Contact: <sip:610089001@202.74.164.72:5060;user=phone> Record-Route: <sip:202.74.164.72:5060;lr> Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 214 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 4912 6655 IN IP4 202.74.164.72 Session Name (s): SIP Call Connection Information (c): IN IP4 202.74.164.72 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 27182 RTP/AVP 0 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 21003 8.341340 202.74.164.44 203.24.54.17 SIP/SDP Status: 183 Session Progress, with session description Frame 21003 (1016 bytes on wire, 1016 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: 1350 (1350) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:07 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Require: 100rel RSeq: 3067 Contact: <sip:610089001@202.74.164.72:5060;user=phone> Record-Route: <sip:202.74.164.72:5060;lr> Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 214 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 4912 6655 IN IP4 202.74.164.72 Session Name (s): SIP Call Connection Information (c): IN IP4 202.74.164.44 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 63054 RTP/AVP 0 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 21044 8.397481 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21044 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21045 8.397734 202.74.164.44 202.74.164.72 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21045 (725 bytes on wire, 725 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291> Route: <sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.215a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 69 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21060 8.419413 202.74.164.72 202.74.164.44 SIP Status: 200 OK Frame 21060 (489 bytes on wire, 489 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.215a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:08 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 21061 8.419532 202.74.164.44 203.24.54.17 SIP Status: 200 OK Frame 21061 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:08 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 21070 8.429213 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 21070 (469 bytes on wire, 469 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 21611 8.897497 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21611 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21612 8.897734 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 21612 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 21625 8.909606 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 21625 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 22692 9.898092 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 22692 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 22693 9.898343 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 22693 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11078 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 22702 9.912093 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 22702 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 24851 11.899557 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 24851 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 24853 11.899808 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 24853 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11081 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 24866 11.911720 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 24866 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 27365 13.900358 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 27365 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 27366 13.900630 202.74.164.44 202.74.164.72 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 27366 (725 bytes on wire, 725 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291> Route: <sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.315a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 69 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 27426 13.938284 202.74.164.72 202.74.164.44 SIP Status: 200 OK Frame 27426 (489 bytes on wire, 489 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.315a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:14 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 27427 13.938422 202.74.164.44 203.24.54.17 SIP Status: 200 OK Frame 27427 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:14 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 27438 13.947350 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 27438 (469 bytes on wire, 469 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 30349 15.901399 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 30349 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 30350 15.901638 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 30350 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 30359 15.913057 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 30359 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol

4 4

[Users] avpops on reply_route
by Klaus Darilion 09 Feb '07

09 Feb '07

Hi! Once again I want to discuss the problem of avpops in reply_routes (e.g. http://www.openser.org/pipermail/users/2005-October/001281.html) I can bypass the syntax parser by putting the AVP-commands into a normal route block which is executed from the reply_route. As far as I understand this may cause problems as the AVP-ops are done without locking. But what may happen in worst case? Bad AVPs or a crash? thanks klaus btw: is there any other way to handle data from a reply_route ?

2 1

Re: [Serusers] failed to load mysql.so module
by Josef.Bajada＠go.com.mt 09 Feb '07

09 Feb '07

3 5

[Users] switch - pseudo vars
by Cseke Tamas 09 Feb '07

09 Feb '07

Hello, I'm using openser 1.1.0 I'd like to make a dinamic routing, the plan is radius set an avp, which contains the routeid, but my config is bad, and i don't know what can be the problem route($avp(i:20)); don't work so i tryied this way: switch($avp(i:20)) { case 0: route(0); break; case 1: route(1); break; case 2: route(2); break; } but this is bad too, can anybody give me an advice? the problem is in the line "switch($avp(i:20)) {" i guess, but switch can accept pseudo vars, so i don't kow what the problem caould be? Thanks any help, Tamas

3 5

[Users] using multiple DNS servers
by Will Quan 09 Feb '07

09 Feb '07

I have a primary and secondary dns server setup in resolv.conf, with a timeout option of 1 second.The idea is when the primary fails, my secondary server picks up the workload.When openser is running, and primary dns is down, I am seeing several re-transmissions of 200 OK (invite) to my originator *after* openser has finished processing ACK. I was thinking the final responses would stop reTX after the ACK was t_relayed. Have you seen such behavior? And better yet- Is there a solution for this?Thanks, Will

4 3

[Serusers] Radius accounting - User-Name parameter in the Start Acct-Status-Type
by tzieleniewski 09 Feb '07

09 Feb '07

Hi! Is it possible to configure acc_radius in such a way that it will send a User-Name parameter in the Accounting Packet because now it doesn't: rad_recv: Accounting-Request packet from host 127.0.0.1 port 32844, id=246, length=474 SER-Attr = "" Acct-Session-Id = "6012E956-D8D3-2367-979B-753579FE59FF(a)192.168.1.2" Sip-To-Tag = "572241288" SER-From = "tomix <sip:tomix@tezet.no-ip.org:7060>;tag=2020612254" SER-Flags = 8 SER-Original-Request-ID = "sip:hellboy@tezet.no-ip.org" Sip-Method = "INVITE" Sip-Cseq = 20886 Sip-Translated-Request-ID = "sip:hellboy@192.168.1.10:5060" Sip-Source-IP-Address = 192.168.1.2 Sip-From-Tag = "2020612254" SER-To = "<sip:hellboy@tezet.no-ip.org>;tag=572241288" SER-Digest-Username = "tomix" SER-Request-Timestamp = 1171023334 Calling-Station-Id = "sip:tomix@tezet.no-ip.org:7060" Sip-Source-Port = 7060 SER-Digest-Realm = "tezet.no-ip.org" Sip-Response-Code = 200 Called-Station-Id = "sip:hellboy@tezet.no-ip.org" SER-Response-Timestamp = 1171023341 Acct-Status-Type = Start Service-Type = IAPP-Register NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Cheers Tomasz

1 0

[Serusers] SER 0.10.99 (Carpo) - NAT call - long ringing after hang up
by Theodor Meier 09 Feb '07

09 Feb '07

hello, using my fritzbox fon with an account from Carpo, ser 0.10.99, to call a cisco 7960 behind the same fritzbox fon, that uses as well an Carpo account, an hang up, the called cisco 7960 continues ringing for a minute. (fritzbox - ser - (same) fritzbox - 7960) Vice versa the same. (7960 - fritzbox - ser - fritzbox) Not so when I use the 7960 with a sipgate account (7960 - sipgate - telefonica gateway - Carpo ser - fritzbox) to call the fritzbox. Is this a ser problem? Perhaps a problem, when the call-phone and the called-phone use the same IP? Or is this a configuration problem of ser? What can I debug to find where the problem is? ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

5 6

[Users] RTP fixes for mediaproxy
by Jeff Williams 09 Feb '07

09 Feb '07

Hi, I was having a few issues with mediaproxy when it came to re-invites and call setup where the SDP does not get sent with the first INVITE. I noticed this was due a couple couple of assumptions in mediaproxy which don't always hold: 1) That SDP is always set up first by the caller (i.e. in the INVITE) then updated by the called party (i.e. in the 200). This is not the case. The first SDP can be sent in the 200 response by the called party and the caller send their SDP in the ACK. 2) That RTP streams don't change once set up. This is wrong for reINVITEs. This would cause problems when a user agent sent a reINVITE to redirect RTP to on hold music for example - the RTP change would be ignored. I think this had an impact on some fax setups as well, but I haven't tested it. I found a patch to fix this by storing the streams in a hash by media type, but that is a limiting assumption also (maybe I want 2 audi streams?). The attached patch fixes this by changing the following: 1) This pathc allows a mediaproxy RTPSession to be set up by either the caller or called party. A 'caller' argument is passed to the RTPSession functions telling it if the caller or called party is doing the setup or update. 2) The official mediaproxy does not even look at SDP details for streams which are already set up. This patch lets every SDP packet update all of the RTP streams for that party (either caller or called). If you have issues with one-way-audio with media proxy after reINVITEs, have calls the first SDP from the called party or are just plain adventurous, please give it a go. It has been used in a production environment for a bit over a month now. The patch was made against mediaproxy 1.7.2, but will work against 1.8.0 as well since none of the rtphander.py code changed. Regards, Jeff

2 2

[Serusers] failed to load mysql.so module
by Josef.Bajada＠go.com.mt 08 Feb '07

08 Feb '07

Hi, I am having this problem with the mysql module on ser. 0(31212) ERROR: load_module: could not open module </opt/ser/modules/mysql/mysql.so>: /opt/ser/modules/mysql/mysql.so: undefined symbol: log 0(31212) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) I can confirm that mysql.so exists in that directory. I have mysql 5.0.27 installed. I have already tried to clean and make all from scratch but the problem remained. Any ideas how to solve this? Thanks a lot, Josef

3 2

[Users] set RURI with dispatcher module
by Klaus Darilion 08 Feb '07

08 Feb '07

Hi! I want to use the dispatcher module to use load-balancing/failover to gateways. Thus, I also want to change the domainpart of the RURI to be the IP address of the gateway - but dispatcher only sets the DURI. Is there an easy way to set the RURI/domain too or do I have to use: avp_write("$rU","$avp(i:666)"); ds_select_dst(...); avp_pushto("$ru","$du"); avp_pushto("$ru/username","$avp(i:666)"); Is there a more easy way? btw - I think above workaround would not work in failure_route as ds_next_dst adds a new branch but avp_pushto works against branches[0] - correct? thanks klaus

3 4

[Users] SIP Proxy - CAC
by Michel Bensoussan 08 Feb '07

08 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using OpenSER for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in OpenSER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

2 3

[Serusers] 0.9.x mysql.so: undefined symbol: log error
by Steve Blair 08 Feb '07

08 Feb '07

Hi: I keep getting the following error on a 0.9.x system. The latest 0.9.x version of SER was just download and built on this server but I continue to get this same error. I've check the mysql Makefile and it seems to be correct. What else could cause this error? ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Thanks,Steve

4 8

[Serusers] sequential hunting
by Steve Blair 08 Feb '07

08 Feb '07

Hello: I would like to implement sequential hunting. Hunting is when a call arrives for a given subscriber, no one answers the original number so the proxy populates the username portion of the r-uri with another number (from the database) and tries again. This process continues until all numbers in the database are exhausted. I have the "failover" working but the r-uri field is not updated on any interation though the failure_route. Does anyone have any ideas how to make the following work? Thanks,Steve # hunting (serial forking) if (avp_db_load("$ruri/username", "s:allow_hunt")) { if (avp_check("s:allow_hunt", "eq/y/i")) { avp_delete("s:allow_hunt"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Enabled for <%ru>\n"); avp_db_load("$ruri/username", "s:huntlist/sam"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Started to <%ru>\n"); t_on_failure("6"); t_relay(); break; }; }; # Iterate through extension list for subscribers with call hunting enabled failure_route[6] { xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Failure Block #6: HUNTING for <%ru> from <%fu> at <%is>\n"); if (avp_pushto("$ruri/username", "s:huntlist/g")) { append_branch(); avp_delete("s:huntlist"); t_on_failure("6"); t_relay(); }; }

5 12

[Users] RADIUS acc ok, RADIUS auth not
by Luca Corti 08 Feb '07

08 Feb '07

Hello, I'm having a hard time getting RADIUS auth to work at all. I've reached the point where I am debugging the network traffic with tshark. I can see RADIUS packets generated from the openser box for accounting, but no RADIUS authentication packets are sent. Also, it seems no response comes back from the RADIUS server even for accounting requests. radclient against my radius server returns me an Access-Reject at least and the tshark dump contains auth request and response. OpenSER 1.1.1 tells me: Feb 6 17:27:04 openser openser[22596]: rc_send_server: no reply from RADIUS server I suspect the issues lie in the radiusclient-ng layer. I'm using radiusclient-ng from Debian Etch (0.5.3-2), but I see many branches in CVS, like RELENG_0_5_5 which seems newer. thanks

2 6

[Users] uac_replace_from() not work for Linksys ATA
by Howard Tang 08 Feb '07

08 Feb '07

HI All, I have a problem with uac_replace_from() function not working for Linksys Unit. This is what i have in the route[1], if (search("From:.*<sip:900*")) { uac_replace_from("anonymous","sip:anonymous@x.x.x.x"); } I have 0900xx as internal sip account, then I have to remove the 0900xx callerid before i send the call to PSTN gateway. Because 0900xx is not a valid number, Mobile phone providers block the call because of that, as a result, I need to replace the callerid to anonymous. I have tested with many other ATA and X-Lite without problem. The problem only ocurrs when i use Linksys ata. I have monitored the SIP message, the first request will change to anonymous then back to the original username. Anyone have an idea on how i can fix this ? -- Howard Tang ICQ : 259083 MSN : howard615(a)hotmail.com

2 7

[Serusers] Digest Authentication in Ottendorf server
by raj gopal 08 Feb '07

08 Feb '07

Hi, I can not use www_challenge(realm,qop)function in ser.cfg file even though auth module is loaded. The readme file has documentation for it. When I use the following in config file and start SER, the following error is thrown if (!radius_www_authorize("")) { www_challenge("", "1"); // parse error break; }; parse error (197,49-50): unknown command, missing loadmodule? I am using Ottendorf release. How to do digest authentication with new release??** <http://ftp.iptel.org/pub/ser/daily-snapshots/unstable/> -Venkat

2 1

[Users] Angle bracket in Route header coming from Contact and strict router
by Marcello Lupo 08 Feb '07

08 Feb '07

Hi to all, little question about populating the Route Header field. I'm dealing with 2 CPE of different vendors (Patton and Netsynt) and an Openser in the middle. Patton in their SIP is not using angle bracket on SIP From,To and Contact headers. Netsynt is using strict routing method to build BYE messages. This is causing that when Netsynt is creating the BYE it is putting the proxy IP in the R-URI and the URI of Contact received from previuos packets from patton in the Route header to let it be processed by the proxy. In this way the Route header field constructed is without the <> enclosing the uri. Netsynt keep it from previous Contact without any manipulation on it. This is causing an error on the proxy parsing the name-addr field of the Route header field and the loose_route() fail. Seems that it is mandatory for Openser to get angle bracket in Route header fields. Now in the section 20 of RFC3261 is said that it is not a MUST that From,To and Contact header have to be enclosed in angle bracket. So i'm searching in the RFC where is said that if the Contact is without the angle bracket the UAC have to check if the Contact have the <> and if it don't have the <> it have to insert it before to out it on the Route (i searched the RFC2543 too). can someone give me clarification on this issue? Where is written the Route header field is mandatory to have <> enclosing the URI. I'm searching all of this because i have to give a good reason to each vendor (Netsynt and Patton) to modify their sip stack for this issue. I don't know who of the 2 should fix the problem. If Patton to add <> in each request or Netsynt to fix the Route field population. Thanks, Bye, Marcello

2 1

RE: [Serusers] xlite and ser
by Su Kerry 08 Feb '07

08 Feb '07

1 0

[Serusers] xlite and ser
by zahra bahar 08 Feb '07

08 Feb '07

Hi all. could any one send me all the things that I should do for having only a call between two xlite phones with ser server? I have installed ser and xlite on 10.10.10.138 and another xlite phone on 10.10.10.195. the xlite phone on 10.10.10.195 sends register massage to 10.10.10.138 but xlite phone that is on 10.10.10.138 doesn't send massage to ser. I didn't change any thing in ser only fork=no,log-stderror=yes.children=4. and I don't user sql. thanks. --------------------------------- Access over 1 million songs - Yahoo! Music Unlimited.

1 0

[Users] Corrupted 'To' after uac_restore_from
by Mahesh Paolini-Subramanya 08 Feb '07

08 Feb '07

I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header. The weird thing is that it happens with only one end point (a Televantage SIP Trunk). With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems. The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway The ACKs and the 200 OK all have their From/To headers appropriately rewritten The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?). Any ideas? FYI, Openser 1.1.0 (latest CVS checkout), and modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","aptelavsf") ---INVITE--- endpoint->proxy From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8d200003+32cc4d0e. To: <sip:17038677006@69.25.47.136>;tag=277e3aa2408f2ad73a234f5bbb99133b.c13b. ---INVITE--- proxy->gateway From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136> Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> --- 183 Session Progress--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496. Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---183 Session Progress--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 ---200 OK--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---200 OK--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- gateway->proxy To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- proxy->endpoint To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. -- ******************************************* 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

[Users] qop="auth" with uac_auth
by Andreas Heise 08 Feb '07

08 Feb '07

Hello, since a lot of providers has change there authentication to qop="auth" the uac_auth(); function of the uac module can't use anymore. by google I found a lot of requests for qop with uac_auth and also the official feature request 1345887. Is a target date known for a solution, maybe it's possible with 1.2.0? Feature Request [ 1345887 ] Implement qop functionality in uac module p.s.: I know it's on the road map, but it is needed so often...... thanks. Andreas

4 10

[Users] Reg:NAT Routing
by venkid 08 Feb '07

08 Feb '07

hai, can any body help me to test audio call between two different private IPs. am using X-lite as client .am not able to get voice between two private ips and able to get voice with public IP .i have run media proxy saparatly. please find attached my openser.cfg file too. Thanks and Regards venkatesh.d debug=9 fork=yes log_stderror=yes listen=192.168.0.25 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openser@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/domain.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/presence.so" modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://openser:openser@localhost/openser") 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("presence", "db_url","mysql://openser:openser@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') modparam("presence", "totag_avpid", 987) modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "force_active", 1) 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=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE log("setting nat as yes *****************"); record_route_preset("192.168.0.25:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { log("setting falg 6 here **********"); setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="ACK") { route(1); exit; } else if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { # if (method="INVITE" || method=="ACK") { # end_media_session(); # }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

1 0

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

07 Feb '07

The OSP toolkit project has been moved from sipfoundry to sourceforge - http://sourceforge.net/projects/osp-toolkit Regards, Dmitry > Date: Mon, 08 Jan 2007 10:10:47 -0500 > From: Dmitry Isakbayev <dmitry(a)transnexus.com> > Subject: Re: [Serusers] Problemswith compiling ser 0.10.99 > To: "tzieleniewski" <tzieleniewski(a)o2.pl> > Cc: "'Support@Transnexus. Com'" <support(a)transnexus.com>, > serusers(a)lists.iptel.org > Message-ID: <45A25EF7.7070502(a)transnexus.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi, > > The OSP toolkit project is being moved from sipfoundry to sourceforge. > It should be available shortly. Meanwhile, the latest snapshot of the > toolkit can be downloaded from > > http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit.htm > > Regards, > Dmitry > > Date: Mon, 08 Jan 2007 11:25:24 +0100 > > From: "tzieleniewski" <tzieleniewski(a)o2.pl> > > Subject: [Serusers] Problemswith compiling ser 0.10.99 > > To: serusers(a)lists.iptel.org > > Message-ID: <54c54dce.20e2e287.45a21c14.f0624(a)o2.pl> > > Content-Type: text/plain; charset="utf-8" > > > > Hi All!! > > I am trying to compile the lattest cvs version of ser but I get errors > > which I cann't handle myself so i kindly ask for your help:) > > What I want to achive basically is the ser instance with presence > > support. > > I am compiling ser in the following way: > > make group_include="standard standard-dep stable" all > > I read that osp requires osptoolkit i tried to get it from sipfoundry > > but i couldn't find any source of that. > > I also cann't fight problems with tls i didn't includethe experimental > > modules(tls) but still I get some errors.

2 1

[Serusers] SIP Proxy - CAC
by Michel Bensoussan 07 Feb '07

07 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using SER (or maybe OpenSER) for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in SER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

1 0

[Serusers] Newbie ser user, moving from asterisk
by Antonio José dos Santos Brandão 07 Feb '07

07 Feb '07

Hi, I'm starting with ser and would like to know if it's possible no make the following: Today i use asterisk with sip users registered to it and a DeadAgi (a2billing) to manage calls and prepaid/postpaid accounts. Works fine. Since i have too many features and customizations i can't abone it. The problem is that many users are behind nat and have to canreinvite=no, forcing the rtp media to passthru asterisk. I would like to implement media_proxy in this scenario to balance the rtp's bandwith with another hosts. If there were some kind of media_proxy for asterisk would be great, but there isn't. So, i wonder: can I put ser in this, after or before asterisk, making possible to user media_proxy without change so much in asterisk setup? tks, -- Antonio J. S. Brandão

2 1

Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sahria Hao 07 Feb '07

07 Feb '07

Hi Edson. I want to know that why my Cisco AS 5300 didn't send BYE for SER...? Maybe... I doubt that maybe my 5300 have only dial-peer 6000 voice "POTS" configure for outgoing PSTN call. In case of PSTN incoming call have no problem about sending BYE for SER, Because it is apply dial-peer voice 5000 "VOIP" confiure as follows: [SER] <- [5300 (VOIP dial-peer)] <- [PSTN] So I'll try to re-configure my 5300 dial-peer, or please give me a hint If anyone have some way to solve this problem. Thanks, Sahria 2007/2/6, Edson <4lists(a)gmail.com>: > > I have this same behaviour, but never give it great importance, since we > didn't bill incomming calls… > > But it would be great to know if it's because of a misconfiguration or a > bug… but we notice that many ports become unavaliable (blocked) over time. > To release we programmed a reboot every day on 3AM…J > > Even with 'ngrep' the BYE, when PSTN side disconnects, didn't show up… > > Edson. > > ------------------------------ > > *From:* serusers-bounces(a)lists.iptel.org [mailto: > serusers-bounces(a)lists.iptel.org] *On Behalf Of *Sahria Hao > *Sent:* segunda-feira, 5 de fevereiro de 2007 08:36 > *To:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's > bug? > > Hi Greger, > > And I'm very sorry for my poor exposition. > > >>Do you get an error on the 5300? > > No, my 5300 works well and there's no error. > > >> Is it sent, but never reaches SER? > > No, when I finished call by PSTN side, 5300 didn't send BYE for SER. > >>Does SER receive, but does not recognize it? > > SER didn't receive a message from 5300 entirely. > > I think that when I finished this call, 5300 must send a BYE message for > SER... but didn't send it. > 2007/2/5, Greger V. Teigre <greger(a)teigre.com>: > > 09. [Cisco] can't send BYE for SER *****why??***** > > What does that mean?! Do you get an error on the 5300? Is it sent, but > never reaches SER? > Does SER receive, but does not recognize it? > g-) > > Sho Aihara wrote: > > Hi all. > > I have a problem for the following scenario. > When I make a call for PSTN and on hook by PSTN side, > Cisco As can't send BYE for SER. > > 01. [UA via Asterisk] dialing "08022223333" -> [SER] > 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") > -> [Cisco] > 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from > "033308022223333" to "008022223333" > 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] > 05. [e.g. Mobile] Catch call > 06. [SER] log CDR start > 07. [Cisco] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] can't send BYE for SER *****why??***** > 10. [UA via Asterisk] On hook > 11. [UA via Asterisk] Send BYE for SER > 12. [SER] log CDR End [Cisco] Call finished > > But another scenario, if make a call from PSTN to Asterisk and > on hook by PSTN side, Cisco As send BYE to SER. > > 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" > 02. [Cisco] receive "77771111" call number > 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: > my.ser.ip.address -> [SER] > 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] > 05. [UA via Asterisk] Catch call > 06. [SER] log CDR start > 07. [UA via Asterisk] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] Send BYE to SER > 10. [SER] log CDR End [Cisco] Call finished > 11. [UA via Asterisk] receive BYE from SER > > And sorry for my diffucult example. > > Why Cisco AS 5300 can't send BYE to SER > When PSTN call is disconnected by PSTN side? > > My ser.cfg as follows: > > # > -------------------------------------------------------------------------- > # global configuration parameters > # > -------------------------------------------------------------------------- > fork=no > log_stderror=yes > check_via=no > dns=no > rev_dns=no > listen=my.ser.ip.address > port=5060 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser:heslo@localhost/ser" > > # > -------------------------------------------------------------------------- > # module loading > # > -------------------------------------------------------------------------- > loadmodule "/usr/local/lib/ser/modules/mysql.so" > loadmodule "/usr/local/lib/ser/modules/sl.so" > loadmodule "/usr/local/lib/ser/modules/tm.so" > loadmodule "/usr/local/lib/ser/modules/rr.so" > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > loadmodule "/usr/local/lib/ser/modules/registrar.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > loadmodule "/usr/local/lib/ser/modules/auth.so" > loadmodule "/usr/local/lib/ser/modules/auth_db.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" > loadmodule "/usr/local/lib/ser/modules/exec.so" > > # > -------------------------------------------------------------------------- > # setting module-specific parameters > # > -------------------------------------------------------------------------- > modparam("usrloc", "db_mode", 2) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("rr", "enable_full_lr", 1) > modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") > modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("permissions", "db_url", "mysql://ser:heslo@localhost /ser") > 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("acc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("acc", "db_flag", 2) > modparam("acc", "db_missed_flag", 3) > > # > -------------------------------------------------------------------------- > # route pattern > # > -------------------------------------------------------------------------- > route { > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > > record_route(); > > if (loose_route()) { > if (method=="ACK") { > acc_db_request("01:CallStart\n", "acc"); > }; > if (method=="BYE" || method=="CANCEL") { > acc_db_request("02:CallEnd\n", "acc"); > }; > t_relay(); > break; > }; > > if (uri==myself) { > if (method=="REGISTER") { > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > break; > }; > save("location"); > break; > }; > > if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) > { > #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 > rewritehost("my.asterisk.ip.address "); > }; > > lookup("aliases"); > > if (!lookup("location")) { > if (method=="INVITE" && !search("^(f|From): > .*(a)(my\.cisco\.ip\.address <.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > break; > }; > if (uri=~"^sip:0[0-9]{10}@") { > # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 > prefix("0333"); > rewritehostport("my.cisco.ip.address:5060"); > avp_write("i:45", "inv_timeout"); > } else { > sl_send_reply("404", "Not Found"); > break; > }; > consume_credentials(); > }; > }; > > }; > > if (!t_relay()) { > sl_reply_error(); > }; > > } > > And my Cisco AS 5300 config as follows: > > voice call send-alert > voice rtp send-recv > > voice service pots > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > > voice service voip > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > sip > min-se 60 > > translation-rule 50 > Rule 0 0333 0 > Rule 1 ^7777 037777 > > voice class codec 2 > codec preference 1 g711ulaw > codec preference 2 g711alaw > > dial-peer voice 5000 voip > tone ringback alert-no-PI > description ser-asterisk-cisco-test > huntstop > destination-pattern 77771111$ > translate-outgoing called 50 > voice-class codec 2 > session protocol sipv2 > session target ipv4:my.ser.ip.address > dtmf-relay rtp-nte > max-conn 1 > > dial-peer voice 6000 pots > application session > max-conn 2 > destination-pattern 0333T > progress_ind alert enable 8 > translate-outgoing called 50 > port 0:D > > Thanks, > Sahria > > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > -- > ---------- > shosuke > msn : anseie(a)hotmail.co.jp > email : sahria.hao(a)gmail.com >

4 6

[Users] distinguish between fr_timer and fr_inv_timer timeout
by Klaus Darilion 07 Feb '07

07 Feb '07

Hi! If a failure route is executed - how can I distinguish between fr_timer timeout (a broken gateway) and fr_inv_timer timeout (nobdy took up the phone)? thanks klaus

3 4

[Users] 180 Ringing / 183 Session Progress
by Sam Lee 07 Feb '07

07 Feb '07

Hey Guys, Is it mandatory for SIP servers to generate to UA 180 Ringing to inform them that the called party is now ringing ? The reason i'm asking is because I am seeing strange results using different UAs. Using SIP clients to call out , instead of seeing 180 Ringing , i see 183 session progress even before call was picked up. On some sip client , ring back can be heard. On some others , ring back cannot be heard. Is there any reason to suspect that these UAs needs to receive 180 Ringing before generating any ringback ? Regards, Sam

4 6

[Users] OpenSER v1.0.1 to v1.1.1
by Max Gregorian 06 Feb '07

06 Feb '07

Hi, with regards to the release of v.1.1.1 I have a few questions. Firstly, what is to become of v1.0.1? Are you recommending all users with 1.0.1 also upgrade directly to v.1.1.1? Secondly, does 1.0.1 suffer from most (if not all) the same bugs as 1.1.0? I am particularly interested in any documented issues with handling response ACKs for 4xx class errors as I am currently experiencing a few issues with these, and I noticed in the changelog for 1.1.1 that a bug with ACK matching for non-200 class responses has recently been fixed. I ask this I noticed you are gradually phasing out all documentation to do with 1.0.1 on the openser website. As such, is 1.0.1 to be deprecated?

2 3

[Serusers] Has anyone got Jabber module to work w/SER
by SER LIST 06 Feb '07

06 Feb '07

Subject says it all. I am trying to make this work. I have installed JABBERD2 with a PyMSNT gateway. I can use a jabber cliient (exodus) and connect to Jabberd server and communicate w/ other Jabber users and MSN users. I now want to forward IM messages from Eyebeam UA to Jabber users and MSN users. I followed all instructions. When the message gets forwarded to SER and it executes the "xjab" function, I get an error saying "Invalid Content-Type". Any pointers will be highly appreciated. _________________________________________________________________ Buy what you want when you want it on Sympatico / MSN Shopping http://shopping.sympatico.msn.ca/content/shp/?ctId=2,ptnrid=176,ptnrdata=08…

1 0

[Users] avp_db_load question
by Klaus Darilion 06 Feb '07

06 Feb '07

Hi! What are the return values of avp_db_load? E.g. is it FALSE on problems accessing the DB or FALSE if it did not found any AVP to load? I couldn't find this documented somewhere. regards klaus -- Klaus Darilion nic.at

2 1

[Serusers] Anyone using CDRTool ?
by Lokesh Kumar 06 Feb '07

06 Feb '07

Hello All, Is there anyone using CDRTool in conjugation with ser, asterisk, radius setup. Can anyone give me a hand to setup it properly? Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.27/671 - Release Date: 2/5/2007

1 0

[Serusers] Request for Daniel-Constantin Mierla
by Giovanni Chirico 06 Feb '07

06 Feb '07

Hi Daniel, Im trying to use the dispatcher module. Im using the call forwarding ser.cfg file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack. I have inserted the function ds_select_dst("1", "0"); In the main route() and the function forward(uri:host, uri:port); in the PSTN Handler, route(5). How I can made the failure route, so if a gateway is down another gateway is load? In attachment there are my ser.cfg and dispatcher.list files. Please help me :-( Bye, Giò

2 3

[Serusers] automated response
by Myron Groening 06 Feb '07

06 Feb '07

Vacation Alert !! I will be out of the office, and will not be checking my message till FEB. 26 2007 Thanks Myron Groening

1 0

[Serusers] new version of radiusclient-ng
by Jan Janak 06 Feb '07

06 Feb '07

Hello, I just released a new version of radiusclient-ng library. The new version, 0.5.4, fixes problems on 64-bit architectures. You can download the new release from http://ftp.iptel.org/pub/radiusclient-ng/radiusclient-ng-0.5.4.tar.gz Credits go to Peter Nixon. Jan.

2 4

[Users] ERROR: module version mismatch for,/usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls,(i386/freebsd); module: openser 1.1.1-notls (i386/freebsd)
by Hansi Lao 06 Feb '07

06 Feb '07

Hello All, I have installed openser-1.1.1 with tls on my box. However, when I try to run openser. It returns me this error: ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd). Would someone be kind enough to help me how to compile a mysql.so which supports tls? Thank you.

2 4

[Users] migrating openser v1.1.x to v1.2.x
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, since we got in testing phase for 1.2.0 release, I think is better to start early a document with migration guidelines. Issues encountered during upgrade should be gathered together in the next page, to share and help each others: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x It is just the skeleton and few stuff there. Hopefully the content will grow quickly. Everybody can contribute. Cheers, Daniel

1 0

[Users] How can i make my OpenSER server with presence module work properly with the db_text database?
by Kc Ayuba 05 Feb '07

05 Feb '07

Hi, I'm trying to use db_text database with the presence module of OpenSER(1.2.x ). User agent is X-Lite(3.0). When I try to send a REGISTER message to OpenSER, OpenSER still works. But if i sent a SUBSCRIBE or a PUBLISH message, then openser crashes. If i use the MySQL database, then everything works just fine. But i prefer to use the db_text database. Here is the core file that was generated when OpenSER crashed. Core generated: Program terminated with signal 11, Segmentation fault. #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) bt #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 #1 0x080a3f6c in timer_ticker () at timer.c:141 #2 0x0806b51f in main_loop () at main.c:724 #3 0x0806d43b in main (argc=3, argv=0xaff0a2e4) at main.c:1438 (gdb) frame 0 #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) list 605 606 subs.from_domain.s = row_vals[from_domain_col].val.str_val.s; 607 subs.from_domain.len = strlen(row_vals[from_domain_col].val.str_val.s); 608 609 subs.event_id.s = row_vals[event_id_col].val.str_val.s; 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); 611 612 subs.to_tag.s = row_vals[to_tag_col].val.str_val.s; 613 subs.to_tag.len = strlen(row_vals[to_tag_col].val.str_val.s); 614 (gdb) 615 subs.from_tag.s = row_vals[from_tag_col].val.str_val.s; 616 subs.from_tag.len = strlen(row_vals[from_tag_col].val.str_val.s); 617 618 subs.callid.s = row_vals[callid_col].val.str_val.s; 619 subs.callid.len = strlen(row_vals[callid_col].val.str_val.s); 620 621 subs.contact.s = row_vals[contact_col].val.str_val.s; 622 subs.contact.len = strlen(row_vals[contact_col].val.str_val.s); 623 624 subs.cseq = row_vals[cseq_col].val.int_val; I really hope that someone can help me with this problem. Thanks in advance. Kind regards, Kc

4 9

RE: [Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

The significant difference of debug from a good call and openser is next. Below that are the full debug logs of both the openser and goodproxy call with an ngrep of openser's call in between. significant differences ======================== Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> VERSES --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: asterisk debug of openser call =========== --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 180 Ringing FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- Called openser/8886000 -- SIP/openser-082966e8 is ringing pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 200 OK FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 30660 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> <-- SIP read from openser:5060: BYE sip:3149@asterisk SIP/2.0 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f TO: <sip:3149@asterisk>;tag=as4aa53078 CSEQ: 1 BYE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 69 Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825 VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6017.0 P-hint: outbound --- (13 headers 0 lines) --- Sending to openser : 5060 (non-NAT) Transmitting (no NAT) to openser:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f To: <sip:3149@asterisk>;tag=as4aa53078 Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:3149@asterisk> Content-Length: 0 ngrep of above call ====================== # U +0.000616 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.067855 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 30660 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U +32.353338 openser:5060 -> asterisk:5060 BYE sip:3149@asterisk SIP/2.0. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. TO: <sip:3149@asterisk>;tag=as4aa53078. CSEQ: 1 BYE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 69. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825. VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. CONTENT-LENGTH: 0. USER-AGENT: RTCC/2.0.6017.0. P-hint: outbound. . # U +0.000734 asterisk:5060 -> openser:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser. Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. To: <sip:3149@asterisk>;tag=as4aa53078. Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:3149@asterisk>. Content-Length: 0. . good call from working proxy ======== We're at asterisk port 17508 Video is at asterisk port 11130 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to goodproxy:5060: INVITE sip:6000@goodproxy SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Feb 2007 14:55:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 381 381 IN IP4 asterisk s=session c=IN IP4 asterisk t=0 0 m=audio 17508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called exchange12/6000 pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport;received=goodproxy From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: M-Networks USR/1.0 Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE Content-Length: 0 --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- SIP/exchange12-082b8e18 is ringing pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTACT: <sip:goodproxy:5060> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 47688 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: ACK sip:goodproxy:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK6c3f8b53;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/exchange12-082b8e18 answered SIP/jonlaptop-082b0fb8 -- Attempting native bridge of SIP/jonlaptop-082b0fb8 and SIP/exchange12-082b8e18 Scheduling destruction of call '109c82647d6340be2a5ab19e7eb18a14@asterisk' in 32000 ms set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set de5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK47d7f855;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (local, 7776000, 3) exited non-zero on 'SIP/jonlaptop-082b0fb8' pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK47d7f855;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;tag=f51839ab98;epid=BD-70-82-06-F9 CSEQ: 103 BYE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- Destroying call '109c82647d6340be2a5ab19e7eb18a14@asterisk' pbx*CLI>

1 0

[Users] CVS frozen - testing phase of 1.2.0
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, starting with today the development for 1.2.0 is closed. No new features should be added until the release. The next month will be dedicated to testing, integration of submitted patches and fixing reported bugs. We have a lot of new code to test, your help is very much appreciated. I want to list the major changes you should focus on while testing: - usage of pseudo-variables and transformations is script (http://openser.org/dokuwiki/doku.php/transformations:devel) - TM timers (millisecond accuracy) - IP blacklists - DNS failover - automatic error handling - flags system (script flags, message flags, branch flags) - usrloc enhancements (licking system refurbished, database content optimization via DB fetch support) - new management interface (mi_fifo, mi_xmlrpc) - SIP Servlet Java Application server interoperability (seas module) - PERL AGI (perl module) - SNMP support via snmpstats module - Presence support (modules presence, pua, pua_usrloc, pua_mi) - xmpp IM gateway (xmpp module) - sip session timer support (module sst) - accounting clean up - federation based peering via domainpolicy module If I forgot something, please contribute. In addition, great help would be with creating 1.2.0 variants of documentation and dokuwiki pages (where is the case), testing the tools (openserctl, DB creation scripts) and developing scripts to migrate database from 1.1.x to 1.2.0. To keep track of discovered issues, please register them on the tracker -- it this way you make sure it is not lost in mail threads: http://sourceforge.net/tracker/?group_id=139143&atid=743020 Have nice testing sessions! Your feedback is always helpful! Cheers, Daniel

2 3

[Serusers] DB disconnects with MySQL 5.0.27
by sip 05 Feb '07

05 Feb '07

Something I've never run into before when I was using MySQL 4.x, but now crops up on MySQL 5.0.27... I get these errors: Feb 5 07:30:40 qabalah ser[26765]: submit_query(): MySQL server has gone away Feb 5 07:30:40 qabalah ser[26765]: db_update: Error while submitting query Feb 5 07:30:40 qabalah ser[26765]: db_upd_ucontact(): Error while updating database Feb 5 07:30:40 qabalah ser[26765]: wb_timer(): Error while updating contact in db I assume there's some sort of timeout of the SER connection to mysql? Is there a standard way to rectify this? Is this something new in MySQL 5? N.

1 1

[Users] Socket Errors
by Martin Burns 05 Feb '07

05 Feb '07

We have over 4 proxies running openser-1.1.0-notls. Recently we started to see the following error in the logs of one of the proxies: Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR: udp_send: sendto(sock,0xb3c35898,594,0,0xb3c257f4,16): Invalid argument(22) Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: msg_send: ERROR: udp_send failed Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR:tm:t_forward_nonack: sending request failed Now the errors suggests a binding to localhost, however our openser.cfg has the following: listen=eth0:5060 After a brief search we came across a couple of over people having a similar problem. Once of the suggested fixes was to add a mhomed=1 line to the cfg. Well we tried this (even though we are not multi homed) and the error changed to: Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: connect failed: Invalid argument Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: no socket found Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:add_uac: can't fwd to af 2, proto 1 (no corresponding listening socket) Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:t_forward_nonack: failure to add branches Does anyone have any suggestions or ideas as to what is causing this error. Thanks mburns

2 7

[Users] dbtext script
by Ovidiu Sas 05 Feb '07

05 Feb '07

Hi, Since dbtext is part of the standard, my suggestion would be to ship/install the sc.dbtext script along with the openserctl script. We can install it as openser_dbtext_ctl and in openserctl we can make a reference to it if no DBENGINE is provided: see the following patch for openserct (openser-v1.1.1): diff -Naurp sip-server/scripts/openserctl openser/scripts/openserctl --- sip-server/scripts/openserctl 2006-09-15 08:41:42.000000000 -0400 +++ openser/scripts/openserctl 2007-01-17 09:39:09.000000000 -0500 @@ -126,6 +126,8 @@ do $f done echo + echo "Note: Use openser_dbtext_ctl for DBTEXT operations!" + echo } Regards, Ovidiu Sas

2 3

[Users] auth_radius.so - ERROR: load_module: could not open module
by Luca Corti 05 Feb '07

05 Feb '07

Hello, I've built openser 1.1.1 from source with radius support. I installed radiusclient-ng. When I execute openser with to loadmodule auth_radius.so in my configuration I get: # openser 0(20268) ERROR: load_module: could not open module </usr/local/lib/openser/modules/auth_radius.so loadmodule >: /usr/local/lib/openser/modules/auth_radius.so loadmodule : cannot open shared object file: No such file or directory 0(20268) parse error (60,13-14): failed to load module 0(20268) parse error (60,14-15): syntax error 0(20268) parse error (60,14-15): ERROR: bad config file (3 errors) But the module is there. # ls -l /usr/local/lib/openser/modules/auth_radius.so -rwxr-xr-x 1 root staff 61369 Feb 5 10:33 /usr/local/lib/openser/modules/auth_radius.so This is really strange. ciao Luca

2 3

Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

have you checked this on the x-lite machine? regards, bogdan raviprakash sunkara wrote: > it didn't see any Ack packet is sending to SIP server, > > > On 2/5/07, *Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > I would say the X-lite does not obey the RR header - please try to get > an trace from the X-lite machine to see where the ACK is sent. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hi BogDan, > > > > Yes, I see Record-Route header in the 200 ok....... > > Please can u check the ngrepped of the X-lite > > with Hard phones iit is fine ........... > > Below one is ngrep from X-lite users, > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

1 0

[Users] default send-socket
by Klaus Darilion 05 Feb '07

05 Feb '07

Hi! What is the default send socket? Is it the first listen=..... statement in openser.cfg? Or can I manually set the default sendsocket? regards klaus

2 1

[Serusers] Re: load mysql.so failed
by Su Kerry 05 Feb '07

05 Feb '07

2 1

[Users] mediaproxy MySQL server has gone away error
by James Holden 05 Feb '07

05 Feb '07

Hi all, I'm having a problem with the mediaproxy rtp proxy. I'm using to do accounting into a MySQL database. After a period of inactivity, it loses it's connection to the MySQL server and the update queries begin to fail. Openser stays connected, however. The error in the logs is: error: accounting failed. MySQL error: (2006, 'MySQL server has gone away') I'm running: mediaproxy 1.8.0 openser-1.1.0 (using the RHEL3 rpm from the openser website) openser-mysql-1.1.0 (again using the openser RHEL3 rpm) Python-2.4.4 MySQL-python-1.2.1_p2 My Python skills aren't up to much. So far, I've tried added the following line into /usr/local/mediaproxy/modules/mysql.py just before the query call, to try and re-establish the connection: def query(self, query): self._query = query for c in range(0, 3): try: self.conn.db.ping() <<<<<<---- added by me return self.cursor.execute(query) except MySQLdb.OperationalError, why: if why[0] not in (CR.SERVER_GONE_ERROR, CR.SERVER_LOST): raise raise ... but it hasn't helped. Has anyone seen this before? Best regards, James -- James Holden AQ Limited Senior Systems Architect 13-15 Hunslet Road james.holden(a)uk.aql.com Leeds, LS10 1JQ 07989 702968 0800 389 5815 ===================================================== http://aql.com/documents/news/about_aql.pdf "The service has been perfect." BSI Inspectorate ltd "really really really useful. easy to use and reduces manpower." Fortnums "without a shadow of a doubt...excellent" Ruella James Plc ----------------------------------------------------- (aq) Limited 13-15 Hunslet Road Leeds LS10 1JQ 194 Harrogate Road Leeds LS7 4NZ http://www.aql.com Sales: 08707 449227 Fax: 08707 449228 PGP public key http://aql.co.uk/A-Q.txt Subject to our standard terms, found at http://aql.com/site/termsofservice.php ----------------------------------------------------- ****************************** CONFIDENTIALITY This e-mail is intended only for the use of the addressees named above and may be confidential or legally privileged. If you are not an addressee you must not read it and must not use any information contained in nor copy it nor inform any person other than (aq) limited or the addressees of its existence or contents. If you have received this email and are not a named addressee, please delete it and notify the (aq) limited customer service department via admin(a)a-q.co.uk. Many thanks. =========================================================

2 3

[Serusers] Dispatcher Module Help
by Giovanni Chirico 05 Feb '07

05 Feb '07

Hi, I would like to use the dispatcher module. I have configured it on my ser.cfg file for the PSTN calls and it works fine. However when the select gateway is unavailable the module don't switch to another gateway. Can you help me to write the "failure_route" so I can use another gateway when the first is unavailable? 'm using the "call forwarding ser.cfg" file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack (in attachment I'm sending you my ser.cfg and dispatcher.list files) Thanks in advance for your help. Bye, Giovanni Chirico

1 0

Re: [Users] strip_tail() string length?
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Just a typo correction On 02/05/07 13:54, Brandon Armstead wrote: > Daniel-Constantin Mierla wrote: >> Stripping is now possible with the new introduced 'transformations': >> http://openser.org/dokuwiki/doku.php/transformations:devel >> >> So if you want to strip last two characters of username in R-URI >> >> $var(x) = $(rU{s.len})-2; >> $rU = $(ru{s.substr,0,$var(x)}); ^^^ this is: $rU = $(rU{s.substr,0,$var(x)}); Cheers, Daniel >> >> Cheers, >> Daniel >> >> >> On 01/30/07 09:08, Juha Heinanen wrote: >>> Brandon Armstead writes: >>> >>> > i.e. strip_tail(strlen($ru) - 2); ... or something like such. >>> >>> sorry, but my suggestion >>> >>> substr($ru, -2, 2) >>> >>> does not work, since it returns the last two chars of $ru not the >>> remaining chars. strlen is indeed needed for the above. using php >>> substr and strlen >>> >>> strip_tail(strlen($ru) - 2) >>> >>> becomes >>> >>> substr($ru, 0, strlen($ru) - 2) >>> >>> -- juha >>> >>> _______________________________________________ >>> 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 >> > Hello Daniel, > > Glad to see this implemented, I think it will be a definite asset > to the future and improvements of OpenSER. Keep up the good work! > > Regards, > Brandon >

1 0

Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Ravi, so you see any Record-Route hdrs in the 200 OK reply? regards, bogdan raviprakash sunkara wrote: > Hi BogDan, > Good AfterNoon, and Thanks For replying to me, > > I used the Following code for Record_route(); > > route { > .... > ..... > if(method=="INVITE" && nat_uac_test("3") ) > { > record_route_preset("sip: > 192.168.2.3:5060 <http://192.168.2.3:5060>;nat=yes"); > }else if ( method ! = " REGISTER ") > { > record_route(); > } > } > where 192.168.2.3:5060 <http://192.168.2.3:5060> is the private IP of > the SIP server behind the NAT. > > > On 2/5/07, * Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Ravi, > > do you do record_route() for the INVITE? if not, the sequential > requests > of the dialog will bypass your server. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hello Users, > > > > When I tested the OpenSER server with CISCO ATA 186/188 IP phones > > SIP EyeP media (Softphnoes ) is working fine with RTP , when Both > > Users and SIP Server is in Behind the NAT, > > > > But Problem with Softphones ( X-lite ) . > > > > When I ngrep the Calls of X-lite users i didn't see any Ack > Packet to > > OpenSER After the Invite Packets received . > > > > Please Help me in this Part > > > > > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> > <mailto:ravi.sunkara@hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> <mailto: > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > www.hyperion-tech.com <http://www.hyperion-tech.com> > <http://www.hyperion-tech.com> > > > >------------------------------------------------------------------------ > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

2 2

Re: [Users] 200 ok not cought in ONREPLY
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Dan, that is rather strange if you are confident you are setting correctly a reply route (not default one) for the INVITE transaction. add the following debug message in modules/tm/t_reply.c, after line 1312: DBG("---------onreply route is %u\n",t->on_reply); recompile ans see the output on runtime. regards, bogdan Dan-Cristian Bogos wrote: > Salut Bogdan, > > theoretically I have. Saw now that with default onreply I can catch it > (log). It is really strange since I am relying both BYE and INVITES in > the same t_relay block with t_on_reply("1"); > t_on_failure("1"); but the 200 OK comes to default. > > Cheers, > Dan > > On 2/5/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> Salut Dan, >> >> are you sure you have an onreply route sent for this transactions? from >> the output I see no problems. >> >> regards, >> bogdan >> >> Dan-Cristian Bogos wrote: >> >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >> >

1 0

[Users] Hijack 'Move Temporarily'
by Sam Lee 05 Feb '07

05 Feb '07

Hey Guys, Normally if the SIP-UA is call-forwarded to a different number , it will send a 'Move Temporarily' to the proxy , which in turn relay this message to the originating gateway. The problem is , if they call forward to a number which the originating gateway doesn't know what to do with it , it will just kill the call. Can i hijack 'Move Temporarily' directly in the proxy and sending this call to a different gateway rather than relaying it to the originating gateway ? Please let me know if you can't understand me. Thanks! Regards, Sam

6 23

[Users] database persistent connection
by tele 05 Feb '07

05 Feb '07

Hi, I've an annoying problem with database persistent connection. I'm using mysql database for location and avp stuff. During the day all it's but after a day i retest some call and seams that the proxy lost connection with the database, and avp and location stuff didn't works anymore. I'm using this version of openser: # openser -V version: openser 1.2.0-dev8-notls (i386/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, 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 poll method support: poll, select, kqueue. @(#) $Id: main.c,v 1.23 2006/10/02 15:51:45 bogdan_iancu Exp $ main.c compiled on 12:38:52 Nov 27 2006 with gcc 3.4.4 i've also configure a small ping_interval for the mysql module modparam("mysql", "ping_interval", 60) modparam("mysql", "auto_reconnect", 1) and the usrloc: modparam("usrloc", "db_mode", 3) modparam("usrloc", "matching_mode", 1) I've also check that the NTP is correct on the servers. I don't know why openser lost connection with the database. any hints? thanks :tele

2 2

[Users] Help. in OpenSER in Post-paid and pre-paid
by raviprakash sunkara 05 Feb '07

05 Feb '07

Hello Users, I'm Using The Following software of My Sip productions, 1) OpenSER is a Sip Server,proxy, in front-end 2) Asterisk is PBX server, in Back-end 3) Radius for AAA to OpenSER, 4) Asterisk-java for IVR for Billing Customer care if I did any mistakes in these mail , forgive me . My Question is 1) If Caller calls to Callee , Before the call established to callee. The Caller has to know his Balance Account. if the Balance is there, then only the call has to establish, 2) I want the OpenSER total in RealTime application Server like Java application . Call Authorization ( Question no 1 ). If Asterisk is Front-end , ---> problem is Solved, How can a b2bua implement with OpenSER, please when -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 2

[Users] strip_tail() string length?
by Brandon Armstead 05 Feb '07

05 Feb '07

Hello everyone, I'm wondering what the easiest method would be to grab how long the RURI is and strip based on that information... or would I have to actually use regex, or is there an easier STRLEN() or LENGTH() operation? Any ideas, or is this something that could be implemented? Thanks. i.e. strip_tail(strlen($ru) - 2); ... or something like such. -- Brandon Armstead

3 5

[Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sho Aihara 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # module loading # -------------------------------------------------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.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" loadmodule "/usr/local/lib/ser/modules/exec.so" # -------------------------------------------------------------------------- # setting module-specific parameters # -------------------------------------------------------------------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") 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("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address"); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

3 2

Re: [Users] 200 ok not cought in ONREPLY
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, On 02/05/07 12:30, Dan-Cristian Bogos wrote: > Dear Daniel, > > many thanks for the reply. I can perfectly catch the OK now. ok > > The only thing which I must find out is why for a normal INVITE the OK > was not coming in the same onreply block as the BYE, since both INVITE > and BYE are relied in the same way out and processed theoretically in > the same route block. BYE is usually following the loose routing -- but I do not know if it is the case of your comfig. Cheers, Daniel > > Anyway, thank you again, it was really great help. > > Cheers, > Dan > > > On 2/5/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> to get a reply in an onreply_route other than the default on, you have >> to arm it via t_on_reply() when you relay the request. >> ... >> t_on_reply("1"); >> ... >> >> t_on_reply[1] { >> } >> >> Otherwise, all replies are processed in the default onreply_route (the >> one which has no index) >> >> onreply_route { >> .... >> } >> >> Cheers, >> Daniel >> >> On 02/01/07 15:48, Dan-Cristian Bogos wrote: >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >

1 0

[Users] OpenSER 1.2.0 + Presence
by Bernard Chan 05 Feb '07

05 Feb '07

Hello, I'm currently experimenting with OpenSER 1.2.0 with Presence module. Earlier I tried OpenSER 1.0/1.1 with PA. I'm using a third-party softphone which for some reasons, I would not like to disclose the name of it. After switching to 1.2.0+Presence, the softphone segfaults on startup. After tracing the source code of the phone and OpenSER and performing hundreds of packet sniffing over the past several days, I have grown to suspect that a certain behaviour with the new presence module may be the cause, but I would like to verify whether that is the case. For some reasons it is infeasible for me to modify the source code of the softphone used (actually I may be able to but that is what I would like to avoid). But it seems to choke on receiving a NOTIFY before the OK for the corresponding SUBSCRIBE is received by the softphone. So, any chance that I can test it by just modifying the config of OpenSER alone without changing the sources, both of OpenSER and the softphone? Thanks in advance for any invaluable insights. Regards, Bernard Chan.

2 1

[Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by raviprakash sunkara 05 Feb '07

05 Feb '07

Hello Users, When I tested the OpenSER server with CISCO ATA 186/188 IP phones SIP EyeP media (Softphnoes ) is working fine with RTP , when Both Users and SIP Server is in Behind the NAT, But Problem with Softphones ( X-lite ) . When I ngrep the Calls of X-lite users i didn't see any Ack Packet to OpenSER After the Invite Packets received . Please Help me in this Part -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 1

[Users] 200 ok not cought in ONREPLY
by Dan-Cristian Bogos 05 Feb '07

05 Feb '07

Hi Guys, By receiving incoming traffic from some other sip server (configured for different domain than mines) I will not be able to catch in the routing, onreply section, the 200 OK for the INVITE , OK which my client, behind my server, is sending to remote side, therefore not beeing able to modify sdp to use mediaproxy. Bellow is a level8 of debug of the OK. Is there any other way of catching this message in the routing? Any help will be appreciated. Thxs in advance, Dan Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply (status): Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: <SIP/2.0> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: <200> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: <OK> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: this is the first via Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After parse_msg... Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: forward_reply: found module nathelper, passing reply to it Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: this is the second via Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 234, <received> = <87.139.12.167>; state=6 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 235, <rport> = <5096>; state=6 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bK521554023>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: add_param: tag=8b2d7e45 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:parse_to:end of header reached, state=29 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to body [<sip:dan@de.babble.net>] Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: get_hdr_field: cseq <CSeq>: <20> <INVITE> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: forward_reply: found module tm, passing reply to it Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_check: msg id=1 global id=0 T start=0xffffffff Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=22 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=8 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_reply_matching: hash 39598 label 705981201 branch 0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:REF_UNSAFE: after is 1 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_reply_matching: reply matched (T=0x4081f430)! Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_check: msg id=1 global id=1 T end=0x4081f430 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:relay_reply: branch=0, save=0, relay=0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: 911, new size: 850 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= SIP/2.0 200 OK^M Via: SIP/2.0/UDP 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: <sip:dan@de.babble.net>;tag=8b2d7e45^M From: <sip:dan@babble.net>;tag=2240571840^M Call-ID: 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 telephone-event/8000^M a=sendrecv^M Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: add_to_tail_of_timer[2]: 0x4081f478 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., shmem=0x40821528: SIP/2.0 2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:UNREF_UNSAFE: after is 0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:destroy_avp_list: destroying list (nil) Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: cleaning up

3 2

Res: [Users] 407 and algorithm=MD5
by Bruno Machado 05 Feb '07

05 Feb '07

The section 3.2.1 (The WWW-Authenticate Response Header) from RFC 2617 says: "algorithm A string indicating a pair of algorithms used to produce the digest and a checksum. If this is not present it is assumed to be "MD5". " I send a INVITE to Openser but it send back a 407 (Proxy Authentication). The parameter response was OK. But, without the parameter 'algorithm', the proxy didn't allow the authentication. It neither consult at Radius. Looking at the RFC above, the SIP proxy must to use MD5 by default, but it don't accept the message without the parameter. Who needs to assume the algorithm? UAC or SIP Proxy? Can it be a bug of OpenSER? Or my interpretation is wrong? Best regards. Bruno Machado ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)openser.org Enviadas: Segunda-feira, 29 de Janeiro de 2007 14:21:31 Assunto: Re: [Users] 407 and algorithm=MD5 Hi Bruno, you cannot configure to have the algorithm param added. If I'm not wrong, the digest RFC says this is an optional param and the default value is MD5. regards, bogdan Bruno Machado wrote: > Hi. > > When I use the function proxy_challenge(), the 407 don't have the > parameter 'algorithm=MD5'. Look below: > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f". > ... > > How can I put this parameter in digest informations? > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f", > algorithm="MD5". > ... > > > Thanks in advance. > > Bruno > > __________________________________________________ > Fale com seus amigos de graça com o novo Yahoo! Messenger > http://br.messenger.yahoo.com/ > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

3 3

[Users] permission.allow example
by Ion Minzu 05 Feb '07

05 Feb '07

Hello , Can someone describe me a real example how to write rules in permission.allow, permission.deny. I use the example from file permission.* and it doesn't work, permission is set ALL to ALL. Or maybe exist other method to make call restrictions? #openser.cfg .... loadmodule "/usr/lib/openser/modules/permissions.so" .... modparam("permissions", "default_deny_file", "/etc/openser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/openser/permissions.allow") .... -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

2 1

[Serusers] Cisco AS 5300 can't send BYE for SER
by Sahria Hao 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # module loading # -------------------------------------------------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.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" loadmodule "/usr/local/lib/ser/modules/exec.so" # -------------------------------------------------------------------------- # setting module-specific parameters # -------------------------------------------------------------------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", " mysql://ser:heslo@localhost /ser") 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("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address "); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

1 0

[Users] OpenSER devel & SNOM 360
by mika.saari＠wipsl.com 05 Feb '07

05 Feb '07

Hi, I do have OpenSER 1.1.0 with TLS support on Debian Sarge working nicely, and my SNOM 360 registers against it nicely. When using OpenSER 1.2 (2 days old CVS build) with TLS on Debian Etch with SNOM 360 and own signed certificates (with tls/tools & own machine names) I get debug info (below) from OpenSER. Any ideas with that "peer did not send sertificate" ? Thank you so many times, -Mika --- clip --- Feb 2 13:15:10 localhost openser[20855]: tcp_receive_timeout: 0xb59e1900 expired (296, 297) lt=0 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:10 localhost openser[20855]: releasing con 0xb59e1900, state 0, fd=18, id=2 Feb 2 13:15:10 localhost openser[20855]: extra_data (nil) Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: reader response= b59e1900, 0 from 0 Feb 2 13:15:10 localhost openser[20859]: DBG: io_watch_add(0x8122820, 23, 2, 0xb59e1900), fd_no=15 Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: CONN_RELEASE 0xb59e1900 refcnt= 0 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: new tcp connection to: 193.65.183.233 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: on port 2063, type 3 Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Entered: Creating a whole new ssl connection Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Looking up socket based TLS server domain [193.65.183.13:5061] Feb 2 13:15:10 localhost openser[20859]: tls_find_server_domain: virtual TLS server domain not found, Using default TLS server domain settings Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Found socket based TLS server domain [0.0.0.0:0] Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 2 13:15:10 localhost openser[20859]: tcpconn_add: hashes: 462, 9 Feb 2 13:15:10 localhost openser[20859]: handle_new_connect: new connection: 0xb5a06c88 24 flags: 0002 Feb 2 13:15:10 localhost openser[20859]: send2child: to tcp child 0 6(20855), 0xb5a06c88 Feb 2 13:15:10 localhost openser[20855]: received n=4 con=0xb5a06c88, fd=18 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_add(0x8122980, 18, 2, 0xb5a06c88), fd_no=1 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_accept: Error in SSL: Feb 2 13:15:11 localhost openser[20855]: tls_error: error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Feb 2 13:15:11 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:11 localhost openser[20855]: releasing con 0xb5a06c88, state -2, fd=18, id=9 Feb 2 13:15:11 localhost openser[20855]: extra_data 0xb59f1a18 Feb 2 13:15:11 localhost openser[20859]: handle_tcp_child: reader response= b5a06c88, -2 from 0 Feb 2 13:15:11 localhost openser[20859]: tcpconn_destroy: destroying connection 0xb5a06c88, flags 0002 Feb 2 13:15:11 localhost openser[20859]: tls_close: Closing SSL connection Feb 2 13:15:11 localhost openser[20859]: tls_update_fd: New fd is 24 Feb 2 13:15:11 localhost openser[20859]: tls_shutdown: Shutdown successful Feb 2 13:15:11 localhost openser[20859]: tls_tcpconn_clean: Entered -- clip ---

2 2

[Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

Greetings list, I'm in the process of configuring Asterisk <-> OpenSER <-> Microsoft Exchange 2007 Unified Messaging. I have a windows based proxy that works great, and want to accomplish the same feat using OpenSER. Below are network captures from the working proxy and OpenSER. From comparing the working call and the OpenSER call it looks like OpenSER needs to modify the CONTACT header in its last "OK" response. I'm not sure that's right, but more importantly I'm not sure how to do this in the config. Any help or direction is greatly appreciated. Thanks, -jon p.s. Please cc me on any replies, I'm not currently subscribed to the list. openSER unsuccessfull call negotiation =========== # U +9.215930 asterisk:5060 -> openser:5060 INVITE sip:8885555@openser SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Contact: <sip:3149@asterisk>. Call-ID: 46109877480231241e3b8e9d57036e69@asterisk. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sat, 03 Feb 2007 18:01:48 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 216. . v=0. o=root 381 381 IN IP4 asterisk. s=session. c=IN IP4 asterisk. t=0 0. m=audio 13614 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U +0.000305 openser:5060 -> asterisk:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. From: "Jon Webster" <sip:3149@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Call-ID: 46109877480231241e3b8e9d57036e69@asterisk. CSeq: 102 INVITE. Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. Warning: 392 openser:5060 "Noisy feedback tells: pid=6890 req_src_ip=asterisk req_src_port=5060 in_uri=sip:8885555@openser out_uri=sip:5555@openser via_cnt==1". . # U +0.025313 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.016314 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. CONTACT: <sip:exchangeUM:5065;transport=Tcp;maddr=exchangeUM>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 59544 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. working proxy =================== # U +8.036801 asterisk:5060 -> goodproxy:5060 INVITE sip:5555@goodproxy SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sat, 03 Feb 2007 18:02:40 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 216. . v=0. o=root 381 381 IN IP4 asterisk. s=session. c=IN IP4 asterisk. t=0 0. m=audio 17160 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U +0.003872 goodproxy:5060 -> asterisk:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport;received=goodproxy. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 INVITE. User-Agent: M-Networks USR/1.0. Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE. Content-Length: 0. . # U +0.139238 goodproxy:5060 -> asterisk:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.010224 goodproxy:5060 -> asterisk:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTACT: <sip:goodproxy:5060>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 32458 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. . # U +0.000894 asterisk:5060 -> goodproxy:5060 ACK sip:goodproxy:5060 SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK440651a7;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U +5.475898 asterisk:5060 -> goodproxy:5060 BYE sip:goodproxy:5060 SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK5fabb75e;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 103 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U +0.028752 goodproxy:5060 -> asterisk:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK5fabb75e;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;tag=eb4f3cf30;epid=BD-70-82-06-F9. CSEQ: 103 BYE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 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) listen=openser 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 = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-cert.pem" #tls_private_key = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-privkey.pem" #tls_ca_list = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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; }; if (method=="INVITE" && uri=~"^sip:888[0-9]*@") { log("\n\nSIP invite received from exchange12\n\n"); strip(3); t_relay("tcp:exchangeUM:5065"); 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=="sip:openser:5060;transport=TCP") { sl_send_reply("200", "OK"); exit; } 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_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # 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") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "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; } route[2] { t_on_reply("1"); }

2 1

[Users] Change a Status Code From 182 to 180
by Marc Schaefer 05 Feb '07

05 Feb '07

Hello all, I've a small problem that causes a lot of problems and is in some way specific: I've a Openser 1.1.1 Server running on a public rootserver. User are registered with FritzBox ( DSL-Router with Voip-Stack) or a Soft/Hardphone being Nated. For the PSTN connectivity i've access to a PSTN Gateway from the BT. Everything works very nice with just one problem remaining. When one User gets a call from PSTN and picks it up, no problems. But the problem comes when a second call is made to the same User. His phone tells him that somebody is knocking and he can pickup the phone without any problem. The problem is on the callers side because he doesn't get any feedback that he is ringing/knocking, so most of the time he will just cancel the call. I did some research about it and here are my conclusions: The INVITE from the second call isn't answered with a "180 Ringing.." but with a "182 Queued.." which doesn't trigger a Ringing for the PSTN Call. When I talked to the BT support they did not really know about the status code 182 and could not give me an answer how their mediagateway will handle it ( They promised me to get the informtion :-) ). On the other side I can't change the Firmware so there's no way of persuading the DSL-Router to use 180 instead of 182. So my questions are: Can i change the status code at the proxy? If so, in which part of the routing logic do I have to change it? I tried a lot of different subst() and replace() commands but none had any effect on the sip messages it always kept being 182. Below you can see my configuration which is based on the template from ixcall ( http://www.ixcall.net/?q=node/32 ) with a lot of changes and support from the "getting started" guide from onsip.org, of course adapted to the openser 1.1.1 version I have attached my config. Keep in mind that this config will get through a m4 parser before being ready to be used ( mechanism can be seen at ixcall ). I hope my english is good enough to have my problem explained in a understandable manner Have fun Marc /* ==================================================================== Schnuffle Openser Configuration V6.3 Based on IxCall template V2.4 (http://www.ixcall.net/?q=node/32) And OnSip.org "Getting Started" Guid (http://siprouter.onsip.org/doc/gettingstarted/) ==================================================================== list of flags used 1 - Accounting 2 - Missed calls Accouting 3 - NAT SIP ping flag 4 - NAT invite detected 5 - registrar module nat_flag 6 - Enable NAT support 7 - Enable Trunk (pstn, asterisk or yate) 13 - Enable debugging -------------------------------------------------------------------- Routes 1 - Relay with NAT helper and no proxy authentication 2 - Method Register handler 3 - Method Invite handler (call routing) 4 - Trunk and PSTN gateway routing 5 - TLS routing ( not really used but still in config to keep the given message flow intact ) 6 - Relay with no NAT support and no proxy authentication 7 - User verify 10 - Relay with NAT helper and proxy authentication Reply 1 - NAT helper Fail 1 - Voicemail and message store ( stripped from all the voicemail and message storage stuff, so in fact just empty ) -------------------------------------------------------------------- */ #Global settings debug=3 log_facility=LOG_LOCAL7 fork=yes log_stderror=no mhomed=no # Multihomed - set this to no if you set a "listen" addres listen=OPENSER_IP # only want to bind to this ip port=OPENSER_PORT children=4 tcp_children=4 tcp_accept_aliases=yes alias=OPENSER_REALM # your realm alias=OPENSER_HOST # alias domains of your SRV records alias=OPENSER_IP # and for the die hard ip users. # uncomment to override config values for debugging /* debug=7 # debug level (cmd line: -ddd) fork=no port=OPENSER_PORT log_stderror=yes # (cmd line: -E) fifo="OPENSER_FIFO" */ dns=no rev_dns=no fifo="OPENSER_FIFO" fifo_mode=OPENSER_FIFO_MODE #TLS specific settings disable_tls = DISABLE_TLS #listen = tls:OPENSER_TLS_IP:OPENSER_TLS_PORT #tls_verify = 1 #tls_require_certificate = 0 #tls_method = SSLv23 #tls_certificate = "DEFAULT_TLS_CERTIFICATE" #tls_private_key = "DEFAULT_TLS_PRIVATE_KEY" #tls_ca_list = "DEFAULT_TLS_CA_LIST" mpath="OPENSER_MODULE_PATH" #path to openser modules loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "acc.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "group.so" loadmodule "xlog.so" loadmodule "enum.so" loadmodule "lcr.so" fifo_db_url="mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB" modparam("avpops|auth_db|alias_db|permissions|uri_db|usrloc|acc|msilo|group|domain|lcr|pdt","db_url", "mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB") modparam("auth_db", "calculate_ha1", CALCULATE_HA1) modparam("auth_db", "password_column", "PASSWORD_COLUMN") modparam("auth_db", "password_column_2", "ha1b") modparam("auth", "secret", "OPENSER_DIGEST_PW") #nathelper, mediaproxy or both can be used #we will only be using mediaproxy modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "MEDIAPROXY_SOCKET") #modparam("mediaproxy","sip_asymmetrics","MEDIAPROXY_SIP_ASYMMETRICS") #modparam("mediaproxy","rtp_asymmetrics","MEDIAPROXY_RTP_ASYMMETRICS") # -- lcr params -- modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","uri_scheme_column","uri_scheme") modparam("lcr","transport_column","transport") modparam("lcr","grp_id_column","grp_id") modparam("lcr","db_mode",0) # -- group params -- modparam("group", "table", "grp") modparam("group", "use_domain", 0) # might be usefull for future expansions modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "desc_time_order", 1) # -- registrar params -- modparam("registrar", "sip_natping_flag", 3) modparam("registrar", "nat_flag", 5) modparam("registrar", "desc_time_order", 1) # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- tm params -- #modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 90 ) #modparam("tm", "wt_timer", 20 ) modparam("tm", "via1_matching", 1) modparam("tm", "fr_inv_timer_avp", "s:inv_timeout") # -- permission params -- modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") # -- acc params -- modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "log_flag", 1) # -- alias_db params -- modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") #--enum params -- #modparam("enum", "domain_suffix", "e164.arpa.") # -- domain params -- modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # -- avpops params -- # avpops is used to store the local prefix from each customer in subscriber table # and append it as soon as a local call is done (meaning a call matching [^0][0-9]*) modparam("avpops","db_scheme","prefix-scheme:username_col=username;domain_col=domain;value_col=prefix;value_type=string;table=subscriber") modparam("avpops","avp_table","subscriber") modparam("avpops","use_domain",0) modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","avp_aliases","prefix=s:800") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Configuration flags # ----------------------------------------------------------------- NAT_SUPPORT_FLAG_6 TRUNK_ENABLED_FLAG_7 DEBUG_FLAG_13 # changing return code 182 to 180 so that the calling phone will ring # does not work at all #if (status=="182") { # xlog("Changing Status Code from 182 to 180"); # subst('/182 Queued/180 Ringing/ig'); #} #xlog("Replacing Status 1820 with 180"); #replace_all("182 Queued","180 Ringing"); if (isflagset(13) && !is_method("ACK") && !is_method("REGISTER")) { #debug route route(13); }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too Many Hops"); }; return; }; if (msg:len > max_len) { if (method!="ACK") { sl_send_reply("513", "Message Overflow"); }; return; }; if (status=="482") { #loop detection xlog("L_ERR", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE") { record_route(); if (client_nat_test("3") && isflagset(6)) { add_rr_param(";nat=yes"); }; }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (isflagset(6)) { if (has_totag() && (method=="INVITE" || method=="ACK" || method=="BYE")) { setflag(1); # not really clear which of the setflags triggered the "BYE" and "CANCEL" to be accounted setflag(2); if (client_nat_test("3")||search("^Route:.*;nat=yes")){ xlog("L_NOTICE","Loose Route: Using MediaProxy"); use_media_proxy(); }; }; }; route(6); return; }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (isflagset(6)) { if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); end_media_session(); }; }; # ----------------------------------------------------------------- # Unauthorized relay # ----------------------------------------------------------------- if (!is_uri_host_local()) { xlog("L_NOTICE","Prevent unauthorized Access is_uri_host_local=false, URI: $u, From: $fu -> Request: $ru\n"); if (is_from_local() || allow_trusted()) { # We prevent unautherised relays xlog("is_from_local or allow_trusted is true\n"); append_hf("P-hint: outbound\r\n"); route(10); # need auth } else { xlog("L_NOTICE","Uauthorized Relay: Reply a Forbidden\n"); sl_send_reply("403", "Forbidden"); }; return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon has gone for a beer"); return; }; # Calls matching means it is a local call without prefix # So prefix needs to be added expect for special numbers if (uri=~"sip:[1-9][0-9]*@" ) { # Check if uri contains special numbers ( 110,112 and some other 11[0-9]+ numbers ) # that should not be changed if (!uri=~"sip:11.*") { # add prefix to number # load user prefix and store it in $avp(s:800) # the prefix is stored in the subscriber table in an supplement attribute prefix if(!avp_db_load("$fu","$avp(s:800)/$prefix-scheme")) { xlog("Failed to load User Prefix"); } else {; # prefix R-URI xlog("Adding Prefix: $avp(s:800) to $ru"); subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:800)\1@\2;orig_uri=\1-\2/i'); xlog("R-URI: $ru"); } } } if (method=="INVITE") { route(3); return; } else if (method=="ACK") { route(1); return; } else if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); route(1); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE", "Lookup Aliases failed"); route(10); return; }; if (!lookup("location")) { xlog("L_NOTICE", "Lookup Location failed"); route(8); return; }; route(1); # Do not proxy authenticate - needed by system to send messages return; } route[1] { # ----------------------------------------------------------------- # Default Route relay Handler with NAT ENABLED # ----------------------------------------------------------------- if (isflagset(6)) { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (isflagset(4) || isflagset(5)) { xlog("L_NOTICE","Route(1) using Mediaproxy"); use_media_proxy(); }; t_on_reply("1"); }; route(6); # Relay the call return; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- # This cool register script came from http://openser.org/docs/scripting.html sl_send_reply("100", "Trying"); if (!route(7)) { # verify the user return(0); }; if (!search("^Contact:[ ]*\*") && client_nat_test("7") && isflagset(6)) { setflag(3); setflag(5); force_rport(); fix_contact(); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!www_authorize("", "subscriber")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization for $fu ($si) requested\n"); }; www_challenge("", "0"); return(0); } else { if (isflagset(13)){ xlog("L_NOTICE","OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization OK\n"); }; }; # Prohibit attempts to grab someone else's To address using valid credentials if (!check_to()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu -> $ru REPLY 401 - Unauthorized check_to() failed\n"); }; sl_send_reply("401", "Unauthorized - account and username must match"); return(0); }; consume_credentials(); }; if (!save("location")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location) Failed\n"); }; sl_reply_error(); return(0); }else { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location)OK\n"); }; }; return; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- setflag(1); #turn on accounting setflag(2); #also missed calls lookup("aliases"); #in the event that an alias resolves to another domain. #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE","No Local Call Check Auth I: r-uri: <$ru> from-uri:<$fu>\n"); route(10); #check authentication return; }; if (!lookup("location")) { xlog("L_NOTICE","Lookup Location Failed: r-uri: <$ru> from-uri:<$fu>\n"); if (src_ip==OPENSER_PBX_IP || !isflagset(7)) { # loopback prevention route(8); return; } else { # if it's not here, go hassle asterisk. xlog("L_NOTICE","Trunk Routing: r-uri: <$ru> from-uri:<$fu>\n"); route(4); return; }; }; t_on_failure("1"); #call waiting,busy,DND,voicemail route(1); # local normal routing return; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- avp_write("$avp(i:120)", "$avp(s:inv_timeout)"); xlog("L_NOTICE","Loading Gateways form Database"); if (!load_gws()){ xlog("L_ERR","****************************** LoadGateways Failed **********************"); sl_send_reply("500", "Internal server error, unable to load phone gateways"); return; }; if (!next_gw()) { xlog("L_NOTICE","******************** Didn'T found Any Gateways *************************"); sl_send_reply("503", "Service not available - No gateways"); return; }; # check if we route through BT # if so, change From Header domain to IP ( @callway-voip.de -> @212.65.6.4 ) # BT Gateway is in GatewayGroup 1 if (to_gw("2")) { xlog("L_NOTICE","Relaying to External Gateway, so rewriting From-HEADER:$fu\n"); #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); subst('/From:([^@]*)@callway-voip.de/From:\1@212.65.6.4/ig'); }; #if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][4] \ time [$Tf] Next Gateway $ruri_user_avp method <$rm> r-uri <$ru> 2nd via <$hdr(via[1])>\n"); #}; route(1); return; } route[5] { # ----------------------------------------------------------------- # TLS Handling # ----------------------------------------------------------------- # if (!is_uri_host_local()) { # if you have some interdomain connections via TLS # if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; # } else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; # }; # }; if (!is_uri_host_local()) { remove_hf("Alert-Info"); if (method=="INVITE" && proto==tls) { append_hf("Alert-info: 1\r\n"); #cisco 7960 append_hf("Alert-info: Bellcore-dr4\r\n"); #cisco ATA # append_hf("Alert-info: http://foo.bar/x.wav\r\n"); #snom }; }; return; } route[6] { # ----------------------------------------------------------------- # Route relay Handler # ----------------------------------------------------------------- #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); if (!t_relay()) { if (isflagset(6)) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; }; xlog("L_NOTICE","Route6: T_relay failed"); sl_reply_error(); }; return; } route[7] { # ----------------------------------------------------------------- # User verify # ----------------------------------------------------------------- # Check for known domain(s) if (!is_from_local()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Unknown domain\n"); sl_send_reply("403", "Unknown domain"); }; return(0); }; # Check to see that the user has not been disabled if (is_user_in("From", "disabled")) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Your account has been disabled\n"); }; sl_send_reply("403", "Your account has been disabled"); return(0); }; if (is_user_in("To", "disabled")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $tu That account has been disabled\n"); }; sl_send_reply("403", "That account has been disabled"); return(0); }; return(1); } route[8] { # ----------------------------------------------------------------- # Handling of unavailable user # ----------------------------------------------------------------- # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); return; }; t_newtran(); t_reply("404", "Not Found"); # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted acc_db_request("404 missed call", "missed_calls"); return; } route[10] { # ----------------------------------------------------------------- # Default Message Handler with Proxy Authentication # ----------------------------------------------------------------- if(method=="ACK") { #these you never proxy authenticate route(1); return; }; if(method=="BYE" || method=="CANCEL") { #these you never proxy authenticate setflag(1); route(1); return; }; xlog("L_INFO", "Proxy auth $fd $dP destination:$du $dd $ds"); if (!route(7)) { #verify the user return(0); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; # if (is_user_in("Credentials", "local")) { # Uncomment to use the group options route(1); # }else{ # sl_send_reply("403", "Denied, you are not allowed this route"); # }; return; } onreply_route[1] { # ----------------------------------------------------------------- # Return Route # ----------------------------------------------------------------- #replace_all("182 Queued","180 Ringing"); if ((isflagset(4) || isflagset(5)) && isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { xlog("L_NOTICE","OnReplyRoute(1) Using Mediaproxy"); use_media_proxy(); }; }; if (client_nat_test("1") && isflagset(6)) { fix_contact(); }; return; } failure_route[1] { # ----------------------------------------------------------------- # No answer, cancel, busy handling # ----------------------------------------------------------------- return; } route[13] { # ----------------------------------------------------------------- # Debuging info # ----------------------------------------------------------------- if (status=="100") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 100 Trying\n"); } else if (status=="180") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 180 Ringing\n"); } else if (status=="181") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 181 Call Is Being Forwarded\n"); } else if (status=="182") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 182 Queued\n"); } else if (status=="183") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 183 Session Progress\n"); } else if (status=="200") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 200 OK\n"); } else if (status=="202") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 202 Accepted\n"); } else if (status=="300") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 300 Multiple Choices $ct\n"); } else if (status=="301") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 301 Moved Permanently $ct\n"); } else if (status=="302") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 302 Moved Temporarily $ct\n"); } else if (status=="305") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 305 Use Proxy\n"); } else if (status=="380") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 380 Alternative Service\n"); } else if (status=="400") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 400 Bad Request\n"); } else if (status=="401") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 401 Unauthorized\n"); } else if (status=="402") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 402 Payment Required\n"); } else if (status=="403") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 403 Forbidden\n"); } else if (status=="404") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 404 Not Found\n"); } else if (status=="405") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 405 Method Not Allowed\n"); } else if (status=="406") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 406 Not Acceptable\n"); } else if (status=="407") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 407 Proxy Authentication Required\n"); } else if (status=="408") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 408 Request Timeout\n"); } else if (status=="409") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 409 Conflict\n"); } else if (status=="410") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 410 Gone\n"); } else if (status=="411") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 411 Length Required\n"); } else if (status=="412") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 412 Conditional Request Failed\n"); } else if (status=="413") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 413 Request Entity Too Large\n"); } else if (status=="414") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 414 Request-URI Too Long\n"); } else if (status=="415") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 415 Unsupported Media Type\n"); } else if (status=="416") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 416 Unsupported URI Scheme\n"); } else if (status=="420") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 420 Bad Extension\n"); } else if (status=="421") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 421 Extension Required\n"); } else if (status=="422") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 422 Session Interval Too Small\n"); } else if (status=="423") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 423 Interval Too Brief\n"); } else if (status=="429") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 429 Provide Referrer Identity\n"); } else if (status=="480") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 480 Temporarily Unavailable\n"); } else if (status=="481") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 481 Call Leg/Transaction Does Not Exist\n"); } else if (status=="482") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); } else if (status=="483") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 483 Too Many Hops\n"); } else if (status=="484") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 484 Address Incomplete\n"); } else if (status=="485") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 485 Ambiguous\n"); } else if (status=="486") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 486 Busy Here\n"); } else if (status=="487") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 487 Busy Here; Request Terminated\n"); } else if (status=="488") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 488 Not Acceptable Here\n"); } else if (status=="489") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 489 Bad Event\n"); } else if (status=="491") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 491 Request Pending\n"); } else if (status=="493") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 493 Undecipherable\n"); } else if (status=="494") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 494 Security Agreement Required\n"); } else if (status=="500") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 500 Server Internal Error\n"); } else if (status=="501") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 501 Not Implemented\n"); } else if (status=="502") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 502 Bad Gateway\n"); } else if (status=="503") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 503 Service Unavailable\n"); } else if (status=="504") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 504 Server Timeout\n"); } else if (status=="505") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 505 Version Not Supported\n"); } else if (status=="513") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 513 Message Too Large\n"); } else if (status=="580") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 580 Precondition Failure\n"); } else if (status=="600") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 600 Busy Everywhere\n"); } else if (status=="603") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 603 Decline\n"); } else if (status=="604") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 604 Does Not Exist Anywhere\n"); } else if (status=="606") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 606 Not Acceptable\n"); } else { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status not listed\n"); }; xlog("L_NOTICE", "OPENSER_NAME: $mi $mb\n"); return; }

2 1

[Serusers] jabber2foreign IM
by Allan Chao ( 策略性新技術開發處申請 ) 05 Feb '07

05 Feb '07

Dear All: I am a ser user, i want to use sip client that can commuicate with foreign IM client(msn), my flow is sip client -> ser(jabber module) (0.10 version) -> jabber server(has msn gateway) -> MSN network. but i did not see any function that can register to MSN server(just like jabber client browse transport in the jabber server and then register to MSN network) and there is no function or program using msn id and password. i doubt jabber module does not implement this part of program.Maybe there is another mathod can solve it. Does anyone can help me ? thanks. allan DISCLAIMER: Sample Disclaimer added in a VBScript.

2 1

[Users] Exec - environment variables - not found?
by Kjell Arild Frigstad 04 Feb '07

04 Feb '07

Hello! I have just upgraded from 1.0.0 til 1.1.1, and I having problem with exec_msg (the exec_msg working, but environment variabels is "not found".) It was OK in 1.0.0 version. The coding: exec_msg("echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages"); At startup (with debuggging) : 0(0) fixing exec_msg, line 160 0(0) xl_parse_format: parsing [echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages] 0(0) xl_lookup_spec_name: not found [SIP_SRCIP] 0(0) xl_parse_format: format parsed OK: [2] items Error message when running the command (with debugging) : 0(28888) xl_get_spec_value: error - null sp->itf 0(28888) exec:w_exec_msg: executing [echo SIP_SRCIP >>/var/log/messages] Error message when running in "normal" mode Feb 4 21:51:24 test /usr/local/sbin/openser[28947]: xl_get_spec_value: error - null sp->itf SIP_SRCIP As you see the text SIP_SRCIP is beeing displayd, but not the variabel. Any ide what can bee wrong? Best regards K.A.Frigstad

2 1

[Users] auth_radius extra attributes
by Cseke Tamas 03 Feb '07

03 Feb '07

Hello, can anybody tell me how can i send extra attributes to radius in authpacket, in acctpacket i can with radius_extra parameter, is there a way to use it in auth too? Thanks any help, Tamas

1 0

[Serusers] Problem with mysql.so on new server
by sip 03 Feb '07

03 Feb '07

So... I've had ser 0.9.6 running on RH EL3, and FC3 and 4. I'm building out a server on RH EL4 for SER, but I've run into a snag. I'm sure it's something simple, but it has me stumped at the moment. I've recompiled mysql.so on the new server... compiles fine (as do the rest of the modules). However, when I try to start up SER with mysql.so loaded, it gives me this: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Undefined symbol: log Never seen this before. Anyone know what it means or, more importantly, how to fix it (and no... I'm not going to scrap it all and go with Ottendorf yet, so that's not a solution ;) )? N.

4 12

[Serusers] Which locking mechanism is useful for usrloc in production environment?
by Ramu Yadav 03 Feb '07

03 Feb '07

Hello, At present we are using "sys v semaphores"(USE_SYSV_SEM) for locking. But we see a some delay while acquiring the lock to modify the usrloc's location table. We are in a thought to use "posix semaphores" (USE_POSIX_SEM), will this improve any performance for usrloc location table. Thoughts and Ideas will be appreciated. -- Ramu Yadav

4 5

[Users] NEW FEATURE: better timers
by Bogdan-Andrei Iancu 02 Feb '07

02 Feb '07

Hi everybody, following to some discussions / requests from users, openser devel branch has now core support for timers will resolution less than a second. It uses time functions that can give a microsecond accuracy, but for performance reasons, the utimer (microtimer) uses a ticking period of 100 milliseconds. Of course, this can be changed at compile time. First "beneficiary" of this timer, is TM, to gain more accuracy for retransmissions: * retransmission timers use the the core utimer (with microsecond resolution), having 100 milliseconds intervals. * this will increase the retransmissions precision to an error of maximum 100 milliseconds. * T1 (first retransmission) is done at 500 milliseconds (instead of 1 sec), as the RFC3261 suggests. * T1 and T2 (see RFC3261) can be configured via module parameters - the inter-steps are automatically computed on startup based on the RFC3261 specifications. Along with increasing the timer accuracy, the TM timer lists were improved for faster operations: * faster timers lists - use shortcut pointers (based on timeout values) for faster walking through the list -> faster insert and expire * for elements to be deleted from the timer lists, do not overwrite the timeout value as this will trigger more processing when inserting new elements into the list The performance tests are not completed - I need more machines to stress the proxy :D -, but I will come back with results during the testing phase. regards, bogdan

1 0

[Users] removed avp_write(); avp_aliases moved in core
by Daniel-Constantin Mierla 02 Feb '07

02 Feb '07

Hello, two major changes were applied to avpops module in development branch (1.2.0): - avp_write() has been removed - same functionality is provided by assignment operator in routing blocks, together with latest commit of PV transformations - avp_aliases parameter has been removed from avpops module and introduced as core parameter. you can use it directly in core, in global parameters section in form of: avp_aliases="uuid=I:660;email=s:email_addr;fwd=i:753" The value is the same as it was for avpops parameter. Now the AVP aliases can be used directly in routing blocks. Cheers, Daniel

1 0

[Users] new feature: pseudo-variable transformations
by Daniel-Constantin Mierla 02 Feb '07

02 Feb '07

Hello, currently in CVS trunk of development version is a new feature 'transformations'. A transformations is basically a function that can be applied to a pseudo-variable (PV). In this way, the value of PV is not altered but the result is something different. They are intended to facilitate access to different attributes of PV (like strlen of value, parts of value, substrings) or complete different value of PV (encoded in hexa, md5 value, escape/unescape PV value for DB operations...). The format is '{id[,parameters]}' and they can be used inside parenthesis of pseudo-variables (note that even short name pseudo-variables must be enclosed in '(' and ')' to be able to apply transformations) For example: - $(ru{s.len}) - returns the lenth of R-URI - $(avp(i:3){s.md5}) - returns the md5 string over value of $avp(i:3) The following classes of transformations are implemented 1) string - the name of transformation starts with 's.'. Available transformations in this class: - {s.len} - returns strlen of PV value - {s.md5} - returs md5 over PV value - {s.substr,offset,length} - returns substring starting at offset having size of 'length'. If offset is negative, then it is counted from the end of PV value, -1 being the last char. In case of positive value, 0 is first char. Length must be positive, in case of 0, substring to the end of PV value is returned. offset and length can be PV as well. Example: "abcd"{s.substr,1,0} = "bcd" - {s.select,index,separator} - returns a field from PV value. The field is selected based on separator and index. The separator must be a character used to identify the fields. Index must be a integer value or a PV. If index is negative, the count of fields starts from end of PV value, -1 being last field. If index is positive, 0 is the first field. Example: "12,34,56"{s.select,1,,} = "34" ; "12,34,56"{s.select,-2,,} = "12" - {s.encode.hexa} - returns encoding in hexa of PV value - {s.decode.hexa} - returns decoding from hexa of PV value - {s.escape.common} - returns escaped string of PV value. Characters escaped are ''', '"', '\' and 0. Useful when doing DB queries (care should be taken for non latin character set) - {s.unescape.common} - returns unescaped string of PV value. Reverse of above transformation. 2) uri - the name of transformation starts with 'uri.'. The PV value is considered to be a SIP URI. This transformation returns parts of SIP URI (see struct sip_uri). Available transformations in this class: - {uri.user} - returns the user part - {uri.host} or {uri.domain} - returns the domain part - {uri.passwd} - returns the password - {uri.port} - returns the port - {uri.params} - returns the URI parameters in a string - {uri.param,name} - returns the value of parameter with name 'name' - {uri.headers} - returns URI headers - {uri.transport} - returns the value of transport parameter - {uri.ttl} - returns the value of ttl parameter - {uri.uparam} - returns the value of user parameter - {uri.maddr} - returns the value of maddr parameter - {uri.method} - returns the value of method parameter - {uri.lr} - returns the value of lr parameter - {uri.r2} - returns the value of r2 parameter 3) parameters list - the name of the transformation starts with 'param.'. The PV value is considered to be a string like "name1=value1;name2=value2;...". The transformations returns the value for a specific parameter, or the name of a parameter at a specific index. Available transformations in this class: - {param.value,name} - returns the value of parameter 'name' Example: "a=1;b=2;c=3"{param.value,c} = "3" - {param.name,index} - returns the name of parameter at position 'index'. Example: "a=1;b=2;c=3"{param.name,1} = "b" Within a PV, many transformation can be applied, being executed from left to right. Example: $var(x) = "a=1;b=22;c=333"; $(var(x){param.value,$(var(x){param.name,1})}{s.len}) = 2 - the length of the value of parameter at position 1 (remember 0 is first position, 1 is second position) The transformations can be used anywhere, being considered parts of PV -- in xlog, avpops or other modules' functions and parameters, in right side assignment expressions or in comparisons. Documentation page for transformations was opened at: http://openser.org/dokuwiki/doku.php/transformations:devel Feedback is welcome! Cheers, Daniel

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[Serusers] meeting (was: Re: [Serdev] SER's core design features(process model/parser/lumps/script) - was: So who/what is SER for, anyway?
by Jiri Kuthan 02 Feb '07

02 Feb '07

We could use the IETF meeting in Prague in March. Actually, I have anticipated that and there is a hotel room reserved already :-) Could folks willing to show up and debate roadmap or whatever else appears important technically speak up now? -jiri At 12:08 25/01/2007, Olle E Johansson wrote: >As I am fighting the same issues and try to make chan_sip easier for >the administrator, would it be >a good idea with a joint Open Source SIP developer meeting somewhere >in Europe this spring? > >We have a lot of common issues, need to settle with the same >terminology and propably fight >the same fights with RFCs to implement... > >A lot of of users use both Asterisk and SER/OPENSER, so they need >interoperability. > >For my work on a new chan_sip, please check http://www.codename- pineapple.org > >/O >_______________________________________________ >Serdev mailing list >Serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] www_challenge() error in auth module
by raj gopal 02 Feb '07

02 Feb '07

Hi, I can not use www_challenge(realm,qop)function in ser.cfg file even though auth module is loaded. The readme file has documentation for it. When I use the following in config file and start SER, the following error is thrown if (!radius_www_authorize("")) { www_challenge("", "1"); break; }; parse error (197,49-50): unknown command, missing loadmodule? Can any one help me to find out whats wrong with this module/function. -Venkat

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[Serusers] automated response
by Patrick Timmons 02 Feb '07

02 Feb '07

IMPORTANT NOTE: if you are receiving this auto responder after accessing Acme Packet's web portal or sending an email to a mailing list, rest assured that your message has been received and will be acted upon by the appropriate personnel. I am taking Friday, February 2 through Tuesday, February 6 off and will be mostly unavailable during this time. I will not have access to a computer, but will be reachable via my cell phone in the event of emergencies. For support issues, please contact support(a)acmepacket.com. For non-support issues, please contact Erin Medeiros, Vice President of Professional Services, at emedeiros(a)acmepacket.com. Thank you, Patrick Timmons Vice President, Systems Engineering Acme Packet, Inc.

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[Serusers] ser 0.10.99 - call forwarding issue
by tzieleniewski 01 Feb '07

01 Feb '07

hi! I am trying to set up the call forwarding feature and I have some question bout the idea how to realize this. My Idea was to do it with the usage of avp's. First thing I do is the check if there is a call_forward parameter set in the database in user_attrs table. I do this by invoking load_attrs("$fu", "$f.uid"); and then I do the check if ($tu.call_forward) On the "what's new site" in the avps and selects article I found another method which performs similar task: lookup_user("$tu","@ruri") What is the difference between those to methods?? Are they part of the SER core? previously in ser 0.9.x I have it done by, first loading the avp by avp_db_load() and then in case of call forwarding by making avp_pushto("$ruri","something") or rewriteuri() and then t_relay(). What is now a preferred way to realized similar functions. I mean can I utilize those methods like load_attrs or lookup_user() and then substitute the ruri without avp_pushto or rewriteuri?? best tomasz

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Re: [Users] Acc'ing + Flatstore, Log Format?
by Bogdan-Andrei Iancu 01 Feb '07

01 Feb '07

Brandon, the log_fmt parameter affects only the syslog accounting (logging via syslog to log files). Flatstore is implemented as Write-Only DB engine, so it be used by the DB part of the accounting module. You need to play with the "db_*" related parameters from the ACC module. regards, bogdan Brandon Armstead wrote: > Bogdan-Andrei Iancu wrote: > >> Hi Brandon, >> >> what version of OpenSER are you using? There were important changes >> in the latest version, so it is important to know it. >> >> regards, >> bogdan >> >> Brandon Armstead wrote: >> >>> Hello everyone, >>> >>> Just a quick question, I have the following in my configuration: >>> modparam("acc", "log_fmt", "miocrdfs01"), however, in the actual >>> logging it is giving me back 14 different log "information", yet the >>> format "miocrdfs01" is only 10, does flatstore use a different >>> formatting mechanism, or does it force it to be that 14, and only >>> that 14, or anyone know whats going on? Thanks >>> >>> Regards, >>> Brandon >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >> > Hello Bodgan, > > I'm currently using OpenSER version 1.1.1-notls. Thanks again! > > Regards, > Brandon >

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[Serusers] src_ip=="%$foo"
by Atle Samuelsen 01 Feb '07

01 Feb '07

Hi guys and girls, is it possible to do a if(src_ip=="%$foo") where $foo contains more then one IP address? - Atle

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[Serusers] Presence client error
by Jens-Martin Groenne 01 Feb '07

01 Feb '07

I'm testing the presence module on 0.10.99-dev in CVS. When using the newest X-lite client I am getting the following error messages in /var/log/messages /usr/local/sbin/ser[30891]: INFO: rls_mod.c:155: reading RLS data from database /usr/local/sbin/ser[30879]: INFO: subscribe.c:286: Unsupported event package /usr/local/sbin/ser[30879]: ERROR: subscribe.c:590: Error while checking message /usr/local/sbin/ser[30879]: INFO: subscribe.c:603: handle_subscription about to send_reply and return -2 /usr/local/sbin/ser[30885]: Warning: run_failure_handlers: no UAC support (0, 256) /usr/local/sbin/ser[30885]: WARN: rl_subscription.c:398: destroying subscription from callback due to 420 response on NOTIFY /usr/local/sbin/ser[30885]: TRACE: rl_subscription.c:400: RLS: subscription destroyed!!! In X-lite I get this error message: [07-01-30]15:44:45.526 | Debug | RESIP:TRANSACTION | "Match!" | [07-01-30]15:44:45.526 | Debug | RESIP:TRANSACTION | "Send to TU: TU: DialogUsageManager size=0 NOTIFY sip:tuser@192.168.1.35:34492 SIP/2.0 Via: SIP/2.0/UDP 85.112.135.70;branch=z9hG4bK7ed6.d015cec1.0 Max-Forwards: 70 Require: eventlist Contact: <sip:85.112.135.70:5060> To: ""Jens""<sip:tuser@test-domain.com>;tag=d204727a From: <sip:tuser@test-domain.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-1a1f Call-ID: OTYwMDE1NmQ5MDQwYjFmMTBlZjQ0NTNhZTExNWM4MGM. CSeq: 10 NOTIFY Content-Type: multipart/related;type=""application/rlmi+xml"";start=""qwW1099041416pPdxX1162343142"";boundary=""RewXdpxR741013657xA447338541""; User-Agent: Sip EXpress router(0.10.99-dev66 (i386/linux)) Subscription-State: active;expires=300 Event: presence Content-Length: 338 --RewXdpxR741013657xA447338541 Content-Transfer-Encoding: binary Content-ID: qwW1099041416pPdxX1162343142 Content-Type: application/rlmi+xml;charset=""UTF-8"" <?xml version=""1.0""?> <list xmlns=""urn:ietf:params:xml:ns:rlmi"" uri=""sip:tuser@test-domain.com"" version=""0"" fullState=""true""> </list> --RewXdpxR741013657xA447338541-- " | [07-01-30]15:44:45.526 | Debug | RESIP | "SipMessage::getContents: multipart/related" | [07-01-30]15:44:45.526 | Debug | RESIP:DUM | "Decrypted message:02815028 " | [07-01-30]15:44:45.526 | Info | RESIP:DUM | "Got: SipReq: NOTIFY tuser@192.168.1.35:34492 tid=7ed6.d015cec1.0 cseq=NOTIFY contact=85.112.135.70:5060 / 10 from(wire)" | [07-01-30]15:44:45.526 | Info | RESIP:DUM | "Received an unsupported option tag(s): SipReq: NOTIFY tuser@192.168.1.35:34492 tid=7ed6.d015cec1.0 cseq=NOTIFY contact=85.112.135.70:5060 / 10 from(wire)" | [07-01-30]15:44:45.526 | Debug | RESIP | "Helper::makeResponse(SipReq: NOTIFY tuser@192.168.1.35:34492 tid=7ed6.d015cec1.0 cseq=NOTIFY contact=85.112.135.70:5060 / 10 from(wire) code=420 reason=" | [07-01-30]15:44:45.526 | Debug | RESIP | "SEND: SipResp: 420 tid=7ed6.d015cec1.0 cseq=NOTIFY / 10 from(tu)" | [07-01-30]15:44:45.527 | Debug | RESIP:DUM | "Failed required options validation NOTIFY sip:tuser@192.168.1.35:34492 SIP/2.0 Via: SIP/2.0/UDP 85.112.135.70;branch=z9hG4bK7ed6.d015cec1.0 Max-Forwards: 70 Require: eventlist Contact: <sip:85.112.135.70:5060> To: ""Jens""<sip:tuser@test-domain.com>;tag=d204727a From: <sip:tuser@test-domain.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-1a1f Call-ID: OTYwMDE1NmQ5MDQwYjFmMTBlZjQ0NTNhZTExNWM4MGM. CSeq: 10 NOTIFY Content-Type: multipart/related;type=""application/rlmi+xml"";boundary=""RewXdpxR741013657xA447338541"";start=""qwW1099041416pPdxX1162343142"" User-Agent: Sip EXpress router(0.10.99-dev66 (i386/linux)) Subscription-State: active;expires=300 Event: presence Content-Length: 334 --RewXdpxR741013657xA447338541 Content-Type: application/rlmi+xml;charset=""UTF-8"" Content-Transfer-Encoding: binary Content-Id: qwW1099041416pPdxX1162343142 <?xml version=""1.0""?> <list xmlns=""urn:ietf:params:xml:ns:rlmi"" uri=""sip:tuser@test-domain.com"" version=""0"" fullState=""true""> </list> - Does any know know how to solve this problem? Is there any other clients that works well with the ser presence module? Kind regards, Jens-Martin Groenne

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