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[Serusers] file missing -> init.d
by Armando Alvarez 11 Feb '05

11 Feb '05

Hello, I had installed the ser-0.8.14_linux_i386.tar.gz in my PC that is using Linux White Box. But in the 3rd point says that I have to include an init.d script. How can I create that? If somebody knows where to find it, please tell me. Its gonna be a glad notice. Regards, Armando Álvarez. Mexico City

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[Serusers] Minor Help
by Terry Mac Millan 11 Feb '05

11 Feb '05

Good day all, I am running ser-0.8.14 and sems 2004-07-27 on a Linux box with the config attached below. The way that the config stands at the moment, It currently logs BYE and PRACK traffice to syslog and the MySQL database. I have tried a few places to try to get it to log the INVITE's so that I can account for the traffic. This config is also a slightly modified version of the sample config from the SIP.edu cookbook. At current I do not have a gateway box to the PSTN. If it is possible, could someone with much more knowledge of ser assist me in cleaning up of the route statements, and provide the correct location to put the setflag(1); so that INVITE's are recorded to the accounting logs. Or if possible, someone provide a config that will do what I am looking for that I can adjust the settings to my setup and get working on my machine. Thanks Terry ----- Attached ser.cfg file ----- # ------------- version 0.8.11-0 # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=216.94.106.12 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". # dns - Uses dns to check if it is necessary to add a "received=" field # to a via. Default is no. # rev_dns - Same as dns but use reverse DNS. dns=no rev_dns=no port=5061 children=4 # check_via - Turn on or off Via host checking when forwarding replies. # Default is no. arcane. looks for discrepancy between name and # ip address when forwarding replies. check_via=yes # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. #syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # user | uid - uid to be used by the server. 99 = nobody. #uid="sip" # group | gid - gid to be used by the server. 99 = nobody. gid="sip" # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. mhomed=0 # ------------- external 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # set ";lr" tag to ";lr=true" modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_url", "sql://ser:xxxx@localhost/ser") modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "failed_transactions", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:xxxx@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:xxxx@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------- routing logic route { # ------------- routine checks # stop forwarding at 10 hops to prevent infinite loops if (!mf_process_maxfwd_header("10")) { log(1, "LOG: Too many hops\n"); sl_send_reply("483", "Too many hops"); break; }; # prevents private ip space from being used if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { if (method=="REGISTER") { log(1, "LOG: Someone trying to register from private IP\n"); sl_send_reply("479", "Please don't use private IP addresses" ); break; }; }; # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); break; }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(216\.94\.106\.12|(hostthe3rd\.)?rivernet\.net)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # challenge/response if (!www_authorize("hostthe3rd.rivernet.net", "subscriber")) { www_challenge("hostthe3rd.rivernet.net", "0"); break; }; # only registered users are allowed if (!is_user("replicator") & !check_to()) { log(1, "LOG: unregistered user registration attempt\n"); sl_send_reply("403", "Only registered users are allowed"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # process traffic local to BigU and the PSTN # Find the canonical username lookup("aliases"); # check domain again, if it is not still local after the alias # table lookup, just send it on its way. We do not authenticate # traffic we forward if (!(uri=~"^sip:(.+@)?(216\.94\.106\.12|(hostthe3rd\.)?rivernet\.net)([:;\?].* )?$")) { route(5); break; }; # now check for destinations through the gateway. 911 and 9911 # are always sent to the gateway. The assumption is that other all # numeric usernames between 5 and 20 digits are really pstn numbers # and so they are routed to the gateway # if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{5,20}@.*") ) { # route(3); # break; # }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # handle local SIP destinations not found in usrloc db # mostly offline or non-existent users if (!lookup("location")) { route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@216\.94\.106\.1([;:].*)*" ) { log(1, "LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ # ------------- process traffic leaving BigU for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==216.94.106.12) & !(proxy_authorize("hostthe3rd.rivernet.net", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log(1, "LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log(1, "LOG: failed outbound authentication for CANCEL granted\n"); } else { setflag(1); proxy_challenge("hostthe3rd.rivernet.net", "0"); break; }; setflag(1); }; # to maintain credibility of our proxy, we check From in INVITEs if (!src_ip==216.94.106.12 & method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); setflag(1); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN # # At this time, I have no external gateway to PSTN # route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately # if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") ) { # rewritehostport("10.1.2.5:5060"); # forward(uri:host, uri:port); # break; # }; # five digit numeric addresses are internal freebies sent to the pbx # without authentication # if (uri=~"^sip:[0-9]{5}@(10.1.2.3|(proxy\,)?\.bigu\.edu)") { # rewritehostport("10.1.2.5:5060"); # forward(uri:host, uri:port); # break; # }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(216.94.106.12|(hostthe3rd\.)?rivernet\.net)") { if (!(src_ip==216.94.106.12 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("hostthe3rd.rivernet.net", "subscriber")) { proxy_challenge( "hostthe3rd.rivernet.net","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; # if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance # if (uri=~"sip:91[0-9]*@.*") { # if (!is_user_in("credentials", "ld")) { # sl_send_reply("403", "Local calls only"); # break; # }; # }; # }; # authenticated and authorized, now accounting is set setflag(1); }; # rewritehostport("10.1.2.5:5060"); # append_hf("P-hint: GATEWAY\r\n"); # if (!t_relay()) { # sl_reply_error(); # break; # }; } # ------------- process calls for users offline route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-BigU domain # are not authenticated by BigU route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail # # I have SEMS installed on the machine, just don't have the intigration code in the cfg yet # failure_route[4] { # append_branch("sip:80000@216.94.106.12"); append_urihf("CC-Diversion: ", "\r\n"); # append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); # t_relay(); }

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[Serusers] re: nathelper in 0.8.14
by Yair Hakak 11 Feb '05

11 Feb '05

hello, can anyone confirm that the "ping_nated_only" modparam was added to nathelper in 0.8.14? I am running 0.8.12 and ser doesn't start when i add the following line: modparam("nathelper", "ping_nated_only", 1) in the error log is: Feb 11 13:12:48 SMD50225 ser: set_mod_param_regex: parameter <ping_nated_only> not found in module <nathelper> also i get a whole bunch of unkown command errors, but one thing at a time. thanks, yair

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Re: [Serusers] SER and Radius, call control question.
by Tom Gaudasinski 11 Feb '05

11 Feb '05

What do you mean by the state of the call? Where in a sip header is that indicated? Daniel-Constantin Mierla wrote: > SER does not keep the state of the call, so it cannot cut the call. > You need a back-to-back user agent to do that. The previous mail was > about controlling the call before ringing the callee, based on > caller's profile. > > Vovida has an implementation of b2bua, but I never tried and I do not > know whether one can control it via Radius or not. To do it only with > ser you need to develop a new module to keep the state of the call. > > Daniel > > On 02/10/05 21:57, Tom Gaudasinski wrote: > >> Ok, so you're saying that the only way to do this is through the auth >> module? Any chance there can be something along the lines of a >> prepaid system done with SER? a constant query to radius and when >> call credits are exhausted, radius tells SER to cut the call? >> >> Daniel-Constantin Mierla wrote: >> >>> you can group the users so you know who is allowed to call or not a >>> certain destination. For RADIUS you have to use group_radius to >>> check user group membership. An example of how to allow/deny local, >>> long distance, international calls you can find in the examples >>> directory -- online on cvs at: >>> >>> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/examples/pstn.cfg?… >>> >>> >>> It uses the group module with database backend -- you would need to >>> replace module "group" with "group_radius". >>> >>> acc_*() methods are only for accounting purposes. >>> >>> Daniel >>> >>> On 02/10/05 20:30, Tom Gaudasinski wrote: >>> >>>> Greetings, >>>> I'd like to know a few things. First, can SER control whether a >>>> call is connected or not through radius? For example, SER contacts >>>> radius and radius returns whether the user is allowed to make the >>>> call or not, then SER dials out or not depending on the answer. So >>>> this could be per destination and per user, some users not being >>>> able to call to certain destinations. Is this possible? If so, >>>> could someone provide a small example. Also I'd like to know what >>>> acc_rad_request() returns, is it a bool? What is it's purpose, does >>>> it help with the formentioned problem, or is this acc_rad only to >>>> do with CDR style data dumping? >>>> >>>> Thank you. >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >> >

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[Serusers] Query on the Back-Up Server
by Jhon miller 11 Feb '05

11 Feb '05

Hi all, Could any one help me to understand the back-up servers concept. In the RFC 3263, section 2 says the following " It is possible for elements to fail in the middle of a transaction. For example, after proxy 2 forwards the request to UA 2, proxy 1 fails. UA 2 sends its response to proxy 2, which tries to forward it to proxy 1, which is no longer available. The second aspect of the flow in the introduction for which DNS is needed, is for proxy 2 to identify a backup for proxy 1 that it can send the response to." This talks about the Proxy 2's work ,but how the backup server will collect the Details of the calls handled by the Proxy 1?. Thanks in advance John Miller -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

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[Serusers] SER compliance with IETFs RFCs and drafts
by regine.vital＠bull.net 11 Feb '05

11 Feb '05

Hello, Please, is anybody able to provide the complete list of RFCs and drafts SER is compliant to: - in the current official release 8.14 - in the source archive ? Thank you for your help

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[Serusers] error in Sql query, line 43
by Rosario Pingaro 11 Feb '05

11 Feb '05

It was a permission problem. Thanks but now I have another problem when I click on email link to confirm a subdcription I get this error: error in Sql query, line 43. It is about confirmation.php but I didnt find anything strange. Any help? thanks ----- Original Message ----- From: "Terry Wilson" <otherwiseguy(a)gmail.com> To: "Rosario Pingaro" <rpingar(a)italycom.it> Sent: Thursday, February 10, 2005 7:19 PM Subject: Re: [Serusers] Warning: fopen(/tmp/ser_fifo): >> When I access the serweb I get this error: >> >> Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied >> in >> /usr/local/apache2/htdocs/serweb/functions.php on line 206 >> >> Howto fix it? > > Try adding 'fifo_mode=0666' to your ser.cfg file and restarting ser. >

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[Serusers] Could SER-0.9 Be Crashing My ATAs?
by Java Rockx 11 Feb '05

11 Feb '05

Hi All. I'm using ser-0.9 with mediaproxy-1.2.0 We've been trying to track down a problem whereby different ATAs stop working either partially or crash entirely. We have a UTstarcom iAN-02EX with firmware 1.8.2.15a (released a few weeks ago). When I telnet to the ATA and enable debug messages I see this: Not a valid SIP request! Error: Parse error... The above error is generated everytime SER sends a UDP keep-alive packet shown below: 23:35:21.494441 pcp02356144pcs.clbrtn01.fl.comcast.net.5060 > ssvsoho137.smartcity.net.58317: udp 4 (DF) [tos 0x10] 0x0000 4510 0020 69c2 4000 4011 24ee 4454 f278 E...i.@.@.$.DT.x 0x0010 425a 32e6 13c4 e3cd 000c 5c37 0000 0000 BZ2.......\7.... Can anyone tell me if this is a SER issue or not? Regards, Paul

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[Serusers] HELPME SER+NATHELPER+RTPPROXY
by Sergio Gerardo Festa 11 Feb '05

11 Feb '05

Hello it help that version I have to use so that operate correctly SER+NAT+RTPPROXY SER version =? Nathelper version =? Rtpproxy version=? thanks Saludos. A su Disposición. Sergio Gerardo Festa Dto. Técnica TEL 011-52190654/52/53 // 4334-1014 24HS 011-15-5337-9684

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RE: [Serusers] UDP Fragmentation Problem
by Giudice, Salvatore 11 Feb '05

11 Feb '05

How large are the sip messages? Do you have an SBC deployed at one/both locations? I've observed that SER will truncate message that it forwards if they become too large. This is usually happens with two clients behind an SBC when large via parameter have been inserted to maintain SBC state within the sip dialog. Both Ingate and Jasomi products do this for example. The result of these large via's is that the message become too large and SER tend to truncate the SDP portion of the message. The connection generally sets up with regard to signaling, but the media path always fails. I have observed this behavior while making a sip to sip call between two xten clients, both positioned behind an ingate separator. Both clients were registered to the same SER with RtpProxy (Redhat Enterprise AS 3.0). -----Original Message----- From: Andrei Pelinescu-Onciul [mailto:pelinescu-onciul@fokus.fraunhofer.de] Sent: Monday, November 01, 2004 12:32 PM To: Bastian Schern Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] UDP Fragmentation Problem On Oct 30, 2004 at 01:28, Bastian Schern <ml01(a)in-bln.de> wrote: > Hi everybody, > > I've got no success to get a friend in Bogota (Colombia) connected to my > SER. He has got a ISDN Internet connection and the UDP packets will be > fragmented. It seems that the MTU of this connection is round about 400 > to 500 Bytes. Therefore most UDP-SIP packages are fragmented. > Is SER not able to handle fragmented UDP packages? The packets are defragmented by the ip stack of the OS, so ser doesn't really care if they arrive fragmented or not. > Is it possible to use SIP over TCP with X-Lite? > Or has somebody another hint for me? If you plan to use TCP, keep in mind that nat traversal won't work (if you use it). Andrei _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Symmetric NATs and RTP Proxy - Question
by Darren Sessions 11 Feb '05

11 Feb '05

We currently do not use an RTP proxy in our service (so the audio does not ride our internet bandwidth). Our biggest issue at the moment is the redundancy between two SER servers in dealing with symmetric NATs (specifically dealing with the individual SER server unique IP addresses and the far end customer's symmetric NAT). If we were to use an RTP proxy, as a backup mechanism for dealing with NATs, would this alleviate the issue of multiple SER servers and symmetric NATs?

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[Serusers] Any way to send a reply to a UA that will put it back to a dial tone?
by Evan Borgstrom 10 Feb '05

10 Feb '05

Does anyone know a way to send a response to a user agent that will force it to return to a dial tone. I've been through the RFC's and can't really find anything and I'm assuming that it's not possible but I thought I'd ask. -Evan

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[Serusers] SER and Radius, call control question.
by Tom Gaudasinski 10 Feb '05

10 Feb '05

Greetings, I'd like to know a few things. First, can SER control whether a call is connected or not through radius? For example, SER contacts radius and radius returns whether the user is allowed to make the call or not, then SER dials out or not depending on the answer. So this could be per destination and per user, some users not being able to call to certain destinations. Is this possible? If so, could someone provide a small example. Also I'd like to know what acc_rad_request() returns, is it a bool? What is it's purpose, does it help with the formentioned problem, or is this acc_rad only to do with CDR style data dumping? Thank you.

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[Serusers] serctl moni question.
by Ricardo Martinez 10 Feb '05

10 Feb '05

Hello list. I used the "serctl moni" command and i have found these statistics : Transaction Statistics Current: 0 (40 waiting) Total: 584 (0 local) Replied localy: 1082 Completion status 6xx: 27, 5xx: 7, 4xx: 212, 3xx: 0,2xx: 356 I'm particular interested in the Transactions Statics - Current and the 40 Waiting. Does this means something? Thanks in advance. Ricardo Martinez

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[Serusers] change NOTIFY to INVITE?
by Sean Watkins 10 Feb '05

10 Feb '05

Hi, Interesting problem I have; a vm system sends out NOTIFY but the PBX I'm using attached via SER & Sip requires a INVITE to a pilot light to turn off the VM for a specific extension? Any way todo this? I want to rewrite the NOTIFY to be a INVITE And possibly change some of the content of the message? Sean

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[Serusers] Warning: fopen(/tmp/ser_fifo):
by Rosario Pingaro 10 Feb '05

10 Feb '05

Sorry about a stupid question but I'm pretty new to linux and ser. When I access the serweb I get this error: Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied in /usr/local/apache2/htdocs/serweb/functions.php on line 206 Howto fix it? thanks Rosario

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[Serusers] SER Accounting to MYSQL - NO newbee-question
by Michael Treiber 10 Feb '05

10 Feb '05

Hi! using ser 0.8.14@gentoo + serweb for several months. want to enable accounting to mysql so i can see transactions in serweb. i can see the INVITE / BYE - logging in log-file: Feb 10 10:32:41 [/usr/sbin/ser] ACC: transaction answered: call_id=20007e1d3f0e277f, totag=ezzzyno2cs, from=XXXXXXX<sip:XX@XXXXXXXX.XXX>;tag=bb025f6b, i-uri=sip:XXXXXX@XXXXXX.XXX, method=INVITE, o-uri=sip:XX@XXXXXXXX.XXX:10003;line=7lr13il5, fromtag=bb025f6b, code=200, to=<sip:XX@XXXXXXXX.XXX>;tag=ezzzyno2cs, uid=n/a, userpart=mtreiber_ Feb 10 10:32:49 [/usr/sbin/ser] ACC: transaction answered: call_id=20007e1d3f0e277f, totag=ezzzyno2cs, from="YYY"<sip:XX@XXXXXXXX.XXX>;tag=bb025f6b, i-uri=sip:XXXXX@XXXXXXXXXX:10003;line=7lr13il5, method=BYE, o-uri=sip:XXXXX@XXXXXXXXXX:10003;line=7lr13il5, fromtag=bb025f6b, code=200, to=<sip:XXXXXXX@XXXXX.XXX>;tag=ezzzyno2cs, uid=n/a, userpart=mtreiber_ BUT SER is NOT logging to MySQL.. :( theres no attempt from ser to write accounting-data to mysql. all other mysql-request (register,userlookups, etc.. ) are ok. i set "DEFS+=-DSQL_ACC" in acc-module@compile. modules are loaded: accounting-loadmodule "/usr/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/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/print.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/uri.so" modparam are set: modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "failed_transactions", 1) flags are set correct: if(method=="INVITE" || (method=="ACK") || method=="BYE" ){ setflag(1); }; if you have ANY idea please let me know... regards, Michael.

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[Serusers] accessing mySQL db by means of external scripts...
by Esteban Maestre 10 Feb '05

10 Feb '05

Hi, SERusers... I have a question: in case i want to access and write into mySQL ser database tables from an external script I call by using exec_msg(), will this data remain there permanently? Will SER overwrite it? Thanks in advance... :) Esteban

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[Serusers] SRR
by Darren Sessions 10 Feb '05

10 Feb '05

Is SRR multi-homed capable ? If so, what is the cmd line and/or srr.cfg entry. Thanks, - Darren

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RE: [Serusers] fifo_db_url
by Giudice, Salvatore 10 Feb '05

10 Feb '05

I added the following global declaration to my script and the warning goes away. Use the same db_url you use for usrloc or the domain module. fifo_db_url="mysql://account:password@1.2.3.4/ser" -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, October 26, 2004 5:45 PM To: Emilio Panighetti Cc: SER Users Subject: Re: [Serusers] fifo_db_url It's just a warning, it is related to new code that can provide database access over fifo. Jan. On 25-10 23:56, Emilio Panighetti wrote: > Hello > > Why am I getting this error with the latest version on CVS? > > Oct 25 23:54:37 is1 ser[16888]: WARNING: no fifo_db_url given - fifo DB > commands disabled! > > Is there a config parameter I need to add? > > Thanks, > E. > > _______________________________________________ > 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

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[Serusers] (no subject)
by Ernesto Lucero 10 Feb '05

10 Feb '05

Hello everybody I am new to the ser Proxy and to Linux does some have a ser step by step manual on how to instal and configure Will greatly appreciate Thanks Ernesto Lucero Valadez Btel Comunicación elucero(a)btel.com.mx <mailto:elucero@btel.com.mx> Tel. 635-63-00 ext 6383

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[Serusers] Pre-paid and post-paid accounting and billing
by Narcis GRATIANU 10 Feb '05

10 Feb '05

Hello ! Does anybody know any pre-paid and post-paid billing system that works with SER. An open source one will be better, but also a commercial version will do. Thank you !

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RE: [Serusers] bad call flow
by Ricardo Martinez 10 Feb '05

10 Feb '05

Hello. I think there is a problen with your configuration file. You have : if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; so you're not relaying all the SIP messages, maybe this can work : if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); break; }; Hope this work. Good luck.!! Ricardo Martinez.- -----Mensaje original----- De: Todor [mailto:todorakis@gmail.com] Enviado el: Jueves, 10 de Febrero de 2005 8:15 Para: serusers(a)lists.iptel.org Asunto: [Serusers] bad call flow Hi All, I have a serious problem with SIP call flow over SER. My SER don't retransmit ACK first and after that don't retransmit BYE message, too. Can any one advice me? what can be a reason? Here are call flow and ser.cfg. SIPPHONE------------------------SER--------------------------CISCO SIP GW |-------------INVITE------------>| | |<----------TRying---------------| | | |-----------INVITE-------------->| | |<----------TRying---------------| | |<------------183----------------| |<------------183----------------| | | |<-------------OK----------------| |---------------ther is two way RTP channel ----------------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| SER don't retransmit ACK | | | and Cisco timedout after 19sec | | |<-------------BYE---------------| | |<-------------BYE---------------| | |<-------------BYE---------------| | SER don't retransmit BYE |<-------------BYE---------------| | |<-------------BYE---------------| # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # ----------- global configuration parameters ------------------------ debug=8 fork=yes 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 ---------------------------------- 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 --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- 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; }; # authenticate register message using radius if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; # route (proxy) invite message to gateway if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; }

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Re: [Serusers] bad call flow
by Todor 10 Feb '05

10 Feb '05

10x Ricardo it is working now ----- Original Message ----- From: Ricardo Martinez To: 'Todor' ; serusers(a)lists.iptel.org Sent: Thursday, February 10, 2005 3:27 PM Subject: RE: [Serusers] bad call flow Hello. I think there is a problen with your configuration file. You have : if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; so you're not relaying all the SIP messages, maybe this can work : if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); break; }; Hope this work. Good luck.!! Ricardo Martinez.- -----Mensaje original----- De: Todor [mailto:todorakis@gmail.com] Enviado el: Jueves, 10 de Febrero de 2005 8:15 Para: serusers(a)lists.iptel.org Asunto: [Serusers] bad call flow Hi All, I have a serious problem with SIP call flow over SER. My SER don't retransmit ACK first and after that don't retransmit BYE message, too. Can any one advice me? what can be a reason? Here are call flow and ser.cfg. SIPPHONE------------------------SER--------------------------CISCO SIP GW |-------------INVITE------------>| | |<----------TRying---------------| | | |-----------INVITE-------------->| | |<----------TRying---------------| | |<------------183----------------| |<------------183----------------| | | |<-------------OK----------------| |---------------ther is two way RTP channel ----------------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| SER don't retransmit ACK | | | and Cisco timedout after 19sec | | |<-------------BYE---------------| | |<-------------BYE---------------| | |<-------------BYE---------------| | SER don't retransmit BYE |<-------------BYE---------------| | |<-------------BYE---------------| # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # ----------- global configuration parameters ------------------------ debug=8 fork=yes 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 ---------------------------------- 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 --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- 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; }; # authenticate register message using radius if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; # route (proxy) invite message to gateway if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; }

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[Serusers] bad call flow
by Todor 10 Feb '05

10 Feb '05

Hi All, I have a serious problem with SIP call flow over SER. My SER don't retransmit ACK first and after that don't retransmit BYE message, too. Can any one advice me? what can be a reason? Here are call flow and ser.cfg. SIPPHONE------------------------SER--------------------------CISCO SIP GW |-------------INVITE------------>| | |<----------TRying---------------| | | |-----------INVITE-------------->| | |<----------TRying---------------| | |<------------183----------------| |<------------183----------------| | | |<-------------OK----------------| |---------------ther is two way RTP channel ----------------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| | | |<-------------OK----------------| |<-------------OK----------------| | |--------------ACK-------------->| SER don't retransmit ACK | | | and Cisco timedout after 19sec | | |<-------------BYE---------------| | |<-------------BYE---------------| | |<-------------BYE---------------| | SER don't retransmit BYE |<-------------BYE---------------| | |<-------------BYE---------------| # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # ----------- global configuration parameters ------------------------ debug=8 fork=yes 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 ---------------------------------- 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 --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- 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; }; # authenticate register message using radius if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; # route (proxy) invite message to gateway if (method=="INVITE") || (method=="CANCEL") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; record_route(); rewritehostport("cisco_sip_gw_ip:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; }

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