- sr-users - kamailio.org

[Serusers] Contract work needed for custom SER configuation
by Andy Pyles 07 Dec '05

07 Dec '05

Please contact me privately if you are very familiar with NATHELPER and AVPOPS modules. I'm looking for a custom configuration solution, that needs to be delivered very quickly. Thanks, Andy

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Re: [Serusers] set timer for "Request Timeout"
by Matteo Piazza 07 Dec '05

07 Dec '05

Hi Jens Can you send me your complete ser.cfg file. I'am interested to see you complete configuration Regards Matteo Hello all, I have following records in the table 'usr_preferences': username attribute value 1000 fwdbusy sip:1001 at example.de 1000 fwdnoanswer sip:1001 at example.de I use these records to get the functionallity of "Call Forwarding - Busy" and "Call Forwarding - No Answer" into the OpenSer - server. I use a failure_route to act on the SIP - messages "Request Timeout (408)" and "Busy(486)" and forward the INVITE - message to the URI saved in the table 'usr_preferences'. See my snippet of the openser.cfg: ----snip --- failure_route[1] { if (isflagset(27) && t_check_error(408)) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); route(6); break; }; }; if (isflagset(26) && t_check_error(486)) { if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); resetflag(26); route(6); break; }; }; }; ----snap --- These configuration works fine. Is there a possibillity to set a timer for the phone after the server receives the message "Request Timeout" from the phone? Or can I let the server ring for example five times and generate the message? Greetings Jens ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

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[Serusers] How to Minimize rtp proxy usage for all cases including when behind the same NAT
by S G 07 Dec '05

07 Dec '05

Hello, I have SER setup to only use the rtpproxy if the client is behind a symmetric NAT. This is accomplished by using STUN since STUN will not modify the contact headers if it detects a symmetric NAT. So SER see's the local address in the contact header and only forces rtpproxy for those types of call. When users are behind the same NAT then ser detects this using AVP's and force_rtpproxy is not used for the call. This only works when STUN is not enabled. Since when STUN succeeds the orignal local address of the phone is changed to the public address in the sip message. So when two STUN enabled clients try to call each other from behind the same nat the call fails. The call fails because the NAT does not support harpin of media as most don't. The only way this call would work is if the contacts could be changed back to use their local addresses pre-STUN. How can SER be instructed to use the local address for calls behind the same NAT when STUN is enabled? One way I can think of doing this is to change nathelper to extract the original IP from the VIA header and rewrite the SDP. In the SIP trace i see only IP's in contact headers and SDP are changed when STUN succeeds. Is there a simpler way to accomplish this? Thanks, Sumeet _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Users] CALL FORWARD / PSTN GW
by Kjell Arild Frigstad 07 Dec '05

07 Dec '05

Hello! I am trying to implement Call forwarding. Server runs openser 1.0.0 and I have a Cisco PSTN GW where only defined calling numbers are permitted (because of accounting on the "upstream" provider). My problem is when a call from PSTN is beeing forwarded and I have to "turn it back" to the PSTN gatway. Calls from : 1) Sip server ---> SIP-CALL-FW ---> PSTN GW OK, the calling number is also changed to the number which has turned call forwarding on. The display on the phone also showing the number which has turned Call forwarding on.. 2) PSTN ---> SIP-CALL-FW ---> SIP SERVER Works OK 3) PSTN ----> SIP-CALL-FW ---> PSTN is not working. Dialpeer matching in Cisco not OK because calling number not OK. I can define a way so that all calls can go through the GW. Then I sometimes get a "482 loop detected". The disply on the phone showing the orginal calling number (not the number wich hat turnd Call forwarding on). Is this the right way to "turn" a call back to where it came from? Have a missed a "rebuild" of the message? if (avp_db_load("$ruri","$fw_to") & method=="INVITE") { # CALL Forward log(1, "INFO: AVP_db_load fw_to true \n"); #save called number avp_write("$ruri/username","s:org_to"); #Insert call forward number avp_pushto("$ruri/username","$fw_to"); #is call forward number local or to PSTN if (does_uri_exist()) { log(1, "INFO:Call FW to Local users\n"); if (!lookup("location")) { log(1, "INFO: call FW to local user, not online\n"); route(4); return; }; log(1, "INFO:Call FW to local users online\n"); uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060"); if (!t_relay()) { sl_reply_error(); return; }; return; } else { log(1, "INFO: Call FW to PSTN\n"); #here is somting wrong. What??? #move Called number as From number and send to PSTN GW uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060"); rewritehostport("80.239.101.34:5060"); append_hf("P-hint: call fwd to PSTN\r\n"); append_branch(); record_route(); if (!t_relay()) { sl_reply_error(); return; }; }; return; } Cisco trace : calling from 38092382 (PSTN) to 38707130 (on the SIP server) witch is forwarded to 95066833 (PSTN) Call comming in from PSTN to SIP-server : Sent: (from Cisco) INVITE sip:38707130@212.125.213.151:5060 SIP/2.0 Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38092382@80.239.101.34>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Date: Fri, 02 Dec 2005 18:41:38 GMT Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 457212883-1654264282-2767847443-3293903248 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: <sip:38092382@80.239.101.34>;party=calling;screen=yes;privacy=off Timestamp: 1133548898 Contact: <sip:38092382@80.239.101.34:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 253 v=0 o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34 s=SIP Call c=IN IP4 80.239.101.34 t=0 0 m=audio 16788 RTP/AVP 18 8 0 c=IN IP4 80.239.101.34 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 *Dec 2 18:41:38.278: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38092382@80.239.101.34>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 CSeq: 101 INVITE Server: OpenSer (1.0.0 (i386/linux)) Content-Length: 0 Warning: 392 212.125.213.151:5060 "Noisy feedback tells: pid=29232 req_src_ip=80.239.101.34 req_src_port=55717 in_uri=sip:38707130@212.125.213.151:5060 out_uri=sip:95066833@80.239.101.34:5060 via_cnt==1" *Dec 2 18:41:38.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:95066833@80.239.101.34:5060 SIP/2.0 Record-Route: <sip:212.125.213.151;ftag=2A699224-CC6;lr=on> Via: SIP/2.0/UDP 212.125.213.151;branch=z9hG4bK1e4.11a1a9d3.0 Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38707130@212.125.213.151:5060>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Date: Fri, 02 Dec 2005 18:41:38 GMT Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 457212883-1654264282-2767847443-3293903248 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 69 Remote-Party-ID: <sip:38092382@80.239.101.34>;party=calling;screen=yes;privacy=off Timestamp: 1133548898 Contact: <sip:38092382@80.239.101.34:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 253 P-hint: call fwd to PSTN v=0 o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34 s=SIP Call c=IN IP4 80.239.101.34 t=0 0 m=audio 16788 RTP/AVP 18 8 0 c=IN IP4 80.239.101.34 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 Dial peer maching (call forward from SIP server). Calling number is wrong, should be 38707130 (number who has put forwarding on) Calling Number=38092382, Called Number=95066833, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Dec 2 19:44:08.330: //-1/D6732AD9A532/DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore: Calling Number=, Called Number=95066833, Peer Info Type=DIALPEER_INFO_SPEECH *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=95066833 Best regards Kjell Arild

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[Users] Limit simultaneous calls
by Christophe Irles 07 Dec '05

07 Dec '05

Hi, Is there a way to limit the number of simultaneous calls ? According the provider, the bandwith will not permit a lot of simultaneous call so in order to prevent very bad feeling I prefer to limit the use. In a second step, I wondering if this type of restriction can be extended according the phone domain. For example, simultaneous call betwwen two sub-sites of the same enterprise is limited to 10 but all others a limited to 5 for example. I know the ACC module but it's not the one fot this even if the information stored could resolve this point. Perhaps another module already exist ? Thanks, Christophe

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[Serusers] Extracting a substring from r-uri
by Steve Blair 07 Dec '05

07 Dec '05

I would like to extract a substring from the r-uri but cannot see a way to accomplish this task. Is there a function already available? 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@net.isc.upenn.edu

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[Users] how to configure SIP OpenSer redirect server
by Ajay Srivastava 07 Dec '05

07 Dec '05

Hi, I made changes in openser.cfg like this to act it as a SIP redirect server: route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); return; }; # rewrite current URI, which is always part of destination ser rewriteuri("sip:parallel@siphub.net <mailto:parallel@siphub.net> :9"); # append one more URI to the destination ser append_branch("sip:redirect@siphub.net <mailto:redirect@siphub.net> :9"); # redirect now sl_send_reply("300", "Redirect"); } then restarted openser, but still not getting redirect messages, should I need to do something more for enabling this option? Please help me and tell me that how can I run my redirect server for SIP. Regards, ajay

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RE: [Users] t_replicate after failure
by Douglas Garstang 07 Dec '05

07 Dec '05

Ok, I worked out that you need to use append_branch, like this: append_branch (proxy1) append_branch (proxy1) t_replicate (proxy3) That's what the devevelopers say anyway... However, all the t_replicate's are going to exactly the same proxy. Here's a post by two people that have reported _exactly_ the same problem. Is there a solution? http://www.archivum.info/serusers@iptel.org/2004-12/msg00268.html Thanks, Doug. -----Original Message----- From: Douglas Garstang Sent: Wednesday, November 30, 2005 4:11 PM To: users(a)openser.org Subject: [Users] t_replicate after failure In the scenario below, if the first t_replicate fails because 192.168.10.5 returns "Not Found", the second t_replicate command does not execute. Why? How can I make it execute? Call me crazy, but this doesn't seem at all intuitive to me. In a functional programming world, if the first fails, the next should still run. if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voip.com", "subscriber")) { www_challenge("voip.com", "0"); exit; }; save("location"); t_replicate("192.168.10.5","5060"); t_replicate("192.168.10.7","5060"); exit; }; Thanks, Doug. _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Monitoring on failure
by Jose Maria Gonzalez 07 Dec '05

07 Dec '05

I want to implement a monitor system when there is a failure route. At first I think to send an snmp message when there was a failure route, but the snmp module is frozen, isn't it? Now I want to write a line in a special log file. So I want to have the normal log file for monitor information (call stablish, call released, etc..) and another log file for errors. I though in use the exec module to inser the errors in the error-log file in this way: exec_dset("echo Error_failure_route >> file_error.log"); Does anyone have a better idea to do this? Because I think that the exec module is slower than use xlog. It is possible to execute that command in background? That openser does not stop the execution until the command has finished. Thanks -- Salu2: *************************************** José María González ***************************************

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[Users] local mysql access problem cont'd
by rick＠eltopia.com 07 Dec '05

07 Dec '05

Alright.. here is some dbg level 9 output of my problem. Basically openser starts, says it starts successfully, and then quits. I think it may have something to do with the fact that it is trying to access a local database even though it has been configured to access a remote one. I am quite sure that it is accessing the proper cfg, cause when I put garbage in it I get an error. Any help, clues, snide remarks, cryptic-guru-riddle-answers would be appreciated. Dec 1 11:48:41 amnesiac openser: 0(0) db_init: Connection 'mysql://openser:openserrw@localhost /openser' not found in pool Dec 1 11:48:41 amnesiac openser: 0(0) new_connection: Opening MySQL connection: mysql://opense r:openserrw@localhost/openser Dec 1 11:48:41 amnesiac openser: 0(0) new_connection: Can't connect to local MySQL server thro ugh socket '/var/lib/mysql/mysql.sock' (2) Dec 1 11:48:41 amnesiac openser: 0(0) register_udomain(): Can not open database connection Dec 1 11:48:41 amnesiac openser: 0(0) domain_fixup(): Error while registering domain Dec 1 11:48:41 amnesiac openser: ERROR: error -1 while trying to fix configuration Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : start Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: unlink_timer_lists : emptying DELETE list Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : emptying hash table Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : releasing timers Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : removing semaphores Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : destroying tmcb lists Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : done Dec 1 11:48:41 amnesiac openser: 0(3791) shm_mem_destroy Dec 1 11:48:41 amnesiac openser: 0(3791) destroying the shared memory lock Dec 1 11:48:41 amnesiac openser: openser startup succeeded my config file is simply the sample config, modified with the suggestions to access a remote mysql database, it looks like this: # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # 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/openser_fifo" # Uncomment this if you want to use SQL database loadmodule "//lib/openser/modules/mysql.so" loadmodule "//lib/openser/modules/sl.so" loadmodule "//lib/openser/modules/tm.so" loadmodule "//lib/openser/modules/rr.so" loadmodule "//lib/openser/modules/maxfwd.so" loadmodule "//lib/openser/modules/usrloc.so" loadmodule "//lib/openser/modules/registrar.so" loadmodule "//lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//lib/openser/modules/auth.so" loadmodule "//lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db", "db_url", "mysql://serusr:testpass@192.168.0.54/ser") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

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[Serusers] different db for different sip domain
by Iqbal 07 Dec '05

07 Dec '05

Hi As the subject says is it possible to configure ser to auth from different db, based on sip domain And has anyone set this up Iqbal

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[Serusers] How to handle retrasmissions?
by Federico Giannici 07 Dec '05

07 Dec '05

What is the correct way to handle message retrasmissions? Usually t_relay() seems to hadle them correctly. But if a transactions was previously created by some module function, I found that t_relay() (and subsequent sl_reply_error()) generates the following errors: Dec 5 18:49:25 eowyn OpenSER[5654]: ERROR: t_newtran: transaction already in process 0x502a56f8 Dec 5 18:49:25 eowyn OpenSER[5654]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) So, I have the following questions: 1) What is the correct script code to relay a message, even with retrasmissions and transaction already created? 2) Is it OK to simply discard retrasmissions (not forwarding or replying them)? 3) How can I recognize retrasmissions? Is t_lookup_request() the correct function for a test? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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Re: [Devel] [Users] nathelper/rtpproxy when both SIP UA are behind same NAT
by Tavis P 07 Dec '05

07 Dec '05

Opps i made a mistake, instead of "avp_write" you need to use "avp_printf" to store the pseudo variable into an AVP and when you are doing the comparison with avp_check you need an alias defined for the second AVP otherwise you might end up comparing the AVP with a literal value modparam( "avpops", "avp_aliases", "dest_ip=i:401" ) avp_printf("i:400", "$si"); avp_printf("i:401", "$rd"); if ( avp_check("i:400", "eq/$dest_ip/g") ) { xlog("L_INFO", "\"$si\" == \"$rd\" "); } Norman Brandinger wrote: > I agree that the solution that was presented is BAD for the reasons > stated by both of us and probably more :) > > Perhaps using $Ri after the successful lookup() and comparing it to > the $src_ip AVP would work. > > The problem was trying to compare a pseudo variable with an AVP. > > > The doc <snipped below> does not state that pseudo variables can be > used by avp_write(). > > Pseudo-variables can be used with following modules of OpenSER: > > * avpops - function “avp_printf()” > * xlog - functions “xlog()” and “xdbg()” > > I did try to use avp_write() to save a pseudo variable to an AVP but > was unsuccessful. > > Do you have an example of writing a pseudo variable to an AVP ? > > Regards, > Norm > > > > Tavis P wrote: > >> Another, more lightweight solution, is: >> >> After a successful lookup(location) you can compare the received IP >> address ($si) with the destination domain/ip address returned from the >> "lookup" function call ($rd), i think these two pieces of information >> are reliable for this purpose >> >> This way you don't have to make extra database calls to determine >> locality >> >> Also if you want to compare pseudo variables you can first write them to >> an AVP using "avp_write" and then you can make use of "avp_check" with >> data sourced from a pseudo variable >> >> Norman Brandinger wrote: >> >> >> >>> I have looked into the problem related to saving saving resources by >>> not calling rtpproxy or mediaproxy when both the caller and callee are >>> behind the same NAT. >>> >>> This topic has been discussed many times however, there have been no >>> "working" examples, or even partial examples posted (that I'm aware >>> of). >>> >>> Below is a solution to the problem, but is a BAD hack at best. >>> >>> This solution is BAD because AVP's are added to the usr_preferences >>> table during the registration process, but there is no notification >>> that a registration has expired and is no longer valid. A cron job >>> could be run periodically to delete all received_ip AVP's with no >>> matching entry in the location table, but this is another hack to fix >>> the first hack. >>> >>> The solution is BAD because a database calls must be made to save the >>> received_ip address into an AVP, then another database call must be >>> made to reload the value just so that it can be tested against the >>> caller's $src_ip. A pseudo variable $Ri already exists that contains >>> the value we use in received_ip, but I have not been able to find a >>> way for avp_check() to use $Ri. >>> >>> The solution is BAD because it will fail when the SIP device(s) are >>> behind more than a single NAT router. I believe this is not an issue >>> for 95% of the users and 99% of the small office or home users. >>> >>> I think that a better solution would be to enhance avp_check() to >>> accept pseudo variables for the "value" parameter. This would remove >>> the need for any database calls. >>> >>> I'm cross-posting this response to the developers list to bring this >>> to their attention and ask for suggestions (it's possible that I've >>> totally missed something). If a decision is made to open the avpops >>> module to pseudo variables (other than just avp_print()), I would >>> suggest the following be looked at: >>> >>> avp_check() the value parm should accept pseudo variables >>> avp_write() the value parm should accept pseudo variables >>> avp_pushto() the name parameter should accept pseudo variables >>> avp_op() the value parameter should accept pseudo variables >>> >>> Regards, >>> Norm >>> >>> >>> 1) During REGISTER processing, place the following code. I would >>> suggest that you put the code "after" all authentication checks and >>> "after" the save("location") statement. >>> >>> # Delete any previously saved IP addressess from the user. >>> avp_db_delete("$from/username","s:received_ip"); >>> >>> # Save the source IP address of the user into an AVP called >>> received_ip. >>> # The saved IP address should be the public address of the NAT router. >>> avp_write("$src_ip", "s:received_ip"); >>> >>> # Save the AVP received_ip into the usr_preferences tables associated >>> with the user that just registered >>> avp_db_store("$from/username","s:received_ip"); >>> >>> 2) During INVITE processing (or wherever you make the decision to use >>> or not use rtpproxy / mediaproxy) , place the following code. >>> >>> >>> #--------------------------------------------------------------------------- >>> >>> >>> # Are the caller and callee behind the same NAT ? >>> >>> #--------------------------------------------------------------------------- >>> >>> >>> avp_db_load("$to/username", "s:received_ip"); >>> if (avp_check("s:received_ip", "eq/$src_ip")) { >>> setflag(CALLER_AND_CALLEE_ ARE_BEHIND_THE_SAME_NAT); >>> }; >>> >>> >>> >>> _______________________________________________ >>> Devel mailing list >>> Devel(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/devel >>> >>> >>> >> >> >> >> >> > > >

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[Serusers] QoS and call quality
by Ashutosh kumar 07 Dec '05

07 Dec '05

Hi, Does SER have any control or any can it be configured (if yes, please guide) to implement various QoS mechanisms like Diffserv,RSVP, Admission control etc.., to improve the voice quality and delays and/or jitter? Because lately, we are facing a decrease in call quality in our ser calls. Or do we have to do this only at the network level, eg. Routers etc..? Any poiners/guides will be most welcome. Regards, Ashutosh Kumar

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[Serusers] avp invite timer not working?
by Kyriakos Mavromichalis 07 Dec '05

07 Dec '05

Im trying to set the invite timer to 6 seconds when i have a call to pstn gw but it does not seem to affect anything, This is how i set the timer: avp_write("5", "inv_timeout"); any ideas? -- Kyriakos Mavromichalis Otenet Telecom

3 7

[Users] Re: [Devel] unixODBC module compilation error
by Bogdan-Andrei Iancu 07 Dec '05

07 Dec '05

Hi Lenir, the warning is fixed on cvs - please update and retry. thanks for report, bogdan Lenir wrote: > Guys, > > > > I downloaded the most recent version of CVS, I get the following > compilation error, when compiling unixodbc module: > > make[1]: Entering directory `/usr/src/sip-server/modules/unixodbc' > > Compiling list.c > > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -minline-all-stringops -malign-double -falign-loops -mtune=athlon > -Werror -DNAME='"openser"' -DVERSION='"1.1.0-dev3 "' > -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 > -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC > -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP > -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK > -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 > -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL > -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c list.c -o list.o > > list.c: In function `insert': > > list.c:56: warning: control reaches end of non-void function > > make[1]: *** [list.o] Error 1 > > make[1]: Leaving directory `/usr/src/sip-server/modules/unixodbc' > > > > Any ideas? > > > > Lenir > > > >------------------------------------------------------------------------ > >_______________________________________________ >Devel mailing list >Devel(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/devel > >

1 0

[Users] UAC to BroadVoice
by Scott MacGregor 07 Dec '05

07 Dec '05

I am trying to route to a BroadVoice PSTN gateway using the UAC. In the failure_route the uac_auth command fails with the following errors: ERROR:uac:parse_authenticate_body: parse error in <DIGEST realm="BroadWorks",algorithm=MD5,nonce="1133904594381"> around 0 ERROR:uac:uac_auth: failed to parse auth hdr body code: failure_route[3] { if( t_check_status("401|407") ) { if( uac_auth() ) { append_branch(); route(1); } } } Thanks,

2 1

[Serusers] How to register MSN Messenger4.6 to non-default port of ser
by Manuel Seleiro 07 Dec '05

07 Dec '05

Manuel Seleiro Portugal!!!

2 2

[Users] symmetric fw
by Paul van Schagen 06 Dec '05

06 Dec '05

Hi, I am using STUN and that works well most of the time. I do have a problem with UA's who sit behind a symmetric firewall. ( a Microsoft server with ISA 2004 to be exact ). I understand this is a well known problem. What practical solutions can I use in combination with openser ? Thanks for any help Best regards Paul van Schagen

2 1

[Users] unixODBC module compilation error
by Lenir 06 Dec '05

06 Dec '05

Guys, I downloaded the most recent version of CVS, I get the following compilation error, when compiling unixodbc module: make[1]: Entering directory `/usr/src/sip-server/modules/unixodbc' Compiling list.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -Werror -DNAME='"openser"' -DVERSION='"1.1.0-dev3 "' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c list.c -o list.o list.c: In function `insert': list.c:56: warning: control reaches end of non-void function make[1]: *** [list.o] Error 1 make[1]: Leaving directory `/usr/src/sip-server/modules/unixodbc' Any ideas? Lenir

1 0

[Serusers] hierarchical VoIP system
by Joao Pereira 06 Dec '05

06 Dec '05

Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt

3 4

[Serusers] Writing TCL CLient to populate SER
by Selvasundar 06 Dec '05

06 Dec '05

hi, i have installed the SER in a REd hat linux machine. i want to write a test stub in TCL , and the stub will throw SIP messages to SER and will receive the same from SER. any one , please guide me in this regard. Thanks in advance Rgds, John The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in

1 0

[Users] session termination
by JF 06 Dec '05

06 Dec '05

Hi, I've been playing around with TM callbacks and the TM API trying to build a module which mantains dialog info in order to permit session termination. For now I managed (in a very experimental way) to have a function exposed to the script which initiates (appropriately built with dialog info) BYE transactions to UAs and effectively terminates the call. Later this could be done via FIFO or unix socket command taking some kind of "dialog identifier" parameter... But sending BYEs to terminate calls only works if the dialog is already in the CONFIRMED state. For early dialogs, CANCELs should be used. But in the TM API there is no cancel function. Is there any other way of canceling a request on behalf of an UA from within an (open)ser module? Or would the TM API need to be extended? Thanks in advance for any insights. JF

1 0

[Users] openserctl
by Douglas Garstang 06 Dec '05

06 Dec '05

How can we change the database, database user name and database password that the openserctl utility tries to connect with? You can set these in openser.cfg with modparam, but if you can't also set them for openserctl, what's the point? Thanks. Douglas Garstang

2 3

[Serusers] Call hunting w/lcr, how to set diff q values?
by Brian McCrary 06 Dec '05

06 Dec '05

Hello, I'm trying to setup a hunt group using the lcr module and different q values. When I try to add a user location with a different q value using the FIFO commands, the only q values that are accepted is 1.00 or lower, such as 0.99. However, these all seem to qualify as the same value as all of my locations are still contacted at the same time. Does anyone have this working, and how did you add the user locations in regards to the q value? Thanks, Brian

2 4

[Serusers] Web registration problem through ser_fifo
by Andrey Kouprianov 06 Dec '05

06 Dec '05

What is wrong with my PHP code, I have no idea. Could you please check it and let me know what's wrong. Im using SER 0.9.3, but ser_fifo documentation is not available, so i used one from ser 0.8.12 (admin guide). Im sure there's a little error, but I can figure out which one. Btw, below the code is the output from message log <? /* config values */ $domain="sip.mydomain.org"; $user="webme"; $web_aor="sip:".$user."@".$domain; $expires=3600; $type="student"; $fifo="/tmp/ser_fifo"; /* open reply fifo */ $replyfilename="webfifo_".rand(); $replypath="/tmp/".$replyfilename; echo "Initiating your request...<p>"; /* open fifo now */ $fifo_handle=fopen( $fifo, "w" ); if (!$fifo_handle) { exit ("Sorry -- cannot open fifo: ".$fifo); } /* construct FIFO command */ $fifo_cmd=":ul_add:".$replyfilename."\n". "location\n". //table $web_aor."\n". //aor "sip:".$user."@".$_SERVER['REMOTE_ADDR']."\n". //contact $expires."\n". //expires "0.00\n". //priority "0\n\n"; /* create fifo for replies */ system("mkfifo -m 666 ".$replypath ); /* write fifo command */ if (fwrite( $fifo_handle, $fifo_cmd)==-1) { unlink($replypath); fclose($fifo_handle); exit("Sorry -- fifo writing error"); } fclose($fifo_handle); /* read output now */ if (readfile( $replypath )==-1) { unlink($replypath); exit("Sorry -- fifo reading error"); } unlink($replypath); echo "<p>You are now signed in<p>"; ?> ///MESSAGE LOG Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: read_line: request line too long Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: ul_add: flags expected Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server: command must begin with :: location Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server: command must begin with :: sip:webme@203.159.32.50;type=student Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server: command must begin with :: 0.00 Dec 6 16:35:54 sip /usr/local/sbin/ser[206]: ERROR: ul_add: flags expected

2 1

Re: [Users] errors when making a call!
by Klaus Darilion 06 Dec '05

06 Dec '05

messa(a)innsof.com wrote: > Hi Klaus > > this command does not work for me! > but, my openser works with those errors! You have to install the "ngrep" application. E.g. "apt-get install ngrep" if you are using debian based distros. klaus > > bests regards! > Serge > > > > >>looks like a bad formated SIP message. >> >>Post the SIP signaling dump (ngrep -t -W byline port 5060) >> >>klaus >> >>messa(a)innsof.com wrote: >> >>>Hi all >>> >>> When imake calls, signalisation goes well but, when, i hang up my >>>client, i have no voice in any client! >>>And my openser server display methese errors: >>> >>>0(4335) lookup(): Error while extracting address of record >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) WARNING: do_action:error in expression >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) extract_aor(): Error while parsing Address of Record >>> 0(4335) lookup(): Error while extracting address of record >>> 0(4335) Warning: sl_send_reply: I won't send a reply for ACK!! >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) loose_route: Error while parsing Request URI >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) WARNING: do_action:error in expression >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) extract_aor(): Error while parsing Address of Record >>> 0(4335) lookup(): Error while extracting address of record >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) WARNING: do_action:error in expression >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) extract_aor(): Error while parsing Address of Record >>> 0(4335) lookup(): Error while extracting address of record >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> >>>How can i fix those errors? >>> Thanks for help >>> Serge >>> >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > > >

1 0

[Serusers] from_uri in the LCR table
by Eric Haskins 06 Dec '05

06 Dec '05

Can Anyone tell me is it possible to route calls based on from_uri with the LCR Module?? I see the from_uri field but I have tried using %(a)192.168.1.1 & 192.168.1.1 I replaced 192 address with an actual IP of another SER box that is Trusted. No matter what I put in there it still routes to the % only group. Even when I dial the prefix related to that entry. Eric

1 1

[Users] transfer not possible!!
by Jayesh Nambiar 06 Dec '05

06 Dec '05

hi everyone, I was just trying to transfer the call to another user. All users are registered at the same openser server. Now when I receive a call, is it not possible to transfer the call to another IP user. I thought REFER mehtod does this, but i think I am wrong. Is this possible, can somebody pls clear my understanding regarding this concept. thanks in advance!! jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

1 0

[Serusers] UNIX socket and FIFO concurrently?
by Andy Pyles 06 Dec '05

06 Dec '05

HI, is it possible to have a unix socket and FIFO at the same time? Is it the same API for both? Andy

2 3

[Serusers] SER and Music on Hold
by Waldo Rubinstein 05 Dec '05

05 Dec '05

I'm new to SER and was wondering how are people offering music on hold with it. Is it possible? Is the solution simply to send the call to Asterisk? What about SEMS? Does it offer music on hold? Thanks, Waldo

2 2

Re: [Serusers] nathelper/rtpproxy when both SIP UA are behind same NAT
by Jan Henkins 05 Dec '05

05 Dec '05

Thanks for the info! :-) If I do not succeed in getting around this hairpinning issue, I'll have to replace my Netgear gateway with one of these. On Mon, 2005-12-05 at 12:29 -0500, sip wrote: > On Mon, 05 Dec 2005 11:59:18 +0200, Jan Henkins wrote > > Hello all, > > > > On Sun, 2005-12-04 at 18:13 -0500, sip wrote: > > > > > If the gateway has no issues with hairpinning, then yes, it's quite possible. > > > I use a Linksys gateway at home and a slightly older, pre-sip-proxy version > > > of Astaro linux firewall at work, and we have multiple UAs behind each in the > > > NAT space of our firewall. They can call each other. They can all outside. All > > > based off registrations with a SER server on the outside of the network. > > > > Could you perhaps tell me the model number of your Linksys gateway? > > > > I'm using a Linksys WRT54G (not sure which series). It's got the latest > Linksys firmware on it (if that makes any difference). > > N. -- Regards, Jan Henkins

1 0

[Serusers] Blind transfer and accounting
by Bjorn Asmul 05 Dec '05

05 Dec '05

Has anyone successfully dealt with "Blind Transfer", such that the original party will be billed for the call he transfers? -- Bjorn

1 0

[Users] Serial forking Help!!
by Matteo Piazza 05 Dec '05

05 Dec '05

I become crazy to realize a serial forking. I try to use load_contacts() and next_contscts(). This is my openser.cfg nd my location table: In teory when I call 309(a)192.168.9.98 the location with the q value equal to 0.80 it is the first but it is processed only the location 309(a)192.168.9.131. Help!!! # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # 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/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/lcr.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("registrar", "append_branches", 1) #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) modparam("usrloc","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("auth_db","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("lcr","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_table","usr_preferences") modparam("tm", "fr_timer", 8) modparam("tm", "fr_inv_timer", 10) modparam("tm", "wt_timer", 5) #modparam("avpops","avp_aliases","fwdbusy=i:665") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (method =="INVITE" && uri =~"^sip:0[0-9]*@*"){ log(1, "Check 1 Start PSTN Call\n"); rewritehostport("192.168.9.97:5061"); t_relay(); exit; } if (method == "CANCEL"){ t_relay(); } if (method =="BYE"){ t_relay(); } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("create-net.it", "subscriber")) { www_challenge("create-net.it", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; if(load_contacts()){ xlog("L_ERR","LOAD CONTACTS!\n"); } xlog("L_ERR","CONTACT <$ru> !\n"); t_on_failure("1"); t_relay(); #route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (next_contacts()) { xlog("L_ERR","NEXT CONTACT <$ru> !\n"); append_branch(); t_relay(); } else { xlog("L_ERR","ERROR!\n"); } } This is my location table ============================================================ username | domain | contact | q 400 | | sip:400@192.168.9.193 | 1.00 308 | | sip:308@192.168.9.181:5060 | 1.00 309 | | sip:309@192.168.9.131;transport=udp | 1.00 309 | | sip:400@192.168.9.97 | 0.80 -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 3

[Users] SER and OpenBSD/amd64 3.8
by Federico Giannici 05 Dec '05

05 Dec '05

Yesterday I upgraded the version of the operating system of our sip server from OpenBSD amd64 3.7 to 3.8. Now I noticed that domain name lookups seems to don't work anymore and the INVITES are replyed with 478. Here is an example of the logs: Dec 5 13:11:03 eowyn OpenSER[15740]: ERROR: mk_proxy: could not resolve hostname: "voip.eutelia.it" Dec 5 13:11:03 eowyn OpenSER[15740]: ERROR: uri2proxy: bad host name in URI <sip:800900800@voip.eutelia.it:5060;user=phone> Dec 5 13:11:03 eowyn OpenSER[15740]: ERROR:tm:t_forward_nonack: failure to add branches Obviously "voip.eutelia.it" is resolved by dig on the same machine. Anybody had the same problem? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

5 7

RE: [Serusers] SER + Radius
by Ricardo Martinez 05 Dec '05

05 Dec '05

Hi again That's the way SER send the information to the RADIUS server, using Digest attributes. Regarding to the "not readable outputs" i don't know why that happen to you, in my case i use RADIATOR and i can read the Digest attributes without any problem. What RADIUS server are you using? Ricardo.- -----Mensaje original----- De: Sait Karalar [mailto:saitkaralar@gmail.com] Enviado el: Lunes, 05 de Diciembre de 2005 11:06 Para: Ricardo Martinez Asunto: Re: [Serusers] SER + Radius Hi, I read that tutorial, but when I send REGISTER request to the radius, I got some un-readable chars only. I expect SIP-Auth-Name or SER-Auth-Name, got instead Digest-Auth-Name etc... Digest-XXXX 's. and values are really not readable outputs. just some chars. so, I send this mail to get more information. :) thanks. On 12/5/05, Ricardo Martinez < rmartinez(a)redvoiss.net <mailto:rmartinez@redvoiss.net> > wrote: Hello There is a RADIUS HOW TO somwhere in the website www.iptel.org <http://www.iptel.org/> , you can authenticate SIP users via RADIUS, and do accounting also. About managing routes, maybe you can try LCR module, but it works only through a mysql local database, perhaps you can try do somethign with the AVP_RADIUS module... Good Luck Ricardo.- -----Mensaje original----- De: Sait Karalar [mailto: <mailto:saitkaralar@gmail.com> saitkaralar(a)gmail.com] Enviado el: Sábado, 03 de Diciembre de 2005 8:59 Para: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> Asunto: [Serusers] SER + Radius Hi list, I am so new to list SER. I just tried to I want to authenticate SIP users via RADIUS and want accounting with radius. also if possible, ask route to radius also. but I did not manage it ;( any simple sample? I read documents etc. no good sample found. -- gMail : google mail Try it! It is the best http://www.gmail.com <http://www.gmail.com/> -- gMail : google mail Try it! It is the best http://www.gmail.com <http://www.gmail.com>

1 0

[Serusers] SER + freeRADIUS with mysql integration
by Noel Sharpe 05 Dec '05

05 Dec '05

Hi List I want to start using RADIUS authentication, and I was wondering whether anyone has integrated the SER database (schema) with freeRADIUS. I have a web application that is using data from the the SER database, and I would if possible rather not change the web application. Has anyone here got an sql.conf file with queries adjusted for the SER schema that they are willing to share? Kind Regards Noel

1 0

[Users] unixodbc module
by Bogdan-Andrei Iancu 05 Dec '05

05 Dec '05

Hi everybody, thanks to Marco Lorrai - marco.lorrai(a)abbeynet.it - contribution, openser has new DB support module. I find it very useful as, as its own, it's a middle layer to other DB drivers For info, see http://www.unixodbc.org/ and for supported drivers see http://www.unixodbc.org/drivers.html. Marco will maintain the module - I already grant you CVS access on the project. Please feel free to work on it. The online documentation is available at: http://www.openser.org/docs/modules/1.1.x/unixodbc.html Marco, thank again! regards, bogdan

3 3

RE: [Serusers] LCR script?
by Ricardo Martinez 05 Dec '05

05 Dec '05

Hello Kyriakos. Check this thread: http://lists.iptel.org/pipermail/serusers/2005-July/022461.html Good Luck Ricardo Martinez.- > -----Mensaje original----- > De: Kyriakos Mavromichalis [mailto:kyriakos@otenettel.com] > Enviado el: Lunes, 05 de Diciembre de 2005 6:56 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] LCR script? > > > Hi, > has anyone written a script for creating the sql tables > needed for the lcr > module? > > > -- > Kyriakos Mavromichalis > Otenet Telecom > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] Question regarding UDP TCP translation
by Ken Stauffer 05 Dec '05

05 Dec '05

Did you get this working? I am trying to do the same thing. Thanks

1 0

[Serusers] 483-too many hops
by crazy 05 Dec '05

05 Dec '05

Hi all Today I found a error when I call other one by Voip phone, indicates that "483-too many hops", but I use the X-lite in computer without this error and everything is OK! What can I do for it ? in hurry!!!!!

4 3

RE: [Serusers] SER + Radius
by Ricardo Martinez 05 Dec '05

05 Dec '05

Hello There is a RADIUS HOW TO somwhere in the website www.iptel.org <http://www.iptel.org/> , you can authenticate SIP users via RADIUS, and do accounting also. About managing routes, maybe you can try LCR module, but it works only through a mysql local database, perhaps you can try do somethign with the AVP_RADIUS module... Good Luck Ricardo.- -----Mensaje original----- De: Sait Karalar [mailto:saitkaralar@gmail.com] Enviado el: Sábado, 03 de Diciembre de 2005 8:59 Para: serusers(a)lists.iptel.org Asunto: [Serusers] SER + Radius Hi list, I am so new to list SER. I just tried to I want to authenticate SIP users via RADIUS and want accounting with radius. also if possible, ask route to radius also. but I did not manage it ;( any simple sample? I read documents etc. no good sample found. -- gMail : google mail Try it! It is the best http://www.gmail.com <http://www.gmail.com>

1 0

[Serusers] T1 timer?
by Andreas Granig 05 Dec '05

05 Dec '05

Hi, I'm currently facing a situation where UAC A sends an INVITE to UAC B via SER, and SER sends two completely identical INVITEs to UAC B, the second about 50ms after the first one (no second INVITE from UAC A though). Could this be a retransmit? Seems to be a little bit too early (should be 500ms according to the rfc), and I don't think it's a parallel fork because the requests are identical regarding dst-ip/port and the message itself, and there is only one usrloc record. Any other possibilities? And another question: does SER itself retransmit requests when relaying, or is this the responsibility of the calling UAC? If so, how is the T1 timer defined in SER? I only found this one: ./modules/tm/config.h:#define RETR_T1 1 So is T1 defined as one second? Thanks, Andy

2 2

[Users] errors when making a call!
by messa＠innsof.com 05 Dec '05

05 Dec '05

Hi all When imake calls, signalisation goes well but, when, i hang up my client, i have no voice in any client! And my openser server display methese errors: 0(4335) lookup(): Error while extracting address of record 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) WARNING: do_action:error in expression 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) extract_aor(): Error while parsing Address of Record 0(4335) lookup(): Error while extracting address of record 0(4335) Warning: sl_send_reply: I won't send a reply for ACK!! 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) loose_route: Error while parsing Request URI 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) WARNING: do_action:error in expression 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) extract_aor(): Error while parsing Address of Record 0(4335) lookup(): Error while extracting address of record 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) WARNING: do_action:error in expression 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) extract_aor(): Error while parsing Address of Record 0(4335) lookup(): Error while extracting address of record 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy How can i fix those errors? Thanks for help Serge

2 1

[Serusers] Authenticate to Active Directory
by Bob McCormick 05 Dec '05

05 Dec '05

Does anyone know if there is a way to authenticate SER to MS Active Directory? I'd like to use an existing user database instead of having to manage users myself, and most of our users are in Active Directory. Has anyone tried this? What are most SER users using for authentication? Is anyone here tying into any kind of existing organization user database?

4 3

[Serusers] LCR script?
by Kyriakos Mavromichalis 05 Dec '05

05 Dec '05

Hi, has anyone written a script for creating the sql tables needed for the lcr module? -- Kyriakos Mavromichalis Otenet Telecom

1 0

[Serusers] Communication between SER as Proxy and a SIP Server
by Anthony Chapellier 05 Dec '05

05 Dec '05

Hi, I need to configure SER as a SIP proxy and that all SIP requests go through SER. To do it, you normally add SER as an outbound proxy into a routing table in your SIP server. But actually I can't modify the server... So, the solution I wish to implement is a way to register SER in SIP server as an "UAC" when it receives a REGISTER request from a real one. So first things first, is it possible ? I think you could do it by doing the following : - Save UAC real location. - Rewrite Contact Header Field in SIP packet request with SER IP adress and forward this request to SIP server. Then when you receive a request from SIP server, SER searches for UAC real location and forward request to it. So finally, how could you do it ? I tought I could use subst() function but it seems I can't master this function perfectly to make it work. So does anyone have an idea ? Thanks, Anthony

1 0

[Serusers] nathelper/rtpproxy when both SIP UA are behind same NAT
by Rafael R. GV 05 Dec '05

05 Dec '05

Hi I need to let the UAs speak directly to each other without force rtpproxy, Is there any solution for this issue? rafael ---------- Forwarded message ---------- From: Rafael J. Risco G.V. <rafael.risco(a)gmail.com> Date: Apr 22, 2005 2:17 PM Subject: UA´s behind same NAT using na thelper/rtpproxy, No Audio... To: serusers(a)lists.iptel.org I am using avpops to enable SER to see if the UAs are located behind the same address and now I need to let the UAs speak directly to each other (without force rtpproxy), please someone see my configuration below and send some advice. rafael ----. if (!lookup("location")) { log(1,"unable to locate user \n"); # handle user which was not found route(4); break; }; ### Test if UAS are in the same NAT: # get the host part of the final uri (IP part) # and store it in AVP ID 13 avp_write("$ruri/domain", "i:13"); if (avp_check("i:13","eq/$src_ip/i")) { log(1, "source IP is the same as destination IP\n"); route(3); break; }; avp_delete("i:13/g"); . . # -------------- Same NAT Call Routing (no force rtpproxy) ---- route[3]{ log(1," route[3]: UAs are in the same nat, NO force_rtp_proxy "); ### Do something!!!! if(!t_relay()){ sl_reply_error(); }; }

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[Serusers] Which all methods does SER supports????/
by nitin ganjsinghani 05 Dec '05

05 Dec '05

Hi all can anyb plz tell me that which all methods r supported by SER regards Nitin

1 0

[Serusers] SER + AD
by Andrej Sirk 05 Dec '05

05 Dec '05

Hello, somebody knows how to integrate SER with AD. Maybe with LDAP module? thanks

1 0

[Serusers] Replicate REGISTER from SER to non-SER sip server
by Matt morris 05 Dec '05

05 Dec '05

Hi all, I have this scenario: A SIP Client needs to register with both a SER server and a non-SER sip server (different domains). How can I achieve this in SER's config? I have seen this example in the list to forward the REGISTER message to other SER servers: 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; } Would it work for my scenario? I tried it but it seems the forward_tcp line gave me this error: "forward_req: ERROR: cannot forward to af 2, proto 2 no coresponding listening socket" I'm using SER 0.8.14, thanks all in advance. _________________________________________________________________ Powerful Parental Controls Let your child discover the best the Internet has to offer. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

[Serusers] Multiple Invites
by ser list 05 Dec '05

05 Dec '05

Hi, I am new to Ser, and hence I need some help on the following: 1. Someone makes a PSTN call and my cisco router forwards the call to my SER sip server. By default, my router will forward it to an account on SER, such as 1000. 2. Upon receiving the request, SER should make ALL registered phones in the same domain ring. When someone picks up the phone, other phones should stop ringing. I am able to implement point 1, but how can I make all phones ring at the same instance? I am thinking of doing something like the following for INVITES : if(uri =~"sip:1002@serproxy.com"){ rewriteuri("sip:1003@serproxy.com"); forward(uri:host, uri:port); rewriteuri("sip:1004@serproxy.com"); forward(uri:host, uri:port); } Can someone pls advise me on the best approach? Regards, zan

1 0

[Serusers] ser.cfg for proxy and registrar
by Simon HEBBO 04 Dec '05

04 Dec '05

Hi all, I am trying to install SER with mysql on a debian distribution. The installation went very good. But I am not able to configure it to work as registrar and proxy in the same time. Ser will serve a private network without any connection to the internet because i need to test some security issues on SIP. I think with a good ser.cfg file it will works. So if anyone have a ser.cfg file that enable ser to work as a proxy and registrar in the same time on a private lan, and could send it to me i will be very thankfull. Thank you all, Simon HEBBO

4 3

Re: [Serusers] UAs and SER behind NAT
by Ladislav Andel 04 Dec '05

04 Dec '05

Actaully, at the end I changed it in the source code nathelper.c and recompiled it and it works fine for me. I will look at your solution too.. Thank you, Lada Michal Matyska wrote: > I have done a little test and have found, that we have to update the doc; > the description of the tests is in opposite way... > - nat_uac_test("1") tests for private IP in contact header > - nat_uac_test("2") tests for mismatch of via IP/port and real IP/port > > Sorry for confusing. If you are getting true for the nated clients even for > nat_uac_test("2"), try to put force_rport() into your script before the > nat_uac_test... > > Michal > > On Sun, Dec 04, 2005 at 10:15:01AM +0100, Ladislav Andel wrote: > >> Thank you for your reply but nat_uac_test("1") will test src_ip >> against RFC1918 addresses anyway. >> I have tried and even seen it in nathelper.c . >> >> Ladislav >> >> > > >

1 0

[Users] Features support cfg file ?
by Norman Brandinger 04 Dec '05

04 Dec '05

Hi Paul, I've placed an example implementation online at the following location: http://openser.org/dokuwiki/doku.php?id=avp_examples Hope it helps. Regards, Norm

3 3

[Serusers] Announcement: New FTP Server with SIP related software
by Jan Janak 04 Dec '05

04 Dec '05

Hello, We have a new FTP server containing the collection of your favorite SIP and RTP related software. The FTP server is reachable either using FTP at ftp://ftp.iptel.org (use anonymous username and your email address as password), or through the web interface at http://ftp.iptel.org ftp.iptel.org is a public FTP server dedicated to freely available SIP (Session Initiation Protocol, RFC3261) based software. Here you can find everything that you will need to implement robust and highly scalable SIP based server systems. Among others, this archive contains: * SIP Express Router (SER) * RTP proxy server for NAT traversal (RTPProxy) * Web based administration interface (SERWeb) * SIMPLE based instant messaging and presence server with support for XCAP * voicemail and announcement server (SEMS) We also keep a local copy of IETF RFCs and Internet drafts. Should you have any problems, requests, or suggestions for improvements, do not hesitate to contact us by e-mail at <ftpmaster(a)iptel.org>. The contents of the FTP server will be updated and expanded regularly, it is not limited to iptel.org or SER related software. Jan.

3 4

[Serusers] UAs and SER behind NAT
by Ladislav Andel 04 Dec '05

04 Dec '05

Hello, I'm sorry for this basic question, but I have SER running on my LAN with (RFC1918) addresses with UAs on the same network. Then there are UAs registering from behind NAT to this SER on the LAN network. (SER is not listening on any public IP and it's just used in the private network) I came across this issue. If a UA on the LAN network is registering to SER the logic of ser.cfg recognize the UA is behind NAT after nat_uac_test. If I want to introduce NAT traversal with RTPproxy for this private network nat_uac_test will always return that the UA is behind NAT because it has RFC1918 IP address. The RTPproxy will be used at all times for UAs on the network or even I introduce STUN for NATed UAs I know this is not a common usage of SER because it is invented to be used on the global internet but I was just wondering is there any possibility not to use RTPproxy at all times for UAs on the private network? Maybe this is a silly question but it would help me for my testing purposes with SER where I will not have to use my setup with usage of public IPs. Thank you for any thoughts. Ladislav

3 2

[Users] Creating permenant contacts with FIFO ul_add
by Doug Meredith 04 Dec '05

04 Dec '05

I am trying to use FIFO to manage permanent contacts. I can create contacts now, but can't figure out what value to specify for the expiry. A positive value represents a real expiry in seconds. Negative numbers, "Never", and "Permanent" all result in "400 Invalid expires format". A zero results in "500 Error while adding contact". The information at http://www.openser.org/dokuwiki/doku.php?id=fifo_relay#ul_add is more of an example. It doesn't really explain the arguments. Does anyone know how to specify that a contact is permanent? The command "openserctl ul add" is able to create permanent contacts, so it must be possible. Doug -- Doug Meredith (doug.meredith(a)systemguard.com) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

2 2

[Users] FIFO response "400 ul_add: replicate expected"
by Doug Meredith 04 Dec '05

04 Dec '05

I am using OpenSER 1.0. When I send the following to the FIFO: ----------------- :ul_add:ans location 81(a)dev.skyridge.com sip:34@sirloin.skyridge.com 3600 1 ----------------------- I get the response "400 ul_add: replicate expected". I have done a google search without finding anything helpful. Can anyone explain what this means? Doug -- Doug Meredith (doug.meredith(a)systemguard.com) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

2 2

[Users] dispatcher relaying instead of dipatching?
by Rick Richardson 03 Dec '05

03 Dec '05

On my test network I have set up a dispatcher server that points to two proxy servers that hit an external database. I have a simple route script on the dispatcher that should be routing requests to the two proxy servers. dispatcher .51 caller .99 recipient .100 proxy1 .52 proxy2 .53 I'm making call requests from .99 to .100, with .51 used as the proxy server. The problem I'm getting, is that the INVITE is being sent directly to the recipient without passing through the proxys, the dispatcher is passing the message directly. At least i think that is what is happening. Here is the sip INVITE: INVITE sip:rick@192.168.0.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.99:5060;rport;branch=z9hG4bK837723126 From: <sip:rick@192.168.0.51>;tag=679508646 To: <sip:rick@192.168.0.100> Call-ID: 125992616(a)192.168.0.99 CSeq: 20 INVITE Contact: <sip:rick@192.168.0.99:5060> Max-Forwards: 5 User-Agent: Linphone-1.0.1/eXosip Subject: Phone call Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 352 So it seems to me that the invite request is being sent to .51 and it is forwarding directly to .100 without dispatching the message to .52 or .53 here is my dispatcher config: # $Id: dispatcher.cfg,v 1.1.1.1 2005/06/13 16:47:37 bogdan_iancu Exp $ # sample config file for dispatcher module # debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 fifo="/tmp/openser_fifo" # for more info: sip_router -h # ------------------ module loading ---------------------------------- loadmodule "//lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("2", "2"); forward(uri:host,uri:port); # t_relay(); }

1 0

[Serusers] Failure route weirdness....
by sip 03 Dec '05

03 Dec '05

Okay... I'm a little stumped on what's going on here, so I'll start with a general explanation before I bother posting my (likely hard to read) ser.cfg. I'm having a weird issue when calling from X-Lite to a SNOM 320 phone (I haven't tried the SNOM 190 with a different firmware yet, but I don't see that it would make much difference). I have a failure route which handles voicemail and such, and I have the invite timer set to 27 seconds (roughly 4.5 standard US rings). Now, when I call from X-Lite to X-Lite, or from Leadtek 8051 to X-Lite, or from SNOM to X-lite, or from SNOM to SNOM, or from Leadtek to Leadtek, etc, etc... everything works EXACTLY as it's supposed to. I have log messages telling me when the failure route is being entered, and it hits it at right on 27 seconds. However, when I call TO the SNOM phone from any other device besides another SNOM phone, I never hit the failure route until a 408 timeout at 58 seconds. Now, I can't think of a reason for a UA to overwrite the invite timer on SER or even how such overall weirdness would occur, but it's 58 seconds every time like clockwork.. error 408. At that point, it enters the failure route and does its processing (forwarding the call to voicemail as far as SER is concerned but not as far as the UA is concerned... which just receives an error message). Any ideas? Also... when I dial SNOM to SNOM, the SIP method I get in the ACC is PRACK (no INVITE). Can someone fill me in on what a PRACK is? Is there something I should be looking at to figure out why calling one UA behaves differently than other UAs? N.

2 2

[Serusers] SER + Radius
by Sait Karalar 03 Dec '05

03 Dec '05

Hi list, I am so new to list SER. I just tried to I want to authenticate SIP users via RADIUS and want accounting with radius. also if possible, ask route to radius also. but I did not manage it ;( any simple sample? I read documents etc. no good sample found. -- gMail : google mail Try it! It is the best http://www.gmail.com

1 0

[Users] Features support cfg file ?
by Paul van Schagen 03 Dec '05

03 Dec '05

Hello, I have setup openser and I have a request to incorporate an unconditional forward feature which can be turned on by dialing a "*21+ number" and can be turned of by dialing #21#. Can anyone provide me with an example piece of .cfg file that can do this ? best regards Paul

1 0

[Users] Handeling Diversion - expired?
by Helge Waastad 03 Dec '05

03 Dec '05

Hi, I have a question more in a general aspect. Since the draft-levy-sip-diversion-08 has expired, should a Proxy strip such headers? According to RFC3261,the UAS MUST ignore headers-fields not defined in the specification or any supported extension... I've realized that some systems need Diversion headers for voicemail (I know Asterisk doesn't, but Cisco Unity does) Any comments? -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 1

[Serusers] STUN support
by Jan Henkins 02 Dec '05

02 Dec '05

Hello all, Is there anything that needs to be done to the ser cfg file to support STUN? If so, could you please share that config snippet? Thanks in advance. -- Regards, Jan Henkins Cell: +27-84-951-4334

2 2

[Serusers] NAT transversal
by Voipers Portugal 02 Dec '05

02 Dec '05

---------- Forwarded message ---------- From: Voipers Portugal <voipers(a)gmail.com> Date: Dec 2, 2005 9:08 AM Subject: Re: [Serusers] NAT transversal To: jan(a)henkins.za.net I think you should start by reading the nathelper module. There you'll find the information that you will need. If you want, try the OnSip config file for NAT, it's quite self explaning, and it works. Jose Simoes On 12/2/05, Jan Henkins <jan(a)henkins.za.net> wrote: > > Hi, > > I'm trying to set up ser (version 0.9.4) on a server that is not behind > any form of NAT. I need to get to this ser server from behind NAT > firewalls (ADSL). To test I'm using a variety of software (linphone, > Twinkle etc.), and I also have a Grandstream Budgetone 100 that I'm > using. I tested the Grandstream phone successfully on my FWD account > with STUN, however try as I might I can not log on to my server. What is > the best way to accommodate this type of setup? At the moment I need a > simple nudge into a direction so that I can research it properly. > > Thanks in advance! > > -- > Regards, > Jan Henkins > Cell: +27-84-951-4334 > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Users] DB Integration
by Douglas Garstang 02 Dec '05

02 Dec '05

All, Thinking of modifying the OpenSER scripts/mysqldb.sh script, and adding some new tables. One for example would be table that Asterisk uses to log CDR's for calls. It can use it's own database instances, but it seems to make more sense to put it all into one database. Would also like to add some new columns to existig tables, such as subscriber. Does anyone have any comments as to why this would be a bad idea? I can't think of any reason why it would cause any problems for OpenSER if there where additional fields or tables. Thanks, Doug

2 1

[Serusers] Planning SER + RADIUS + mySQL
by Paco Orozco 02 Dec '05

02 Dec '05

Hello, I'm using SIP in order to be able to call to my PBX extensions. I've configured SER + asterisk and It's working well, but I've got some troubles with authentication: I want to be able to autehnticate with RADIUS in an LDAP (It's working) And I'd like to manage groups with mySQL tables (provided by SER package). But... this is my problem... How can create users in mySQL, It's obligatory to register (via web, for example) to create records is mySQL table? I've visit this homepage: http://www.ethworld.ethz.ch/technologies/sipeth/notes and I've saw some modules very useful, but w/o documentation. Has anyone used it? Thanks in advance -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

1 0

[Serusers] SER Stress Test & Sipsak
by Ramin Nikaeen 02 Dec '05

02 Dec '05

Friends, Does anyone know of any other tool for stress-testing the SER other than Sipsak? Is sipsak good?! Thanks ramin

4 3

[Serusers] using subst function from textops module
by Anthony Chapellier 02 Dec '05

02 Dec '05

Hi, I wish to use subst function and looked into module textops documentation to know how to use it. The documentation provides an example which is using a variable or flag named \u. Looks like this '\u' is a replacement for the current uri of the SIP request. # replace the uri in to: with the message uri (just an example) if ( subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {}; So, I would like to know if there are more variables like this one that can be used with subst ? And are there variables too for log function that can be used the same way ? Thanks,

1 0

[Serusers] SER Comment
by Selvasundar 02 Dec '05

02 Dec '05

hi andres , i request your help to install SIP SER as a proxy . THanks selva The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in

2 1

[Users] Crash (core)
by Lars Sundqvist 02 Dec '05

02 Dec '05

I have switched from SER to OpenSER in our production enviroment since I need a few of the "new" modules, I am however having troubles with OpenSER hanging and crashing, so much so, that I had to change back. There is no clues in the logs to why this is happening but i have a coredump from a crash: (gdb) where #0 0x0052f7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x005707d5 in raise () from /lib/tls/libc.so.6 #2 0x00572149 in abort () from /lib/tls/libc.so.6 #3 0x08054863 in del_lump (msg=0x0, offset=186, len=-135181242, type=HDR_OTHER_T) at data_lump.c:288 #4 0x00f8bd20 in replace_from (msg=0x8130d98, from_dsp=0xbff977a0, from_uri=0x0) at from.c:207 #5 0x00f8e57b in w_replace_from2 (msg=0x8130d98, dsp=0x812f700 "\030W\022\b\v", uri=0x0) at uac.c:311 #6 0x08051703 in do_action (a=0x8126658, msg=0x8130d98) at action.c:677 #7 0x080529e3 in do_action (a=0x8126680, msg=0x8130d98) at action.c:773 #8 0x0805297b in do_action (a=0x8126a58, msg=0x8130d98) at action.c:773 #9 0x0805297b in do_action (a=0x8126f80, msg=0x8130d98) at action.c:773 #10 0x0805297b in do_action (a=0x8127058, msg=0x8130d98) at action.c:773 #11 0x0805297b in do_action (a=0x812b6d0, msg=0x8130d98) at action.c:773 #12 0x080529e3 in do_action (a=0x812b6f8, msg=0x8130d98) at action.c:773 #13 0x080529e3 in do_action (a=0x812b720, msg=0x8130d98) at action.c:773 #14 0x0805297b in do_action (a=0x812b7f8, msg=0x8130d98) at action.c:773 #15 0x08052b42 in run_actions (a=0x8122da0, msg=0x8130d98) at action.c:773 #16 0x080753e9 in receive_msg ( buf=0x80eb300 "INVITE sip:987654321@xxx.xxx.xxx SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK0ffd9a6274aeb347617736a6da35e918\r\nTo: 987654321<sip:987654321@xxx.xxx.xxx>\r\nFrom: 123456789<si"..., len=943, rcv_info=0xbff98b80) at receive.c:160 #17 0x0808dd1c in udp_rcv_loop () at udp_server.c:464 #18 0x08064e7f in main_loop () at main.c:1033 #19 0x08065615 in main (argc=3, argv=0xbff98d34) at main.c:1570 (Numbers and sip domain edited) Anyone who has a clue whats going wrong? Lars Sundqvist

2 1

[Serusers] Does SER support SIP SUBSCRIBE/NOTIFY methods
by Vivek 02 Dec '05

02 Dec '05

Hi All, Myself Vivek n i m working as a Q A engineer at VeriSign Inc., India. Due to some requirement I am planning use SER but before that i just want to assure whether it supports SUBSCRIBE/NOTIFY and other extended SIP methods. Please help me by providing the correct information. Thanks, Vivek --------------------------------- Yahoo! Shopping Find Great Deals on Gifts at Yahoo! Shopping

2 1

[Serusers] MYSQL
by Colin Jordan 02 Dec '05

02 Dec '05

Every time I activate the command below ser does not start. Please help. fifo_db_url="mysql://ser:heslo@localhost/ser" DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

2 1

[Serusers] SER performance degrades dramatically when invalid users send REGISTER messages continuously
by tulay sazak 02 Dec '05

02 Dec '05

Hello everybody, I am using SER 0.9.3 and freeRADIUS 1.0.4 with digest authentication. I am having problems when users with invalid credentials (wrong username or password) try to register to SER. The same thing happens when I insert some records to usergroup table as Reject type in Radius DB. SER is becoming unfunctional in this case in our system. It cannot perform registration and cannot make call for even valid subscribers. SIP clients with invalid credentials try to register and send REGISTER messages continuously. A SIP client sends 20 request/min and there are 7-8 clients available at the moment. i.e. it is not a big traffic. But every client is tried to be authenticated using digest during registration and at the beginning of the call. In such a case, SER slows down very much. Retransmissions starts and SER becomes unfunctional. Did anyone experienced this problem? Do you have any idea why does it happens? Thanks in advance.. Regards, Tulay Sazak NGN, Borusan Telekom --------------------------------- Yahoo! DSL Something to write home about. Just $16.99/mo. or less

4 3

[Serusers] Multiple PSTN routes
by Kyriakos Mavromichalis 02 Dec '05

02 Dec '05

Hi, i have been doing some searching for how to route to a standby pstn gateway when primary pstn gateway fails. From what i gathered we can either have a failure route that sends the call to the second pstn gateway after an unsuccessful attempt of relaying to the primary gateway, or do a parallel forking to both and the gateway that is active responds back. Which is the best method ? Are there any resources i can read for setting this up? -- Kyriakos Mavromichalis Otenet Telecom

2 2

[Users] mediaproxy problem
by Klaus Darilion 02 Dec '05

02 Dec '05

Hi! I'm having problems with the mediaproxy which I can not resolve. Sometimes, when the session is terminated, mediaproxy does really terminate the session and the ports keep allocated. At the moment I'm having 400 open sessions and mediaproxy uses 80% of the CPU. Attached you can see the logfile of mediaproxy (to avoid line breaks). The first call does not terminate correctly and the ports are still opened. The second call terminates correctly. Can someone help me digging into this problem? regards klaus Dec 2 10:30:49 philipp mediaproxy[440]: command request 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167 1.2.32.167:19514:audio 1.2.32.167 1.2.32.167 remote 3.4.240.143 remote Cisco-SIPGateway/IOS-12.x info=from:06503836629@1.2.32.167,to:43159966509977@1.2.32.160,fromtag:16588354-2D7,totag: Dec 2 10:30:49 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: started. listening on 1.2.32.83:36888 Dec 2 10:30:49 philipp mediaproxy[440]: command execution time: 1.33 ms Dec 2 10:30:57 philipp mediaproxy[440]: command lookup 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167 3.4.240.143:16434:audio 3.4.240.143 1.2.32.167 remote 1.2.32.160 unknown Linksys/PAP2-2.0.13(LSb) info=from:06503836629@1.2.32.167,to:43159966509977@1.2.32.160,fromtag:16588354-2D7,totag:df301db23bf537dbi0 Dec 2 10:30:57 philipp mediaproxy[440]: command execution time: 0.73 ms Dec 2 10:30:57 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: called signed in from 3.4.240.143:16434 (RTP) (will return to 3.4.240.143:16434) Dec 2 10:30:57 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:19514 (RTP) (will return to 1.2.32.167:19514) Dec 2 10:31:01 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:19515 (RTCP) (will return to 1.2.32.167:19515) Dec 2 10:33:22 philipp mediaproxy[440]: command delete BF809954A357C8D45FCF51D81299(a)84.151.154.100 info= Dec 2 10:33:22 philipp mediaproxy[440]: command execution time: 0.27 ms Dec 2 10:36:54 philipp mediaproxy[440]: command request 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 1.2.32.167:18412:audio 1.2.32.167 1.2.32.167 remote 1.2.33.19 remote Cisco-SIPGateway/IOS-12.x info=from:069911160036@1.2.32.167,to:4359966366102@1.2.32.160,fromtag:165E1650-2D0,totag: Dec 2 10:36:54 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: started. listening on 1.2.32.83:36894 Dec 2 10:36:54 philipp mediaproxy[440]: command execution time: 1.52 ms Dec 2 10:36:57 philipp mediaproxy[440]: command lookup 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 1.2.33.19:32240:audio 1.2.33.19 1.2.32.167 remote 1.2.32.160 unknown CSCO/6 info=from:069911160036@1.2.32.167,to:4359966366102@1.2.32.160,fromtag:165E1650-2D0,totag:000dedfb04cc2d723b9fccb6-64a00b22 Dec 2 10:36:57 philipp mediaproxy[440]: command execution time: 0.60 ms Dec 2 10:36:58 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:18412 (RTP) (will return to 1.2.32.167:18412) Dec 2 10:36:58 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: called signed in from 1.2.33.19:32240 (RTP) (will return to 1.2.33.19:32240) Dec 2 10:37:01 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:18413 (RTCP) (will return to 1.2.32.167:18413) Dec 2 10:37:08 philipp mediaproxy[440]: command delete 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 info= Dec 2 10:37:08 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: 510/505/1015 packets, 30800/30300/61100 bytes (caller/called/relayed) Dec 2 10:37:08 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: ended. Dec 2 10:37:08 philipp mediaproxy[440]: command execution time: 0.55 ms

1 0

[Serusers] SER questions
by Andrew Gutierrez 02 Dec '05

02 Dec '05

We are evaluating a hosted VOIP platform from a company called Lignup ( www.lignup.com <http://www.lignup.com/> ). They tell us we need a Session Border Controller to work as an intermidaetary between the customers and the VOIP PBX and the PSTN. They use or recommend NexTone. Will or does SER perform the same functions. Business Class Internet Services Andy Gutierrez Phoenix Internet <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=2922+W.+Clarendon+Ave.&csz=P hoenix%2C+AZ+85017&country=us> 2922 W. Clarendon Ave. Phoenix, AZ 85017 <mailto:andy@phxinternet.net> andy(a)phxinternet.net IM: andyznetandyznet <http://www.phoenixinternet.net> www.phoenixinternet.net tel: fax: 602-234-0917 x 115 602-234-0699 <https://www.plaxo.com/add_me?u=21475002481&v0=312722&k0=883971886> Add me to your address book... <http://www.plaxo.com/signature> Want a signature like this?

3 2

[Users] (open)SER and ICE
by Christophe Irles 02 Dec '05

02 Dec '05

Hi, As I'm a newbie on this subject, I'm wondering how two SIP devices with ICE compatibility will interact with a (open)SER+Mediaproxy implementation since ICE implementation could permit to avoid RTP flow on the (open)SER server. Is there any update to do on the conf file on the (open)SER ? Since they are ICE compatible, the SIP flow (signalization) will be analysed by this two devices in order to create a direct RTP flow ? Or is it because i know on the (open)SER side that this devices are ICE compatible I send a special SIP message ? In fact I want to know if a server update is necessary or if it's "free" to have this great feature. :) Thanks, Christophe

7 7

[Users] a router error
by messa＠innsof.com 02 Dec '05

02 Dec '05

Hi all I use d-link 704P router. When i want to make a call, signalisation is good, but, when the client is hang up, i have voice to the client which is connect in the D-link LAN and the other client has no voice. My server has a public IP. When i'm find the log of the router, i see these messages: unallowed access from 192.168.0.53:to 66.249.93.104:protocol=6 rule=3(by firewall) 192.168.0.53, is the adress of the client behind the d-link lan, the client which has voice and cannot transmit it! what means theses messages? how can i solve the router problem to make my calls? please help best regards! Serge

2 1

[Users] OpenSER attempting local mysql connection
by Rick Richardson 02 Dec '05

02 Dec '05

I have openser.cgf pointing openser to use a remote database using this line: modparam("auth_db", "db_url", "mysql://serusr:mypass@192.168.0.54/ser") Upon initializing the auth_db module it seems to be insisting on connecting locally (which fails) I compiled all the modules for mysql5 using the devel libs from the mysql official redhat enterprise rpms. I can add/rem users to this remote database using openserctl. When I start openser I get this output: 0(3076) WARNING: fix_socket_list: could not rev. resolve 192.168.0.52 0(3076) WARNING: fix_socket_list: could not rev. resolve 192.168.0.52 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.0.52 [192.168.0.52]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.0.52 [192.168.0.52]:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: amnesiac:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: amnesiac:5060 stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) new_connection: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) 0(0) register_udomain(): Can not open database connection 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration any ideas? Rick

2 4

[Serusers] NAT transversal
by Jan Henkins 02 Dec '05

02 Dec '05

Hi, I'm trying to set up ser (version 0.9.4) on a server that is not behind any form of NAT. I need to get to this ser server from behind NAT firewalls (ADSL). To test I'm using a variety of software (linphone, Twinkle etc.), and I also have a Grandstream Budgetone 100 that I'm using. I tested the Grandstream phone successfully on my FWD account with STUN, however try as I might I can not log on to my server. What is the best way to accommodate this type of setup? At the moment I need a simple nudge into a direction so that I can research it properly. Thanks in advance! -- Regards, Jan Henkins Cell: +27-84-951-4334

1 0

[Users] Can anyone share his openser.cfg..in which openser is routing calls to pstn gateway
by Lokesh Kumar 01 Dec '05

01 Dec '05

Hello, I am trying to configure pstn gateway with openser.when i start openser with default openser file it works fine, but i one direction. i can dial on my SIP user agent from outside. but when i am using pstn.cfg file as a openser.cfg(after rename). I am unable to even start openser server. it gives me errors and crashed.I puts my pstn gateway ip's in that file where requested. Please please help Regards

4 6

[Users] AVPs are lost on relayed INVITE errors
by Federico Giannici 01 Dec '05

01 Dec '05

I'm using OpenSER 1.0.0 on OpenBSD 3.7 amd64. I have a strange problem with the accounting: I set a couple of AVPs for every message that arrives at the server. I'm sure they are there because they are written in the syslog logging. Sometimes, when an INVITE is relayed (with transactions) and receives an error (488, 422, etc.), in the SQL logging there is no more presence of the AVPs! Is this a known problem? How can I avoid this? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

4 20

[Users] Re: [Devel] nicer compiling
by Bogdan-Andrei Iancu 01 Dec '05

01 Dec '05

Hi, thanks to Anand, you have now the liberty to get a more friendly output during compilation. Just do: NICER=1 make all normal "make all" still produces same amount of output - it;s the default since it's useful to see the compilation flags...at least for me... also I got rid of the messages regarding the missing dependency files (.d) during first compilation.... so, have a nicer compiling ;) regards, bogdan Anand Kumria wrote: >Hi, > >Attached is a diff that makes it nicer to compile openser. Should you >want to see the exact compilation sequence Q="" make <...> will show it >to you. > >Thanks, >Anand >

2 2

[Users] Serial fork with AVPOPS
by Matteo Piazza 01 Dec '05

01 Dec '05

Hi, I want to realize a serial fork. With the function avp_db_load() I load the addresses to forward the call and the function avp_pushto() rewrites the address. I have this problem (see the debug output): I try to call 309(a)192.168.9.97 309 does not answer Go to failure_route to write 400(a)192.168.9.97 The problem is that it comes executed another time avp_db_load(), and I don't understand why the voicemail is executed two times. Thank's for all! 7(30331) AVP found for r-uri <sip:309@192.168.9.131;transport=udp> for Call forward on Busy 14(30345) AVP inserted: r-uri <sip:400@192.168.9.97> for Call forward on Busy 6(30329) AVP found for r-uri <sip:400@192.168.9.193> for Call forward on Busy 14(30345) R-uri <sip:309@192.168.9.97> for VoiceMail 14(30345) ERROR:tm:t_forward_nonack: no branch for forwarding 14(30345) ERROR: w_t_relay_to: t_relay_to failed 14(30345) R-uri <sip:309@192.168.9.97> for VoiceMail 14(30345) ERROR:tm:t_forward_nonack: no branch for forwarding 14(30345) ERROR: w_t_relay_to: t_relay_to failed 9(30335) AVP found for r-uri <sip:400@192.168.9.193> for Call forward on Busy 9(30335) ERROR: t_newtran: transaction already in process 0xb5da3d88 6(30329) ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 9(30335) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) 6(30329) ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 route{ .................. if (avp_db_load("$ruri","")) { xlog ("L_ERR", "AVP found for r-uri <$ru> for Call forward on Busy\n"); } t_on_failure("1"); .......... } failure_route[1] { if (avp_pushto("$ruri", "s:fwdbusy")) { xlog ("L_ERR", "AVP inserted: r-uri <$ru> for Call forward on Busy\n"); avp_delete("s:fwdbusy"); append_branch(); t_relay(); t_on_failure("1"); } else{ avp_pushto("$ruri","s:voicemail"); xlog ("L_ERR", "R-uri <$ru> for VoiceMail\n"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay_to_udp("192.168.9.97","5061"); t_relay(); } ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 4

[Serusers] Creating CDRs from SERs accouning records
by Frank Fischer 01 Dec '05

01 Dec '05

Hi all i'm looking for a free/opensource application that generates CDRs (not XDRs, must not be rated) from the accounting data/events generated by SER (i think this is usually called normalization). I think this would have to be an application that is able to match INVITEs and BYEs and form CDRs with call start timestamp, call duration and so on. I had a look at the list of accounting software on the iptel.org website, but most of there applications are highly integrated accounting/billing solutions, far more than what i need (and also most of them seem to be commercial products). In the mailinglist archive i only found some references to CDRTool which seems also to be commercial. Obv. there must be a free version of CDRTool but neither i'm sure whether it implements exactly what i need so i could "grab it" for my own purposes nor could i find a place to download the free version). Does anyone know if there is such a kind of application available? Or any other idea how to deal with my requirements? Thanks a lot for your help Frank

9 18

[Users] hi all
by hemant kumar 01 Dec '05

01 Dec '05

hello list, i am new to openser. i am trying to setup openser and asterisk together where asterisk will work as a voicemail server. i am trying to find a sample config which can help me understand how to route the calls to asterisk when user is unavailable, busy or time out after sometime. thanks, hemant

2 1

[Serusers] assertion failed in t_retransmit_reply; race condition in t_newtran?
by Valentin Nechayev 01 Dec '05

01 Dec '05

Hi, we had got series of SER cores with failed assertion in t_retransmit_reply(): assert(t->uas.response.dst.send_sock); Essential backtrace part: #0 0x48167fc4 in kill () from /usr/lib/libc.so.4 #1 0x481a993e in abort () from /usr/lib/libc.so.4 #2 0x481858d3 in __assert () from /usr/lib/libc.so.4 #3 0x48281da3 in t_retransmit_reply () from /usr/local/lib/ser/modules/tm.so #4 0x48277433 in t_newtran () from /usr/local/lib/ser/modules/tm.so This place from ser.cfg: if (!t_newtran()) { sl_send_reply("500", "could not create transaction"); break; }; response.dst.send_sock is filled in init_rb(), which is called from t_newtran(). t_newtran() has the following: === cut === UNLOCK_HASH(p_msg->hash_index); /* now, when the transaction state exists, check if there is a meaningful Via and calculate it; better do it now than later: state is established so that subsequent retransmissions will be absorbed and will not possibly block during Via DNS resolution; doing it later would only burn more CPU as if there is an error, we cannot relay later whatever comes out of the the transaction */ if (!init_rb( &T->uas.response, p_msg)) { LOG(L_ERR, "ERROR: t_newtran: unresolvable via1\n"); put_on_wait( T ); t_unref(p_msg); return E_BAD_VIA; } return 1; === end cut === Does there race condition between UNLOCK_HASH(), init_rb() and parallel t_newtran() from another SER process exist? Details: FreeBSD 4.10, SER 0.9.3 (with PortaOne additions, but not essential for the question). -- Valentin Nechayev PortaOne Inc., Software Engineer mailto:netch@portaone.com

1 0

[Users] Serial forking
by Bogdan-Andrei Iancu 01 Dec '05

01 Dec '05

Hi, I just committed serial proper forking support into core - it was migrated from LCR module. I mean proper, since it has q value support and it can be used by any module without any inter-module dependencies. The idea behind was to allow to all module that performs parallel forking to do also serial forking - exec, enum, registrar, etc. There are two new script functions : *serialize_branches(n)* : it inherits the functionality of load_contacts() from LCR; gets all parallel branches and convert them into AVPs for serial forking; numerical parameter 'n' says if any previous AVP should be removed (if non-0) or not (if 0). Returns true is no error (even if no serialization happened). *next_branches()* : it inherits the functionality of next_contacts() from LCR; get (based on q value) the next contact(s) to be used in sequential forking. Returns true only if a new contact was got to be used. The AVP containing the branches is accessible only via alias - its ID is not configurable or visible; the alias (automatically exported by core) is "serial_branch" - it is visible from any module that uses the avp core aliasing system. here are some examples of how to uses this new functionality: 1) enum doing serial forking: { ..... enum_query("e164.arpa.","voice"); serialize_branches(1); if (!next_branches()) { sl_send_reply(...no enum found...); exit; } t_on_failure("1"); t_relay(); } failure_route[1] { if (next_branches()) { t_relay(); } } 2) same for registrar, just replace enum_query() with lookup("location"). NOTE that this will not work yet if there are nated contacts: there is no support to save also the flags and the dst_uri in order to perform proper nat traversal. The code was not yet heavily tested, so any help on this is highly appreciated. On the TO-DO list, related to this subject are: 1) SRV lookup to use this functionality to provide any additional entries. 2) full support for registrar for nat traversal 3) remove load/next_contacts() from LCR (once this new functions prove to be stable). regards, bogdan

4 5

RE: [Users] Sending Authorisation Credentials
by Douglas Garstang 01 Dec '05

01 Dec '05

I don't think this will work. That means not only would the Asterisk box need to let REGISTER requests through without authentication, it would also need to let every other type of request through (INVITES come from OpenSER as well). You can't drill down to that level of detail in Asterisk and tell it to not authenticate for REGISTERs from OpenSER, but to authenticate for INVITE requests. Doug -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, November 30, 2005 2:54 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Sending Authorisation Credentials configure asterisk to accept REGISTER without authentication, if it is received from your proxy (IP address) klaus Douglas Garstang wrote: > I have user agents registering with OpenSER. I wanted to 'copy' the registration to an Asterisk server, so I modified a standard openser.cfg as such: > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest authentication > if (!www_authorize("voip.com", "subscriber")) { > www_challenge("voip.com", "0"); > exit; > }; > > save("location"); > t_replicate("192.168.10.7","5060"); # Added this line... > exit; > }; > > I just stuck the t_replicate() call in there. OpenSER sends the REGISTER packet to the Asterisk box who requests credentials. How can I get OpenSER to send credentials (ie act as a UAC) for a given user, to a UAS? > > Thanks, > Doug > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

3 2

[Serusers] Ser + mysql auth
by Paco Orozco 01 Dec '05

01 Dec '05

Hiya, I'm trying to configure ser in order to authenticate to mysql database. I've installed ser from debian package version: ser 0.9.3 (i386/linux) I've look for help into Google but I can't find any valuable help. This is the error message: 0(0) DEBUG: init_mod: auth 0(0) auth module - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(0) DEBUG: init_mod: auth_db 0(0) auth_db module - initializing 0(0) find_mod_export: <db_use_table> in module sql not found 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: auth_db_bind: unable to bind to the database module 0(0) init_mod(): Error while initializing module auth_db Can anyone help me? -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

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[Serusers] Mediaproxy question
by George Lambson 01 Dec '05

01 Dec '05

Anyone know what force_rport() does? or does anyone have additional info on the exported functions for this module?

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[Serusers] SER as a Proxy
by Selvasundar 01 Dec '05

01 Dec '05

hi , this is john, i need to write a TESt Stub in TCL for P/S/I CSCFs , need to install as a Proxy for the test setup. do not find simple guidelines to install the same Thanks & Regards john The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in

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[Serusers] Adding a user
by Xavier TRENTIN 01 Dec '05

01 Dec '05

Hi all, I use ser with dbtext. I have two questions. 1) I 've added many fields in my subscriber table, and I'd like to know if all the table is loaded in memory when ser starts or if it only loads the fields it needs? 2)When I add users in my subscriber table, I must restart ser to load the new accounts. I thought there were a specific serctl which worked with dbtext, but I can't find it. Can any one give me a link where I can dowload it, and does it allow to insert new users in the text table and in the in loaded table at the same time? If not, how can I add my new users directly in memory? Thanks in advance. B.R. Xavier.

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FW: [Serusers] Repeated Register Message
by Pravesh 01 Dec '05

01 Dec '05

Hi Chan, I also faced a similar problem in which when UA sends REGISTER to SER from different IP address than what is present in the contact header of the REGISTER message, then SER actually stores two AORs for the UA, i.e. one AOR is the actual contact header value & other is the same username with the IP address & port from which I sent the REGISTER message. I am not sure for the reason of this kind of behavior. Anyone from the SER developer list should be able to answer this query. So also forwarding this message to developers list also. Regards, Pravesh -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of tcchan Sent: Wednesday, November 30, 2005 8:57 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Repeated Register Message Hi All, I encountered a situation that somehow a UA keep on Registering with the server with the same IP but on different port. So SER, thought that there are multiple instances of the same username and save the various location into the database. The problem is it registers so often that the memory is full not very long. I suspect it has something to do firewall but I am not sure. Has anyone encounter this problem before? Any hint?? For the time being, is there a way to limit the number of REGISTER message that the SER will take? Regards, TC Chan _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] can't connect to RTP proxy
by lo ™ 01 Dec '05

01 Dec '05

hi, i read the onsip.org <http://onsip.org> getting started manual. i tried again with ser and rtpproxy but i received same errors... then i decided to try with mediaproxy: i downloaded ser-0.9.0 (emerged from gentoo portage) and mediaproxy-1.4.2. i followed this instructions: http://www.voip-info.org/wiki-MediaProxy but i cannot even start it (neither --help works) $ /etc/init.d/mediaproxy start : bad interpreter: No such file or directory and: $ /usr/local/mediaproxy/mediaproxy.py : No such file or directory i got ensured that EOL are not dos-style and i tried that a simple file called test.py (it just prints something) just to make sure that python works... sorry, i understand it is not properly a ser problem but i don't know where else to ask... thank you so much!

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[Serusers] Any mailing list about using SER?
by lin9k＠yahoo.com 01 Dec '05

01 Dec '05

Hi, I have encountered a problem while trying to use Radius with ser. Whenever I try to load the module "auth_radius.so" the ser couldn't start up and print the error message "ERROR while initialzing module auth_radius". I followed the procedure very carefully in http://www.iptel.org/ser/doc/ser_radius/ser_radius.html but this still happen. I don't know who to ask for. Is there any mailing list, its archive, or anyone that I can ask the question? Thanks, lin __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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Re: [Serusers] domain trouble
by George Lambson 30 Nov '05

30 Nov '05

Nevermind, I was putting the username only in the authorization user and it needs the username and the domain ie. bob(a)foobar.com. Thanks >>> Klaus Darilion <klaus.mailinglists(a)pernau.at> 11/29/2005 1:56 AM >>> The SIP_DOMAIN is only used for serctl as default domain (when you create a new user without specifiying the domain of the SIP AoR). You can specify the domain for which ser is responsibly (== myself) by using the alias=... parameter in ser.cfg, e.g.: alias=foo.com bar.com iptel.org mysipproxy.com Then you can verify using e.g. if (uri == myself) ... Another approach is using the domain module. Then the domains are stored in a DB table and you can add/delete domains without restarting ser. Then you have to use the functions provided my the domain module, e.g. if (is_uri_host_local()) ... regards klaus George Lambson wrote: > For some reason my SER isn't using the sip domain I specify (export SIP_DOMAIN="whatever.edu"). When my UA registers it will only register if I take out all of the digect authentication from my config file (I leave all of them blank instead of putting in a domain) and then use the host name of my server instead of a domain. > > Any ideas what I am doing wrong? > > Thanks, > George > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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Re: [Serusers] domain trouble
by George Lambson 30 Nov '05

30 Nov '05

Okay, that makes a lot of sense. But... I did it and changed the realm accordingly in my UA and now it won't authenticate with either realm. What now? >>> Klaus Darilion <klaus.mailinglists(a)pernau.at> 11/29/2005 1:56 AM >>> The SIP_DOMAIN is only used for serctl as default domain (when you create a new user without specifiying the domain of the SIP AoR). You can specify the domain for which ser is responsibly (== myself) by using the alias=... parameter in ser.cfg, e.g.: alias=foo.com bar.com iptel.org mysipproxy.com Then you can verify using e.g. if (uri == myself) ... Another approach is using the domain module. Then the domains are stored in a DB table and you can add/delete domains without restarting ser. Then you have to use the functions provided my the domain module, e.g. if (is_uri_host_local()) ... regards klaus George Lambson wrote: > For some reason my SER isn't using the sip domain I specify (export SIP_DOMAIN="whatever.edu"). When my UA registers it will only register if I take out all of the digect authentication from my config file (I leave all of them blank instead of putting in a domain) and then use the host name of my server instead of a domain. > > Any ideas what I am doing wrong? > > Thanks, > George > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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