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[Serusers] Call transfer with 7960s
by Steve Blair 17 Feb '05

17 Feb '05

Hello: I have a Cisco 7960 running v7.3 SIP image. The phone registers to SER 0.8.14 stable. Everything works fine except call transfer. Has anyone experienced problems with call transfer? Has anyone done anything specific in their ser config to handle call transfer? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] av_print() not working
by Java Rockx 17 Feb '05

17 Feb '05

Hi All. I'm using ser-0.9. What do I have to do to get the avpops avp_print() function to spit out something? Right now I'm just calling it after avp_write() Regards, Paul

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[Serusers] nathelper ERROR: extract_mediaip: no `c=' in SDP
by Charles Wang 17 Feb '05

17 Feb '05

Thanks Marian's help and answer. I can make a call from domainA to domainB via PDT module. But I got another error message from nathelper or rtpproxy. It is "ERROR: extract_mediaip: no `c=' in SDP" from /var/log/messages. I also find this message at source code of nathelper. So I think it should happen from this module. It seems voice channel can't pass throught it. I make a call from UA_A ( under NAT ) from domainA and using PDT to redirect to UA_B via domainB. UA_B has ring and it can answer. I can hear the UA_A's sound from UA_B. The sound of UA_B is failed to UA_A. UA_A(NAT) ==> domainA(PDT) ==> domainB(nathelper+rptproxy) ==> UA_B(NAT) SOUND_A ==========================================> UA_B UA_A <===== XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX === SOUND_B Do I put the prefix2domain() in a wrong place?? Or Is it a bug ofi nathelper & rtpproxy? And my ser.cfg as below: # # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=8 # 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 */ listen=xxx.xxx.xxx.xxx alias=ser.xxx.net alias=ser alias=xxx.xxx.xxx.xxx check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/pdt.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_mode", 1) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # -- db_url params -- modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|pdt", "db_url", "mysql://ser:heslo@localhost/ser") # -- use_domain params -- modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0) # -- permissions params -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- accounting params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) # -- domain params -- modparam("domain", "db_mode", 1) # -- avp params -- modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") #modparam("avpops", "use_domain", "1") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") # To use more than one tables example #modparam("avpops", "db_scheme", "scheme1:table=subscriber;uuid_column=uuid;value_column=first_name") # -- tm params -- modparam("tm", "fr_timer", 15) modparam("tm", "fr_inv_timer", 22) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # -- pft params -- modparam("pdt", "db_table", "prefix_domain") modparam("pdt", "prefix", "") #modparam("pdt", "start_range", 2000) #modparam("pdt", "terminator", 0) modparam("pdt", "hsize_2pow", 2) modparam("pdt", "sync_time", 300) modparam("pdt", "clean_time", 600) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE" || method=="BYE") { setflag(1); }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("SER: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; prefix2domain(); if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("ser.xxx.net", "subscriber")) { www_challenge("ser.xxx.net", "0"); break; }; save("location"); break; }; - lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } On Thu, 17 Feb 2005 12:49:10 +0100, Marian Dumitru <marian.dumitru(a)voice-sistem.ro> wrote: > Hi Charles, > > You have to use interdomain prefixes. Set on domainA a prefix for > recognizing dialed numbers in domainB. > The logic is like: > PrefixToB+numberInB@domainA -> numberinB@domainB > > you can do this directly from script or by using PDT module. > > Best regards, > Marian

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Re: [Serusers] Call transfer with 7960s
by Klaus Darilion 17 Feb '05

17 Feb '05

Capture the SIP packets from/to the gateway and take a look at the SDP. I Don't know if Cisco can handle this scenario (sending RTP to itself) proper. regards, klaus Steve Blair wrote: > > Thanks I guess but the problem is that if a call comes in from the PSTN > to an IP phone via a Cisco 2620 gateway and is then transferred back > through > the same gateway to a Centrex extension the call connects but no media > is exchanged. > > > > Klaus Darilion wrote: > >> Hi Steve! >> >> You don'T have to do anything special for call transfer. Call >> transfer is end2end feature - implementied by the SIP clients. The >> REFER message will be forwarded by the SIP proxy as any other message. >> >> regards, >> klaus >> >> >> >> Steve Blair wrote: >> >>> >>> Hello: >>> >>> I have a Cisco 7960 running v7.3 SIP image. The phone registers to SER >>> 0.8.14 stable. Everything works fine except call transfer. Has anyone >>> experienced problems with call transfer? Has anyone done anything >>> specific in their ser config to handle call transfer? >>> >>> Thanks,Steve >>> >

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[Serusers] Selective failure_route handling
by Evan Borgstrom 17 Feb '05

17 Feb '05

Has anyone found a way to selectively respond to failures using failure_route's? For instance I want to only run the actions in the failure_route if the code that triggered the failure route is certain 400 or 500 error codes. Since the sip msg passed into the failure_route is the initial message that triggered the failure (usually the INVITE) using methods from the textops module don't work. Thanks, Evan

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[Serusers] Can't run SER with radius
by Roman 17 Feb '05

17 Feb '05

Anybody please help with subj 0.8.14 binaries from iptel.org ser -dddddddd says: ERROR: bad config file (4 errors) config as in the manual: # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5065 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- auth params -- modparam("auth_radius", "radius_config","/etc/radiusclient/radiusclient.conf") modparam("auth_radius", "service_type", 15) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #modparam("acc", "log_level", 1) #modparam("acc", "radius_flag", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "1"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }

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[Serusers] Serusers list
by Kenichi Satoh 17 Feb '05

17 Feb '05

Sir, Please entry my email address to your mailing list. My address is as below. ksatoh(a)asgent.co.jp Thanks and Regards, Ken Sato ------------------------------------ Ken Sato MoIP Business Asgent, Inc. 19-7 Nihonbashi-Koamicho Chuo-ku, Tokyo 103-0016, Japan TEL: +81-3-5643-2561 FAX: +81-3-5643-2571 http://www.asgent.co.jp/ ------------------------------------

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[Serusers] DTMF / asterisk / ser
by Patrick Baker 17 Feb '05

17 Feb '05

Hello all, Im having trouble getting DTMF working on my sip client phone - asterisk doesnt receive any DTMF tones - i've even tried inband. setup; sip phone --> SER --> asterisk Thanks, Patrick

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[Serusers] serweb and fifo [setting serweb on remote server]
by Iqbal 17 Feb '05

17 Feb '05

Hi I have been toying with serweb, however in order for it to scale, I woulkd need it to be installed on a different server than SER itself. Looking at the code, it seems that it needs to talk to ser via ser_fifo (is that correct ?), if so I obviously cant get this to happen over a remote connection unless I go and share drives...which cant happen. I looked into the ser_fifo in ser, and it seems that you can run ser_fifo out of a DB, has anyone done this, and if so, is it possible to link serweb to this DB also, rather than the file. Iqbal

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[Serusers] Goofy question about mutiple contact records and CANCEL messages
by Java Rockx 17 Feb '05

17 Feb '05

Hi All. I'm using ser-0.9. When someone calls a subscriber who has multiple records in the location table SER sends out an INVITE message to each of the contact locations. But when the caller CANCELs the call, SER still seems to send INVITE retransmissions to some of the contact locations. Can this be stopped? Regards, Paul

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RE: [Serusers] SER 0.9.0 adding users?
by Jiri Kuthan 17 Feb '05

17 Feb '05

At 03:45 PM 2/16/2005, Matt Schulte wrote: >I still get errors, our sip provider requires a + anyway :-) > >I seem to get errors no matter what commands I run. Here's the latest >trying to add a "domain". > >./serctl domain add blah.blah.net > >500 command 'domain_reload' not available >Feb 16 09:43:36 brak /usr/local/sbin/ser[4315]: ERROR: fifo_server: >command domain_reload is not available That's easy. If you wish to use multidomain processing, you need to load and use the domain module. You can't access a module via fifo if the module is not used. >Is there something that has to be done with the fifo_server before doing >all this? > >ser *is* running > >[serctl] >SER_FIFO=/tmp/ser_fifo > >[ser.cfg] >fifo="/tmp/ser_fifo" > > > >-----Original Message----- >From: Felipe Martins [mailto:fmartins@mundivox.com] >Sent: Wednesday, February 16, 2005 8:31 AM >To: serusers(a)lists.iptel.org >Subject: Re: [Serusers] SER 0.9.0 adding users? > > >Try > ># serctl add 18165551212 blahblah blah@blah > > > >On Wed, 16 Feb 2005 07:06:13 -0600 >"Matt Schulte" <mschulte(a)netlogic.net> wrote: > >> I get this when adding a user in ser (using serctl) >> >> root@brak sbin]# ./serctl add +18165551212 blahblah blah@blah MySql >> password: >> error: 400; check if you use aliases in SER >> >> Um error 400?? I'm lost. no docs, frustrated. venting. That's easy too. Read the manual to avoid frustration. http://www.iptel.org/ser/doc/seruser/seruser.html#AEN1087 -jiri >> >> Matt >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> > > >-- > Felipe Martins > TEP Solution & New Technologies > Mundivox Communications > fmartins(a)mundivox.com > > Site: www.mundivox.com > Tel.: +55 +21 +3820 8839 > Cel.: +55 +21 +9823 8602 > Fax.: +55 +21 +3820 8844 > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] only INVITE accounted
by Rosario Pingaro 17 Feb '05

17 Feb '05

I configured ser to work with an pstn gw accounting all call to mysql db. But I when I call a local user I get accounted the INVITE and the BYE messages, but when I call an outging uri I get accounted only BYE messages and not the INVTE. Where is the probelm? THANKS below there is may ser.cfg file: ______________________________________________________________ # # Italycom SER server ver 0.052 # # modificato il 16-02-2005 # # con nathelper e rtpproxy funzioannte # da perfezionare l'accounting # # # ----------- global configuration parameters ------------------------ /* Uncomment these lines to enter debugging mode debug=9 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 listen=213.178.216.44 # ------------------ 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # # ----------------- setting module-specific parameters --------------- # # # -- usrloc params -- # modparam("usrloc", "db_url", "mysql://user:password@domain.us/ser") modparam("usrloc", "db_mode", 1) # # # -- auth params -- # modparam("auth_db", "db_url", "mysql://user:password@domain.us/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # # # -- nathelper params -- # modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # # # -- acc params -- # modparam("acc", "db_url", "mysql://ser:heslo@ns2.italycomnet.it/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "report_ack", 0) # # # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- tm params -- modparam("tm", "fr_timer", 20) modparam("tm", "fr_inv_timer", 40) modparam("tm", "wt_timer", 5) # # # # # ------------------------- request routing logic ------------------- # main routing logic route{ if(nat_uac_test("3")) { if((method == "REGISTER") || !(search("^Record-Route:"))) { log("LOG:Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); if(method == "INVITE") { fix_nated_sdp("1"); }; force_rport(); setflag(6); }; }; # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # # Accounting if (method=="INVITE" || method=="BYE") { setflag(1); }; # # # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { # mark routing logic in request t_relay(); append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("voip.mydomain.it", "subscriber")) { www_challenge("voip.mydomain.it", "0"); break; }; save("location"); break; }; # lookup("aliases") ; # # if (!lookup("location")) { if (method=="INVITE") { if ( (uri=~"sip:[0-9]{5,20}@.*") ) { rewritehostport( "pstngw.com:5060" ); forward( pstngw.com, 5060 ); break; }; sl_send_reply("404", "Not Found"); break; } else { # forward to user's current destination t_relay(); break; }; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } # route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { if (isflagset(6) && status =~ "(180)|(183)|2[0-9][0-9]") { # fix_nated_contact(); if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); }; # force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

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[Serusers] How to use a numeric ID to call another UA at another server?
by Charles Wang 17 Feb '05

17 Feb '05

Dear ALL: My scenario is two domains(domainA, domainB), two SER sip server (SER_A, SER_B) and two UAs (UA_A, UA_B). If UA_A registers to SER_A using a numberic ID (886701234567) and another UA_B registers to SER_B using a numberic ID (886207654321). How can SER_A know to forward to call to UA_B under SER_B with only numberic ID? Is it possible? If it is impossible, now SER_A and SER_B using the same DB, does this question have answer? How can SER_A search the DB and find the UA_B's URI or 886207654321(a)domainB.com with only 886207654321? Can someone give me a ser.cfg to reference? Please.... Charles

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Re: [Serusers] Carrier-grade framework for SER
by Andreas Granig 17 Feb '05

17 Feb '05

Hi Greger, > Any conclusions or progress you have made after the discussion on the > mailing list? I learned a few things :-) Tried to summarize my > understanding of the different issues and how I understood the > "recommended" approach, but didn't see any response from you. There was a lot that had to be done in the meanwhile, sorry ;o) But here's my current (not perfect but perfectly working) solution: Locations are replicated on SIP-layer using forward_tcp(...) (thanks to Juha for the hint) on one side and save_noreply(...) on the other side. This works very reliable with three SERs although it wouldn't scale much, so this has to be improved in the future. The config looks like: if(method == "REGISTER") { if(src_ip==<IP of peer1> || src_ip==<IP of peer2>) { save_noreply("location"); } else { if(!www_authorize("<my domain>", "subscriber")) { www_challenge("<my domain>", "0"); break; } if(!check_to()) { sl_send_reply("403", "Use To=username next time"); break; } if(!save("location")) { sl_reply_error(); break; } forward_tcp("<IP of peer1", 5060); forward_tcp("<IP of peer2", 5060); } break; } The usrloc modules operates in write-through-mode so locations are always synched to the db. When a SER goes down, I've a script for recovering the location tables, and it looks like: mysqldump -h${SER_PEER} -u${DB_RUSER} -p${DB_RPASS} -Q \ --extended-insert --add-drop-table ser ${SER_LOCATIONS} \ | mysql -u${DB_LUSER} -p${DB_LPASS} ser /usr/local/ser/sbin/ser.sh start where $SER_LOCATIONS holds the location tables like "location", "aliases" and so on. What still has to be done is replicating the nonce value so UAs with DNS-SRV implementations which change the server on each request and SERs behind load balancers will work too. It's not as nifty as using diameter or something like that but it grants me some reprieve ;o) For distributing FIFO commands to provision call forwards and read SER's status etc I wrote an XML-RPC server which communicates with SER's unixsocket-FIFO-interface. So a simple PHP script communicating with a SER looks like this: $url = "http://my.server:5080/xmlrpc"; $method = "fifo.execute"; $fifocmd = "ul_show_contact"; $fifoparams = array("location", "012345"); $ar = array("cmdname"=>$fifocmd,"cmdparams"=>$fifoparams); $request = xmlrpc_encode_request($method, $ar); $ch = curl_init(); curl_setopt($ch, CURLOPT_URL, $url); curl_setopt($ch, CURLOPT_RETURNTRANSFER,1); curl_setopt($ch, CURLOPT_POST, 1); curl_setopt($ch, CURLOPT_POSTFIELDS, $request); $response = curl_exec($ch); $xmlar = xmlrpc_decode($response); if(xmlrpc_is_fault($xmlar)) { /* fault processing, print the error */ } else { /* normal processing, print the response parameters etc */ } The XML-RPC solution seems to be more flexible to me than the TCP-interface for SER because it can not only execute FIFO commands but can easily be extended to communicate with the mediaproxy for example. But this opinion may depend on one's needs. The licensing question of the XML-RPC server is still not clear (chiefs, you know), but I hope I get it licensed as GPL. Cheers, Andy

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Re: [Serusers] Why is it empty at my MYSQL's location table?
by Cesc Santasusana 17 Feb '05

17 Feb '05

Try changing usrloc::db_mode to 1 ... write-through. modparam("usrloc", "db_mode", 1) For the domain thing ... change use_domain to 1. Cesc >>> Charles Wang <lazy.charles(a)gmail.com> 02/17/05 10:00AM >>> Dear ALL: I add a user such as "serctl add 1011 1011 aaa(a)sip.xxx.net" (format: serctl add userid passwd email) It will insert one record to my MYSQL's "subscriber" table. But I can not find any information in my another table "aliases". Because when I wanna look the missed calls from serweb, it always UNION two tables missed_calls and aliases. So, I think, it should be some records at this aliases table. I add records to aliases table manually with command: "serctl alias add 1011 aaa(a)sip.xxx.net" but the field "domain" is still empty. So I login to mysql and "update aliases set domain='sip.xxx.net'; " I finally see the missed calls information. unclassified It is not a good idea to create a user with such more steps(add user, add alias, change domain). Is it my false in my ser.cfg or my config.php at serweb? There is no record at my MYSQL's location table. How can I make it not empty via correct cfg setting? My ser.cfg is as below: # # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=8 # 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 */ listen=xxx.xxx.xxx.xxx alias=ser.xxx.net alias=ser alias=xxx.xxx.xxx.xxx check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # -- db_url params -- modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") # -- use_domain params -- modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0) # -- permissions params -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- accounting params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) # -- avp params -- modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") #modparam("avpops", "use_domain", "1") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") # To use more than one tables example #modparam("avpops", "db_scheme", "scheme1:table=subscriber;uuid_column=uuid;value_column=first_name") # -- tm params -- modparam("tm", "fr_timer", 15) modparam("tm", "fr_inv_timer", 22) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE" || method=="BYE") { setflag(1); }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("SER: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("ser.xxx.net", "subscriber")) { www_challenge("ser.xxx.net", "0"); break; }; save("location"); break; }; if (method=="INVITE" && uri =~ "^sip:0[0-9]*@") { append_hf("P-hint: PSTN\r\n"); rewritehostport("xxx.xxx.xxx.xxx:5060"); #t_relay_to_udp("xxx.xxx.xxx.xxx","5060"); #t_relay_to_tcp("xxx.xxx.xxx.xxx","5060"); log(1, "SER: matched ^sip:[^0][0-9]*@ - process in route(1)\n"); route(1); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Why is it empty at my MYSQL's location table?
by Charles Wang 17 Feb '05

17 Feb '05

Dear ALL: I add a user such as "serctl add 1011 1011 aaa(a)sip.xxx.net" (format: serctl add userid passwd email) It will insert one record to my MYSQL's "subscriber" table. But I can not find any information in my another table "aliases". Because when I wanna look the missed calls from serweb, it always UNION two tables missed_calls and aliases. So, I think, it should be some records at this aliases table. I add records to aliases table manually with command: "serctl alias add 1011 aaa(a)sip.xxx.net" but the field "domain" is still empty. So I login to mysql and "update aliases set domain='sip.xxx.net'; " I finally see the missed calls information. It is not a good idea to create a user with such more steps(add user, add alias, change domain). Is it my false in my ser.cfg or my config.php at serweb? There is no record at my MYSQL's location table. How can I make it not empty via correct cfg setting? My ser.cfg is as below: # # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=8 # 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 */ listen=xxx.xxx.xxx.xxx alias=ser.xxx.net alias=ser alias=xxx.xxx.xxx.xxx check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # -- db_url params -- modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") # -- use_domain params -- modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0) # -- permissions params -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- accounting params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) # -- avp params -- modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") #modparam("avpops", "use_domain", "1") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") # To use more than one tables example #modparam("avpops", "db_scheme", "scheme1:table=subscriber;uuid_column=uuid;value_column=first_name") # -- tm params -- modparam("tm", "fr_timer", 15) modparam("tm", "fr_inv_timer", 22) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE" || method=="BYE") { setflag(1); }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("SER: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("ser.xxx.net", "subscriber")) { www_challenge("ser.xxx.net", "0"); break; }; save("location"); break; }; if (method=="INVITE" && uri =~ "^sip:0[0-9]*@") { append_hf("P-hint: PSTN\r\n"); rewritehostport("xxx.xxx.xxx.xxx:5060"); #t_relay_to_udp("xxx.xxx.xxx.xxx","5060"); #t_relay_to_tcp("xxx.xxx.xxx.xxx","5060"); log(1, "SER: matched ^sip:[^0][0-9]*@ - process in route(1)\n"); route(1); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

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[Serusers] avp_radius module
by Martin Koenig 17 Feb '05

17 Feb '05

Hello, is it possible to add more than one response for one user in the radius users database? Will the module process this correctly? Is there a list of all known AVPs available somewhere? Regards, Martin Koenig

3 4

[Serusers] nathelper patch
by support 17 Feb '05

17 Feb '05

Hi, I have found the following message from the mailing list. I also saw your question on RFC compliance and the Sonus equipment: In order to make Grandstream phones register properly when using STUN behind symmetric NAT, I had to patch nathelper with the rport != port of received address check. (I use 0.8.14 and I guess you already have the patch with the development version). The reason is that Grandstream attempts to rewrite the address using STUN even though it correctly detects a symmetric NAT. I have seen that this was introduced in a new firmware not long ago (release notes). This pussles me as sources I have seen claims this to beinvalid behavior (which seems correct to me). It seems that I am facing the same problem now. Is there any solution to us? Where could I download the nathelper patch as described above? Thank you. Best Regards, Thomas

2 1

[Serusers] mediaproxy/rtpproxy
by Nhadie 17 Feb '05

17 Feb '05

Hi All, Can anyone help me or kindly send me a sample part of configuration wherein I will automatically use mediaproxy/rtpproxy for calls coming from a NATted UA, going to either PSTN or a local extension on my SIP server, without setting the outbound proxy on the UA's. Just like a transparent proxy in squid(sorry if i compared it to squid). TIA Regards, Nhadie

2 1

[Serusers] Question abount avpops's avp_db_load()
by Charles Wang 17 Feb '05

17 Feb '05

Dear ALL: I have defined these lines at my ser.cfg. modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") Q1: But I don't know what it mean about 700 or 701. Is it a numeric ID of one UA? And I have refered to http://www.voice-system.ro/docs/avpops And find many examples in it. But I still can't understand what it mean about avp_db_load(). my avp table is named as "usr_preferences" and it is empty before I use avpops modules. Q2: Is it should have some initilized records in it? Otherwise, what will be loaded with avp_db_load()? For this web page said: avp_db_load("$ruri/domain","i:/domain_preferences"); - loads all AVPs with ID from 'domain_preferences' table for domain from RURI Q3: Is "$ruri" the caller user's URI? and the "domain" is the same? Is "domain_preferences" table name? Is it equal to my "usr_preferences"? Or it is just a virtual table in memory? Thank you for your answer my stupid questions. Charles

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[Serusers] STUN, rtpproxy problem
by support 17 Feb '05

17 Feb '05

Please ignore my previous email. Here is the correct one. Thank you. ----- Original Message ----- From: support To: serusers(a)lists.iptel.org Sent: Thursday, February 17, 2005 11:59 AM Subject: STUN, rtpproxy problem Hi, I am running ser-0.8.14 with rtpproxy and nathelper. I have also enabled STUN on the same server. My SIP UA supports STUN, but my PSTN gateway does not. If I don't enable STUN on SIP UA, I can route the call to PSTN gateway successfully using rtpproxy. If I enable STUN on SIP UA, I cannot route the call out to PSTN gateway. On ringing sound is heard on callie, but no voice passing through. I use t_relay_to_udp to forward the call to PSTN gateway. It seems that if STUN is supported on SIP UA, the call cannot be routed to PSTN gateway. Hope someone could fix this problem. Best wishes, Thomas My ser.cfg is as follows: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen="" port=5060 #children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- Nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # Nathelper if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (uri=~"^sip:866*") { log(1, "going to PSTN route\n"); route(2); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); break; }; } route[2] { force_rtp_proxy(); t_on_reply("1"); t_relay_to_udp("T1 gateway IP","T1 Gateway UDP port"); } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # Not all 2xx messages have a content body so here we make sure # out Content-Length > 0 to avoid a parse error if (!search("^Content-Length:\0")) { force_rtp_proxy(); }; # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

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[Serusers] STUN, rtpproxy problem
by support 17 Feb '05

17 Feb '05

Hi, I am running ser-0.8.14 with rtpproxy and nathelper. I have also enabled STUN on the same server. My SIP UA supports STUN, but my PSTN gateway does not. If I don't enable STUN on SIP UA, I can route the call to PSTN gateway successfully using rtpproxy. If I enable STUN on SIP UA, I cannot route the call out to PSTN gateway. On ringing sound is heard on callie, but no voice passing through. I use t_relay_to_udp to forward the call to PSTN gateway. It seems that if STUN is supported on SIP UA, the call cannot be routed to PSTN gateway. Hope someone could fix this problem. Best wishes, Thomas My ser.cfg is as follows: # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received #if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; #}; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && ! search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else { fix_nated_contact(); }; }

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Re: Re: [Serusers] mediaproxy/rtpproxy
by Nhadie 17 Feb '05

17 Feb '05

Hi Sir, Thank you for your reply. I was able to run ser using the configuration below. How would I know Sir that it's forcing it to use rtpproxy, I was able to make test calls, but using also the SIP sevrer as the NAT gateway (I don't have that much equipment to test with :( ) So I think voice will really go through on each since the SIP server is just a hop away from the UA's. Also i'd like to force only rtpproxy when only making calls to local extensions, not going to PSTN. I tried adding this location was save: if (uri=~"^(sip:)[1-9]...@nhadzter\.com) { route(1); break; }; But I can't make any calls, and when I do ngrep on port 5060, I can see only INVITE's coming from the caller. It seems that it can't find the CALLEE or the CALLE doesn't respond. I'd really appreciate any help you can give me. Thank you so much. ===================================================== loadmodule "/usr/local/lib/ser/modules/nathelper.so" # We will you flag 6 to mark NATed contacts modparam("registrar", "nat_flag", 6) # Enable NAT pinging modparam("nathelper", "natping_interval", 60) # Ping only contacts that are known to be # behind NAT modparam("nathelper", "ping_nated_only", 1) ): # special handling for NATed clients; first, nat test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding used); also, # the received test should, if complete, should check all # vias for presence of received if (nat_uac_test("3")) { # allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # # Forcing media relay if necessary # route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(6)) { force_rtp_proxy(); t_on_reply("1"); append_hf("P-Behind-NAT: Yes\r\n"); }; if (!t_relay()) { sl_reply_error(); break; }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); }; }

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[Serusers] Logging of Destination Set
by Martin Koenig 17 Feb '05

17 Feb '05

Hello, is it possible to log to syslog the current destination set (according to loose_route() or append_branch() etc.)? With best regards, Martin

2 1

[Serusers] nathelper & STUN
by Tamas J 17 Feb '05

17 Feb '05

Hello! I used to use ser+nathelper+rtpproxy and it worked well. Unfortunately I found some buggy routers which have some SIP support but what they have is buggy (e.g. Draytek Vigor 2104 v2.3.6). They rewrite the IP in SIP message (in this version REGISTER message is not rewritten, but INVITE). While I have www/proxy_authorize and ser sends back auth. request, the phone (GS BT101) does not respond. I guess this is because it remembers for internal IP and receives public IP in message - because router rewrote it. For this I think there are 2 solutions: 1. Upgrade firmware which is many times not an option while the user may not have the knowledge to make it and this can be a real pain. 2. Use STUN, so this rewriting of router won't make any effect. While not every router is buggy, I would like to leave nathelper as is and somehow detect if the UA is behind NAT or not. Andrei suggested to use force_rport() in the main section (not just inside the NAT section), but I guess, this will mean, that every call's every RTP will pass through rtpproxy - which is not desirable. As I was looking for messages, the Call-Id also has the private IP portion, e.g.: Call-ID: 1400999426(a)192.168.24.37 My question would be: how to filter this part of sip message in ser configuration? I'm far-far not an expert to be able to do that :( I'm using 0.8.14. Thanks in advance, Tamas

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