- sr-users - kamailio.org

RE: [Serusers] need help
by Rodrigue ESTEVE 22 Sep '05

22 Sep '05

Hi Julien, Can you do this in command line on your linux : Netstat -an And check that your ser is listening on udp port 5060. rodrigue ________________________________ De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de julien bossart Envoyé : mercredi 21 septembre 2005 12:02 À : serusers(a)lists.iptel.org Objet : [Serusers] need help Hi all, I am new to this forum. Today, i download ser package, then i install in my fedora core 4. Everything is ok. Next, i start ser and use serctl add to add account to ser server. Then i use sjphone to connect to ser server. Sjphone try to send REGISTER to ser server. I use ethereal to get the package. I see that ser server responds ICMP host unreacheable all the time. So, REGISTER is failed. Can you give me some suggestion? Thank you for that.

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[Serusers] npdi strip from uri
by Jason Mohler 22 Sep '05

22 Sep '05

Please help. I have ndpi=yes in my uri and i need to get it stripped out. I have tried replace and it did nto work. Does anyone have any suggestions? Thanks. example: INVITE sip:3302471580;npdi=yes@xxx.xxx.xxx.xxx;user=phone SIP/2.0..Via: SIP/ 2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3accrq304g50udkp01s1.1sr..From: <sip:4432316574@xxx.xxx.xxx.xxx:5060;isup-oli=0;GX=GXDIDGOLD-dkdc1o22 11mf4>;pstn-params=808481808882;tag=SD5gi2b01-121..To: <sip:3302471580@xxx.xxx.xxx.xxx :5060>..Call-ID: SD5gi2b01-74753aacf8cebc27afa2abeba01ef67b-v3000 i1..CSeq: 31346 INVITE..Allow: OPTIONS, SUBSCRIBE, INVITE, CANCEL, ACK, PRA CK, INFO, REFER, NOTIFY, BYE..Accept: multipart/mixed, application/sdp, app lication/isup, application/dtmf, application/dtmf-relay..Contact: <sip:4432316574@xxx.xxx.xxx.xxx:5060;GX=GXAPTELADIDGOLD-dkdc1o2211mf4;transport=udp> ..Remote-Party-ID: <sip:4432316574@xxx.xxx.xxx.xxx:4000>;privacy=off..Content- Length: 161..Content-Disposition: session;handling=required..Content-Type: application/sdp..Max-Forwards: 70....v=0..o=Sonus_UAC 31345 16101 IN IP4 xxx.xxx.xxx.xxx. .s=SIP Media Capabilities..c=IN IP4 xxx.xxx.xxx.xxx..t=0 0..m =audio 50050 RTP/AVP 0..a=sendrecv..a=ptime:20..

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[Serusers] How to enable aliases without expiration?
by Ryan Pagquil 21 Sep '05

21 Sep '05

Hi guys, How can I make an alias that will never expire? I want my users to use their username to register in SER but I want them to make calls using numbers or aliases? is this possible? I checked the alises table and it has an expire field, does it really expire on that time? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Users] RE: AW: [Serusers] CDRtool-3.3.3
by harry gaillac 21 Sep '05

21 Sep '05

Hello, > That's what i find in > http://cdrtool.ag-projects.com/EVALUATION.txt I read it but can we set the billing application for non commercial use ? I just can't install the basic gui which one read the radacct table . I can't set rates tables, ... Does ag-projects.com provide a very limited of CDRTool ? Thanks for help Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] re: SER, rtpproxy, and zyxel wifi phone
by Yair Hakak 21 Sep '05

21 Sep '05

hi all, i have a zyxel wifi phone running with SER. it registers fine, and i can dial out. However, i can't hear anything. configuration is as follows: zyxel behind nat (d-link home router) <->SER on public IP:5060 <->asterisk on same public IP:5070 i set up an extension on asterisk to play music, called it from the zyxel. no music. i am using rtpproxy. and the zyxel is set up with NAT=outbound proxy. if i do the same from an ATA behind the same nat, i get the music. does anyone have a zyxel working behind a nat, and have a config to share? or any other help? all help is much appreciated, yair

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[Users] Radius Authorization and AVP Retrieval
by Tavis P 21 Sep '05

21 Sep '05

I'm having some trouble trying to integrate both Radius authentication and Radius AVP storage. I'm using freeradius and there doesn't seem to be much documentation regarding configuration semantics and such. Could anyone share some information on how they were able to use a Radius server to authenticate requests and (with avp_radius) as an AVP retrieval mechanism? I'm uncertain as to how I can stack these two uses of data on the freeradius server, the problem is that when a client authenticates (either INVITE proxy-authorization or REGISTER) the radius transaction is successfull and several SIP-AVP attributes are returned. Later on in the script i'm trying to use the avp_radius module (avp_load_radius function) to load SIP-AVPs for the callee which fails on the radius server due to the lack of digest authentication credentials in the radius requst. Is there some technique i can employ on the freeradius server to allow this second transaction? I'm not a freeradius expert and so i have not yet found a way to do this tavis

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[Serusers] Stun Exception
by rajesh volluru 21 Sep '05

21 Sep '05

hi Would anyone please help me handling below exception while using sip communicator 2594 [NetworkDiagnosticsThread] ERROR common.NetworkDiagnostics - The network diagnostics process has failed with the following exception net.java.stun4j.StunException: An Exception occurred while sending message bytes through a network socket! at net.java.stun4j.stack.NetAccessManager.sendMessage(NetAccessManager.java:397) at net.java.stun4j.stack.StunClientTransaction.sendRequest(StunClientTransaction.java:205) at net.java.stun4j.stack.StunProvider.sendRequest(StunProvider.java:95) at net.java.stun4j.client.BlockingRequestSender.sendRequestAndWaitForResponse(BlockingRequestSender.java:79) at net.java.stun4j.client.NetworkConfigurationDiscoveryProcess.doTestI(NetworkConfigurationDiscoveryProcess.java:284) at net.java.stun4j.client.NetworkConfigurationDiscoveryProcess.determineAddress(NetworkConfigurationDiscoveryProcess.java:186) at net.java.sip.communicator.common.NetworkDiagnostics.run(NetworkDiagnostics.java:94) Caused by: java.lang.NullPointerException: null address || null buffer at java.net.PlainDatagramSocketImpl.send(Native Method) at java.net.DatagramSocket.send(DatagramSocket.java:611) at net.java.stun4j.stack.NetAccessPoint.sendMessage(NetAccessPoint.java:200) at net.java.stun4j.stack.NetAccessManager.sendMessage(NetAccessManager.java:393) ... 6 more Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

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[Users] CDRtool-3.3.3
by harry gaillac 21 Sep '05

21 Sep '05

Hello, I wish to setup CDRTool-3.3.3 in order to test a billing system. I can't configure neither rating engine nor SOAP/XML server in non commercial release ! Does anybody know this billing system ? Regards harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] | operator
by Federico Giannici 21 Sep '05

21 Sep '05

The "|" operator (in expressions like method == "REGISTER|INVITE|MESSAGE") seems to DON'T work when used more than once in the string (with more than two choices). Is this a known limitation? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] using postgresql
by Nhadie 21 Sep '05

21 Sep '05

Hi All, I'd like to use postgres instead of mysql. Is there a script like ser_mysql.sh to create the database/tables or should I add it manually? Regard Nhadie

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[Serusers] reach a radius attribute
by Mihaly Zachar 21 Sep '05

21 Sep '05

Hi everybody, I would like to build a condition, based on an attribute (h323-preferred-lang), what I get while the registration process. How can I reach this value in ser.cfg ?? What should I read about it ? Thanks, regards, Misi

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[Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
by Aisling O'Driscoll 21 Sep '05

21 Sep '05

Hello everyone, I am trying to modify the Onsip call forward script to forward to asterisk voicemail given a users preference. I am using the script that Java RockX (Paul) posted in January... http://mail.iptel.org/pipermail/serusers/2005-January/014968.html I think I am getting very confused though with the logic...I want users to be able to load their preference using the avpops module of whether they want call forward on no answer or voicemail. They get directed to voicemail if they dial their own extension number....I have posted the modified script which is a combination of the Pauls script and the Onsip script...When I start SER I currently have these errors but I'd say these are a consequence of my messed up logic ;).... 0(20031) ERROR:parse_avp_name: unsupported type [1] 0(20031) ERROR:add_avp_galias_str: <500> not a valid AVP name 0(20031) parse error (70,20-21): Can't set module parameter The config is below.... If anyone can offer some insight, Id be very grateful. Many thanks, Aisling. # $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ #debug=3 #fork=yes #log_stderror=no #listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE #port=5060 #children=4 check_via=no dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:password@localhost/ser" alias="x.x.x.x:5060" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/cpl-c.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/group.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://root:password@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) #ping every 30 seconds modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("cpl-c", "cpl_db", "mysql://root:password@localhost/ser") modparam("cpl-c", "cpl_table", "cpl") modparam("cpl-c", "cpl_dtd_file", "/tmp/ser-0.9.3/modules/cpl-c/cpl-06.dtd") modparam("cpl-c", "lookup_domain", "location") modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("speeddial", "db_url", "mysql://root:password@localhost/ser") modparam("speeddial", "user_column", "userid") modparam("speeddial", "sd_user_column", "short_user") modparam("speeddial", "sd_domain_column", "short_domain") modparam("speeddial", "new_uri_column", "real_uri") modparam("avpops", "avp_url", "mysql://root:password@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "avp_aliases", "voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type =1:702") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; #------------------------------------------------------------------ # NOTIFY Keep Alive Section #------------------------------------------------------------------ if ((method=="NOTIFY") && search("^Event: keep-alive")){ sl_send_reply("200", "OK"); break; }; #------------------------------------------------------------------ # Speed Dialing Section #------------------------------------------------------------------ if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")) { sd_lookup("speed_dial"); }; #------------------------------------------------------------------ # Do not Disturb Section #------------------------------------------------------------------ #if(avp_db_load("$ruri/username", "s:donotdisturb")){ # if(avp_check("s:donotdisturb", "eq/y/i")){ # route(x) #whereever asterisk voicemail is # break; # }; #}; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; if (method=="INVITE"){ if(is_user_in("Request-URI", "voicemail")){ setflag(31); }; }; # ---------------------------------------------------------------------- -- # URI Compare Section # # Here we compare the "from" and "to" to see if the caller is dialing # their own extension. If so then we route to voicemail if needed # ---------------------------------------------------------------------- -- if (method=="INVITE") { avp_write("$from", "i:34"); if (avp_check("i:34", "eq/$ruri/i")) { if (isflagset(31)) { route(8); break; } else { sl_send_reply("486", "Busy"); break; }; }; }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { if(!cpl_run_script("incoming", "is_stateless")) { # script execution failed t_reply("500", "CPL script execution failed"); }; route(3); break; } else if (method=="REGISTER") { #handle REGISTER messages with CPL script cpl_process_register(); route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { if(isflagset(31)){ setflag(19); }; #sl_send_reply("404", "User Not Found"); #break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; #consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; #Blind Call Forwarding if (avp_db_load("$ruri/username", "s:callfwd")){ setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdbusy")){ if(!avp_check("s:fwdbusy", "eq/$ruri/i")){ setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")){ if(!avp_check("s:fwdnoanswer", "eq/$ruri/i")){ setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5]{ #----------------------------------------------------------------- # PSTN Handler #----------------------------------------------------------------- revert_uri(); rewritehostport("x.x.x.x:5064"); append_branch(); t_relay_to_udp("x.x.x.x", "5064"); break(); t_on_failure("1"); route(4); route(1); } route[6]{ #------------------------------------------------------------------ # Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot # be called from the failure_route block #------------------------------------------------------------------- if(uri=~"^sip:1[0-9][10]@"){ strip(1); }; lookup ("aliases"); if(uri!=myself){ if(!isflagset(22)){ append_branch(); }; route(4); route(1); break; }; if(uri=~"^sip:011[0-9]*@"){ route(4); route(5); break; }; if(!lookup("location")){ if (uri=~"^sip:[0-9][10]@"){ route(4); route(1); break; }; sl_send_reply("404", "User Not Found"); }; route(4); route(1); } route[8] { # voicemail route #2 # # this path this executed during these conditions: # # cond 1) the called number is not in the location table # cond 2) the from_uri == to_uri (ie, caller==callee) if (method == "INVITE" || method == "ACK"){ force_rtp_proxy(); }; rewritehostport("x.x.x.x:5064"); t_on_reply("1"); if (!t_relay()) { if(method == "INVITE" || method == "ACK"){ unforce_rtp_proxy(); }; sl_reply_error(); }; } route[9]{ #voicemail route 1 # #this path is executed during these conditions: # # cond 1) the called number is in the location table # but the callee did not answer the phone (i.e. failover to voicemail) rewritehostport("x.x.x.x:5064"); append_branch(); t_on_reply("1"); if (!t_relay()){ if(method == "INVITE" || method=="ACK"){ unforce_rtp_proxy(); }; sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1]{ if(t_check_status("487")){ break; }; if(isflagset(26) && t_check_status("486")){ if(avp_pushto("$ruri", "s:fwdbusy")){ avp_delete("s:fwdbusy"); resetflag(26); append_branch(); route(6); break; }; }; #Here we can have either voicemail OR forward no answer #forward on no answer is flag 27 #voicemail is flag 31 if(isflagset(27) && t_check_status("408")){ if(avp_pushto("$ruri", "s:fwdnoanswer")){ avp_delete("s:fwdnoanswer"); resetflag(27); append_branch; route(6); break; }; } else if(isflagset(31) && avp_pushto("$ruri", "$voicemail")){ avp_delete("$voicemail"); route(9); break; }; } -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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RE: [Serusers] how to load radius-related modules in ser 0.8.14?
by Ricardo Martinez 21 Sep '05

21 Sep '05

Hi Shuai You need to edit the ser "Makefile" to include the radius related modules and then recompile SER. There is a HOWTO for Radius somewhere in the website. Regards, Ricardo Martinez > -----Mensaje original----- > De: zhangshuai [mailto:zhangshuai@goldentek.biz] > Enviado el: Miércoles, 21 de Septiembre de 2005 7:42 > Para: serusers > Asunto: [Serusers] how to load radius-related modules in ser 0.8.14? > > > Dear all, > > > I'm new to SER. I wish to use softphone X-lite on SER and > radius in db_mode. > > I have installed SER 0.8.14 (by tarball) and Mysql 3.23.54 in > Redhat 9. Also, I have added some subscribers into DB "ser". > After setting /usr/local/etc/ser/ser.cfg as the document > "iptel.org SIP Express Router v0.11.0 -- Admin's > Guide"(http://www.iptel.org/ser/doc/seruser/seruser.html#DEFAU > LTSCRIPT) guided, I could load the modules "auth" and > "auth_db" and successfully log any subscriber that Mysql > contains in X-lite. > > And now, I want to load radius-related modules: "acc", > "auth_radius", "group_radius", "uri_radius" in SER. But when > I put only one line: "loadmodule > "/usr/local/lib/ser/modules/auth_radius.so"" in ser.cfg and > then run ser, there is an error reply: "ERROR: bad config > file (1 errors)". > > How come? How to load those modules properly? > > Many thanks. > > > > Shuai > http://www.godentek.biz > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] db mode radius
by Nhadie 21 Sep '05

21 Sep '05

Hi All, I was able to authenticate via radius using the file /etc/raddb/users 1001(a)my.domain.com Auth-Type := Digest, User-Password == "mypassword" Reply-Message = "Authenticated" Next thing I want to do is authenticate via radius using postgres, I've already created the tables but I don't know what things i should on there. Anyone can give me a hint? I would like to do accounting via radius/postgres. If there are any documents you know I can read kindly send it. Thank You. Regards, Nhadie

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Re: [Serusers] Handoff to PSTN
by Iqbal 21 Sep '05

21 Sep '05

do you have ngrep installed, also to see if you are in the loop, see if you LOG line is being displayed in /var/log/messages or wherever else you are dumping ser's output Colin Jordan wrote: >It is my first time with this, so sorry for the stupid question. I run >ngrep, it does not see it as a command. How can I see if I am falling into >the loop? > >-----Original Message----- >From: Iqbal [mailto:iqbal@gigo.co.uk] >Sent: Wednesday, September 21, 2005 3:11 PM >To: Colin Jordan >Cc: 'Leonid Fainshtein'; serusers(a)lists.iptel.org >Subject: Re: [Serusers] Handoff to PSTN > >Can u post a ngrep of the call, to see what is actually being sent to >and from where. > >Also have you checked to see if you are actually falling into the loop, >i.e the if statement is matching > >Iqbal > >Colin Jordan wrote: > > > >>Hi there >> >>Yes I have just tried it but still do not work. >> >>Best regards, >> >>Colin >> >>-----Original Message----- >>From: Leonid Fainshtein [mailto:LFAIN@boscom.com] >>Sent: Wednesday, September 21, 2005 3:27 PM >>To: cjordan(a)econetwireless.com >>Cc: serusers(a)lists.iptel.org >>Subject: RE: [Serusers] Handoff to PSTN >> >>Hi Colin, >>First of all, according to RFC3261 the pound sign can't be used in the >>user name. Secondly, it may be considered by SER as start of the >>comment. >>Have you tried to put prefix 666 without '#'? >> >>Best regards, >>Leonid Fainshtein >>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. >> >> >> >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >>. >> >> >> >> >> >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. > > > > > >. > > >

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RE: [Serusers] Handoff to PSTN
by Leonid Fainshtein 21 Sep '05

21 Sep '05

Hi Colin, First of all, according to RFC3261 the pound sign can't be used in the user name. Secondly, it may be considered by SER as start of the comment. Have you tried to put prefix 666 without '#'? Best regards, Leonid Fainshtein

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[Users] Freebsd segmentation fault problem
by Diego Valencia 21 Sep '05

21 Sep '05

I compiled the openser 0.9.5 with acc module, and radiusclient-ng 0.5.1 on freebsd 5.4. The compilation is fine, but when I configure the config file to load acc.so module, I see this error: blah..blah..blah WARNING: no fork mode and more than one listen address found(will use only the the first one) 0(28499) shm_mem_init: success 0(0) core dump limits set to 134596096 0(0) DEBUG: init_mod: acc 0(0) ACC - initializing 0(0) find_export: found <load_tm> in module tm [/home/dvalencia/usr/local/lib/openser/modules/tm.so] Segmentation fault (core dumped) Anybody can help me about this? Thanks! ____________________ Diego Valencia Network Support PowerVT SA Tel: 0351-4257748 Movil: 0351-155194286

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[Serusers] how to load radius-related modules in ser 0.8.14?
by zhangshuai 21 Sep '05

21 Sep '05

Dear all, I'm new to SER. I wish to use softphone X-lite on SER and radius in db_mode. I have installed SER 0.8.14 (by tarball) and Mysql 3.23.54 in Redhat 9. Also, I have added some subscribers into DB "ser". After setting /usr/local/etc/ser/ser.cfg as the document "iptel.org SIP Express Router v0.11.0 -- Admin's Guide"(http://www.iptel.org/ser/doc/seruser/seruser.html#DEFAULTSCRIPT) guided, I could load the modules "auth" and "auth_db" and successfully log any subscriber that Mysql contains in X-lite. And now, I want to load radius-related modules: "acc", "auth_radius", "group_radius", "uri_radius" in SER. But when I put only one line: "loadmodule "/usr/local/lib/ser/modules/auth_radius.so"" in ser.cfg and then run ser, there is an error reply: "ERROR: bad config file (1 errors)". How come? How to load those modules properly? Many thanks. Shuai http://www.godentek.biz

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[Serusers] failure route
by Jason Mohler 21 Sep '05

21 Sep '05

Okay everyone...I have an easy one for you. I am trying to do a simple fail over from one ip to the other if I get a 4040 and it fails. Can one of you type the correct syntax for me. Below is what I am trying to accomplish. if (uri=~"sip:^1234567890.*@.*"){ prefix("1234"); rewritehostport("xxx.xxx.xxx.xxx:5060"); I need the hostport to be able to be rolled to another hostport ip if this one comes back with a 404 or any other failure. Can on eof you please help?

2 1

[Serusers] Handoff to PSTN
by Colin Jordan 21 Sep '05

21 Sep '05

I have a big problem handing my call of to the PSTN, It give me a 404 error (not connected). What can I do to fix this? My handoff script looks like this: if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); prefix("666#"); t_relay_to_udp("217.15.11.87","5060"); break; }; The PSTN interconnect description: 1) I need to send a prefix of 666# followed by full intl number without leading 0's. eg. 666#442071016211 2) The PSTN are connecting me to a Cisco SIP Proxy server, which uses Tech-prefixes eg 123# to separate and authenticate the calls into different domains. What can I do to get it working? 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.

4 6

[Serusers] Problem with Notify
by Sam Lee 21 Sep '05

21 Sep '05

Dear all, I am getting 404 User Not Found whenever i received a NOTIFY method. Don't know what is wrong with it. All i did in the ser.cfg was a simple lookup("location") but it seems that they cannot find the record in the location table. I have check and confirm that the user location record is in the table. What is going on ? What are they suppose to compare or what is the lookup suppose to do ? What i cannot understand is why are there no if else to take in method==NOTIFY . Please help. Regards, Sam

1 0

[Serusers] warning(4349) Warning: could not rev.resolve 192.168.0.1
by julien bossart 21 Sep '05

21 Sep '05

Hi all experts I install ser in server linux 192.168.0.1 <http://192.168.0.1> fedora. After start ser, i see in the console: Warning:could not rev.resolve 192.168.0.1 <http://192.168.0.1> So that when i send REGISTER to server SER. SER always responds ICMP host unreachable. What do you think? Many thank for your help

1 0

[Serusers] ser.cfg
by julien bossart 21 Sep '05

21 Sep '05

Hi all, I use ser.cfg defaul, and uncomment some line to support mysql. After start ser, i use: /usr/sbin/serctl ps i see in the console that: stand-alone receiver @ 127.0.0.1:5060 <http://127.0.0.1:5060> so that i always got ICMP host unreachable when i use SIP client to send REGISTER. How can i add IP server for ser? Thank you

1 0

RE: [Serusers] SER with asterisk as voicemail problem... please help!
by Quang Minh PHAN 21 Sep '05

21 Sep '05

Hi, It seems to be a problem in Asterisk configuration, not in SER's. Showing your asterisk's extensions.conf file may help. Regards, Minh -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Ryan Pagquil Envoyé : mardi 20 septembre 2005 09:50 À : serusers(a)lists.iptel.org Objet : [Serusers] SER with asterisk as voicemail problem... please help! Hi, I configured SER to forward certain calls to asterisk for the voicemail functionality. But when I forward the SIP request to Asterisk my xlite softphone prompts "403", "Forbidden". This is my ser.cfg part where i forward the certain uri to the asterisk. if (uri=~"^sip:1[0-9]{3}@sip.philonline.com.*$") { strip(1); rewritehostport("202.84.24.47:5060"); t_relay(); break; }; Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] need help
by julien bossart 21 Sep '05

21 Sep '05

Hi all, I am new to this forum. Today, i download ser package, then i install in my fedora core 4. Everything is ok. Next, i start ser and use serctl add to add account to ser server. Then i use sjphone to connect to ser server. Sjphone try to send REGISTER to ser server. I use ethereal to get the package. I see that ser server responds ICMP host unreacheable all the time. So, REGISTER is failed. Can you give me some suggestion? Thank you for that.

1 0

[Users] Compilation error on Mac OS
by Amos Nungu 21 Sep '05

21 Sep '05

I'm installing SEMS-0.9 on Mac OS (Darwin 8.2.0). upon running the make all command, I got an error as shown below ------------------------------------------------- # make all make[1]: Nothing to be done for `deps'. g++ -o sems AmApi.o AmAudio.o AmCmd.o AmConfig.o AmEventQueue.o AmIcmpWatcher.o AmMail.o AmPlugIn.o AmRequest.o AmRtpPacket.o AmRtpPacketTracer.o AmRtpScheduler.o AmRtpStream.o AmSdp.o AmServer.o AmSession.o AmSmtpClient.o AmThread.o AmUtils.o EmailTemplate.o SemsConfiguration.o SerClient.o SerDBQuery.o log.o sems.o -lm -ldl -lsocket -lnsl -lpthread /usr/bin/ld: can't locate file for: -lsocket collect2: ld returned 1 exit status make[1]: *** [sems] Error 1 make: [all] Error 2 (ignored) ------------------------------------------------------- Any idea on how to go about it will be great. Regards

2 1

[Users] SER + MediaProxy or rtpproxy
by Frank Kostin 21 Sep '05

21 Sep '05

Hi All, Using Onsip SER with Mediaproxy --no-fork I can see caller/called/relayed. This is OK for local calls only. With or without mediaproxy I have audio both directions. When I call to /from another network the phone rigning but no audio. Can I relay /forward by IP address using mediaproxy and eventually redirect media streams throught a 3rd party ? Thanks, Frank --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 1

[Serusers] Problem redirecting to voicemail
by Chris St Denis 20 Sep '05

20 Sep '05

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail. GATEWAY---PROXY---FEATURE | | UA For simplicity, hostnames and IPs replaced with the above names. USERNAME, DSTNUM and SRCNUM also used to replace the UA's username, the source number of the call, and the destination number of the call. The basic SIP dialog goes: Gateway invites proxy Gateway invites UA UA replies 180 Ringing. Transaction times out and drops to failure route PROXY invites FEATURE server (INVITE sip:*voicemail-busy-2002006@ FEATURE:5060 SIP/2.0.) PROXY cancels UA FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY UA tells PROXY '487 Request Terminated.' FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY ... For some reason, the asterisk gateway doesn't seem to be ACKing the 200 ok. I don't see any new invite or reinvite going to the gateway which I think may be what is confusing it. Ser.cfg excerpt ## # User did not answer phone, or could not connect, or is on the phone and does not use call waiting. ## failure_route[3] { if(isflagset(10)) { if(t_check_status("486")) { if (!subst_user('/^/*voicemail-busy-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail Busy\n"); } else { if (!subst_user('/^/*voicemail-noanswer-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail No answer\n"); } rewritehostport("FEATURE:5060"); append_branch(); t_relay(); } } Ngrep from PROXY's point of view http://pastebin.ca/23469 Ngrep from GATEWAY's point of view http://pastebin.ca/23470 Ngrep from FEATURE's point of view http://pastebin.ca/23471

1 0

[Serusers] Can SER determine NAT type?
by Chris St Denis 20 Sep '05

20 Sep '05

I want to use STUN as a primary nat solution, with the nathelper or mediaproxy module doing rewrites as a secondary/backup solution. However, this will not work for symmetric NATs. I want to proxt the rtp data from these (and only these) kind of nat. Can SER determine that? My client adds a Warning: 399 spa "Restricted Cone NAT Dectected". Header to the register message (determined by stun), but that looks proprietary and not something I can expect to be the same from other clients so I'm not sure I want to header match on that.

1 0

[Serusers] isntalling SER on Red hat linux
by rajesh volluru 20 Sep '05

20 Sep '05

hi Can we install SER on Red Hat linux. If yes, then please send me the instructions. Also, I get Transaction unavailable exception when I try to communicate through Sip Communicator. Does installing SER solve that problem? Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

3 2

Re: [Serusers] COnfiguring a PSTN number with SER
by Iqbal 20 Sep '05

20 Sep '05

in ur messages file, i have u traced to see where in ur config its getting stuck, i mean when it gos through with the INVITE when it does alias tabel lookup is it matching anything, also look in mysql log file to see what info it is selecting upon. what is the format of your entry in aliases, it will map your username to alias, wheras you want number mapped to username i think iqbal Ashutosh Kumar wrote: >Hi, > I assigned the DID to the user but serctl add alias, and the entry >got updated in ser.aliases table. > But when I call the number from pstn, the registered user does not get >the call. The nextone contacts the SER, but SER responsds to it with a "478 >unresolvable desintation". > Following is the ngrep excerpt. > >U PSTN_GW_IP:5060 -> SER_IP:5060 >INVITE sip:@SER_IP SIP/2.0. >Max-Forwards: 70. >Session-Expires: 3600;Refresher=uac. >Supported: timer. >. >v=0. >o=msw1 1234 1929 IN IP4 x.y.z.101. >s=sip call. >c=IN IP4 x.y.z.101. >t=0 0. >m=audio 27468 RTP/AVP 0 8 3 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:8 PCMA/8000. >a=rtpmap:3 GSM/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-16. >a=silenceSupp:off - - - -. > >## >U SER_IP:5060 -> PSTN_GW_IP:5060 >SIP/2.0 100 trying -- your call is important to us. > > >############# >U SER_IP:5060 -> PSTN_GW_IP:5060 >SIP/2.0 478 Unresolvable destination (478/TM). > >Relevant code from ser.cfg is as follows: > >... >... >modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") >modparam("usrloc", "db_mode", 2) > > > if (uri==myself) { > > if (method=="ACK") { > route(7); > break; > } else if (method=="CANCEL") { > route(3); > break; > } else if (method=="INVITE") { > route(3); > break; > } else if (method=="MESSAGE"){ > route(4); > break; > } else if (method=="REGISTER") { > route(2); > break; > }; > > lookup("aliases"); > if (uri!=myself) { > route(5); > route(1); > break; > }; > > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > break; > }; > > >I don't know where I am wrong. > >Regards, >Ashutosh > > > > > >-----Original Message----- >From: Iqbal [mailto:iqbal@gigo.co.uk] >Sent: Tuesday, September 20, 2005 10:35 PM >To: Ashutosh Kumar >Cc: 'Andreas Granig'; serusers(a)lists.iptel.org >Subject: Re: [Serusers] COnfiguring a PSTN number with SER > >just alias that number to a voip username which is bound to that user, >so when 123456789 is dialed nextone sends it to ser, ser looks up in >aliases, sees the match and sends it to that user > >Iqbal > >Ashutosh Kumar wrote: > > > >>Hi Andreas, >> Thanks for the reply, I was thinking of another approach to do this, >>tell me if I am wrong. >> May I use the 'phone' field in subscriber table to run an external >>sql script to map the DID (phone) with a username, and then rewrite the >> >> >uri, > > >>and then forward the call to that user? >> >>Regards, >>Ashutosh >> >> >>-----Original Message----- >>From: Andreas Granig [mailto:andreas.granig@inode.info] >>Sent: Tuesday, September 20, 2005 6:22 PM >>To: Ashutosh Kumar >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] COnfiguring a PSTN number with SER >> >>Ashutosh Kumar wrote: >> >> >> >> >>> I have a DID number to which I want to assign to a user in >>>SER, the call will come through a NExtone. How / what changes I have to >>>make in ser.cfg to arrange that whenever a call comes on that number on >>>nextone, that call should be forwarded to the sip-user bound to that >>> >>> >>> >>> >>number. >> >> >> >> >>>How can I use aliases in this scenario. >>> >>> >>> >>> >>Check the PSTN chapter of the Getting Started document at http://onsip.org >> >>Andy >> >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >>. >> >> >> >> >> > > > >. > > >

1 0

[Serusers] COnfiguring a PSTN number with SER
by Ashutosh Kumar 20 Sep '05

20 Sep '05

Hi, I have a DID number to which I want to assign to a user in SER, the call will come through a NExtone. How / what changes I have to make in ser.cfg to arrange that whenever a call comes on that number on nextone, that call should be forwarded to the sip-user bound to that number. How can I use aliases in this scenario. Please guide. Regards, Ashutosh Kumar

3 4

[Serusers] obsfuctaing ser.cfg
by Iqbal 20 Sep '05

20 Sep '05

is this possible other than just removing the spaces and all. Iqbal

1 0

[Serusers] accounting and radius/sql
by harry gaillac 20 Sep '05

20 Sep '05

Hello ser users, I set accounting with freeradius 1.0.2 and mysql for testing billing (CDRTool). I configured radius to work with mysql. I load acc module with radius support it's ok. I have a problem with accouting and radacct table. In fact "AcctStopTime" and related to time fields are not updated . I would need experienced users with freeradius server and ser to debug accounting . Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] asterisk vm and ser
by Sebastian Kühner 20 Sep '05

20 Sep '05

Hello, I'm using ser with asterisk as voicemail server. I send a notify message to a user that has a new message. If the user deletes his messages, asterisk (and sipsak) sends the notify message correctly and the VM-Indicator turns off... but after a while, my SIPURA turns on the vm indication without receiving a new notify. Do I have to send this notify message periodically? Thanks! Sebastian

1 0

[Serusers] Possible bug
by Rubén Mazariegos del Rey 20 Sep '05

20 Sep '05

Hi! I am using version 0.9.3 of SER and I have observed the following behaviour: When a user sends a SIP message to another user through SER, and the body of the SIP message ends with the byte "0x00", this byte is removed by SER from the message before forwarding it to the destination. Is this a known behaviour? I suppose it is a bug, isn't it? Is there a newer version of SER fixing this bug? Thanks in advance, Rubén

1 0

[Serusers] sdp and rtp errors
by Rick Thompson 20 Sep '05

20 Sep '05

Hi All I hope someone can help me with an issue I"ve had since installing ser. For some reason, whenever I call fix_nated_sdp or force_rtp_proxy I get the following errors Sep 19 10:31:06 localhost ser[3921]: Mon Sep 19 10:31:06 2005 18 a.b.c.d 62141 ****** Fixing NAT on reply with rtp Sep 19 10:31:06 localhost ser[3921]: ERROR: extract_body: message body has length zero Sep 19 10:31:06 localhost ser[3921]: ERROR: fix_nated_sdp: cannot extract body from msg! Sep 19 10:31:06 localhost ser[3921]: ERROR: extract_body: message body has length zero Sep 19 10:31:06 localhost ser[3921]: ERROR: force_rtp_proxy2: can't extract body from the message Sep 19 10:31:06 localhost ser[3921]: ERROR: on_reply processing failed Here is the part of the script that is generating the error onreply_route[1] { xlog("L_ALERT", "%Tf %mf ****** On Reply From: <%si> <%tu> <%fu> <%rm> <%rs> <%rr>\n"); if(status=~"[12][0-9][0-9]" && nat_uac_test("19")) { fix_nated_contact(); force_rport(); }; if(status=~"[12][0-9][0-9]" && nat_uac_test("19") && src_port>=10000) { xlog("L_ALERT", "%Tf %mf %si %sp ****** Fixing NAT on reply with rtp\n"); fix_nated_sdp("3"); force_rtp_proxy(); } else if(status=~"[12][0-9][0-9]" && nat_uac_test("19")) { xlog("L_ALERT", "%Tf %mf %si %sp ****** Fixing NAT on reply with sdp\n"); fix_nated_sdp("3"); }; } Any help would be appreciated. Thanks Rick

2 3

[Serusers] UltraEdit
by Rick Thompson 20 Sep '05

20 Sep '05

Hi all I was wondering if anybody has written a ser script wordlist.txt for ultraEdit and would mind sharing it. Just so I don't have to re-invent the wheel. Thanks Rick

1 0

[Serusers] Solaris8/SPARC compilation
by Ryan Pagquil 20 Sep '05

20 Sep '05

Hi guys, I'm having a problem with SER-0.9.3 installation in Solaris8/SPARC i've got these errors: bash-2.03# make all Makefile.rules:84: action.d: No such file or directory Makefile.rules:84: crc.d: No such file or directory Makefile.rules:84: daemonize.d: No such file or directory Makefile.rules:84: data_lump.d: No such file or directory Makefile.rules:84: data_lump_rpl.d: No such file or directory Makefile.rules:84: dprint.d: No such file or directory Makefile.rules:84: dset.d: No such file or directory Makefile.rules:84: error.d: No such file or directory Makefile.rules:84: fifo_server.d: No such file or directory Makefile.rules:84: flags.d: No such file or directory Makefile.rules:84: forward.d: No such file or directory Makefile.rules:84: hash_func.d: No such file or directory Makefile.rules:84: ip_addr.d: No such file or directory Makefile.rules:84: main.d: No such file or directory Makefile.rules:84: md5.d: No such file or directory Makefile.rules:84: md5utils.d: No such file or directory Makefile.rules:84: modparam.d: No such file or directory Makefile.rules:84: msg_translator.d: No such file or directory Makefile.rules:84: pass_fd.d: No such file or directory Makefile.rules:84: proxy.d: No such file or directory Makefile.rules:84: qvalue.d: No such file or directory Makefile.rules:84: re.d: No such file or directory Makefile.rules:84: receive.d: No such file or directory Makefile.rules:84: resolve.d: No such file or directory Makefile.rules:84: route.d: No such file or directory Makefile.rules:84: route_struct.d: No such file or directory Makefile.rules:84: script_cb.d: No such file or directory Makefile.rules:84: socket_info.d: No such file or directory Makefile.rules:84: sr_module.d: No such file or directory Makefile.rules:84: stats.d: No such file or directory Makefile.rules:84: tcp_main.d: No such file or directory Makefile.rules:84: tcp_read.d: No such file or directory Makefile.rules:84: timer.d: No such file or directory Makefile.rules:84: tsend.d: No such file or directory Makefile.rules:84: udp_server.d: No such file or directory Makefile.rules:84: unixsock_server.d: No such file or directory Makefile.rules:84: usr_avp.d: No such file or directory Makefile.rules:84: mem/f_malloc.d: No such file or directory Makefile.rules:84: mem/mem.d: No such file or directory Makefile.rules:84: mem/memtest.d: No such file or directory Makefile.rules:84: mem/q_malloc.d: No such file or directory Makefile.rules:84: mem/shm_mem.d: No such file or directory Makefile.rules:84: mem/vq_malloc.d: No such file or directory Makefile.rules:84: parser/hf.d: No such file or directory Makefile.rules:84: parser/msg_parser.d: No such file or directory Makefile.rules:84: parser/parse_allow.d: No such file or directory Makefile.rules:84: parser/parse_content.d: No such file or directory Makefile.rules:84: parser/parse_cseq.d: No such file or directory Makefile.rules:84: parser/parse_disposition.d: No such file or directory Makefile.rules:84: parser/parse_diversion.d: No such file or directory Makefile.rules:84: parser/parse_event.d: No such file or directory Makefile.rules:84: parser/parse_expires.d: No such file or directory Makefile.rules:84: parser/parse_fline.d: No such file or directory Makefile.rules:84: parser/parse_from.d: No such file or directory Makefile.rules:84: parser/parse_hname2.d: No such file or directory Makefile.rules:84: parser/parse_hostport.d: No such file or directory Makefile.rules:84: parser/parse_methods.d: No such file or directory Makefile.rules:84: parser/parse_nameaddr.d: No such file or directory Makefile.rules:84: parser/parse_param.d: No such file or directory Makefile.rules:84: parser/parse_rpid.d: No such file or directory Makefile.rules:84: parser/parse_rr.d: No such file or directory Makefile.rules:84: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:84: parser/parse_to.d: No such file or directory Makefile.rules:84: parser/parse_uri.d: No such file or directory Makefile.rules:84: parser/parse_via.d: No such file or directory Makefile.rules:84: parser/parser_f.d: No such file or directory Makefile.rules:84: parser/digest/digest.d: No such file or directory Makefile.rules:84: parser/digest/digest_parser.d: No such file or directory Makefile.rules:84: parser/digest/param_parser.d: No such file or directory Makefile.rules:84: parser/contact/contact.d: No such file or directory Makefile.rules:84: parser/contact/parse_contact.d: No such file or directory Makefile.rules:84: db/db.d: No such file or directory Makefile.rules:84: db/db_fifo.d: No such file or directory Makefile.rules:84: lex.yy.d: No such file or directory Makefile.rules:84: cfg.tab.d: No such file or directory yacc -d -b cfg cfg.y conflicts: 1 shift/reduce make: *** [cfg.tab.c] Broken Pipe what could be the possible cause of these errors? How will i deal with it? Another question, does the ser-0.9.3 package for solaris support mysql, radius accounting and authentication? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

4 4

[Serusers] busy
by Sebastian Kühner 20 Sep '05

20 Sep '05

Hello, Does anybody know how I can detect if a phone is busy. I make accounting with ser and sometimes the BYE messages don't reach my server, so I have to test if a phone is busy without using INVITE messages... Thanks! Sebastian

1 0

Re: [Serusers] LDAP authentication (without Radius)
by Arek Bekiersz 20 Sep '05

20 Sep '05

Yes, there is a nice module, it was created by Marcel Baur, ETH Zurich. You can find it under: http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldaps It should be helpful to take a look at the code and just get idea how it works. Greetings, Arek ----- Original Message ----- From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> To: <clona(a)camaro.no>; <sip(a)perceval.net> Cc: <serusers(a)lists.iptel.org> Sent: Monday, September 19, 2005 10:32 AM Subject: Re: [Serusers] LDAP authentication (without Radius) > If I don't recall wrong, there was an LDAP module for SER developed in > some university/research center from Switzerland, and I am pretty sure you > could dopwnload it....just try to google, because I haven't got internet > access right now... > > Samuel. >

2 1

[Serusers] LCR Module on MaC OS (Reverse IP)
by Amos Nungu 20 Sep '05

20 Sep '05

> > I managed to get LCR running on both Debian-Sarge and MacOS-Darwin, > thanks. > I'm running ser 0.9.3 from source on both machines. > However there exists some di-similarities between in operations between > the two installations dispite of using the same installation source and same > ser.cfg logic. > 1.I convert my IP to decimal using www.dnsstuff.com<http://www.dnsstuff.com/>( > 192.168.0.101 <http://192.168.0.101>) to get 1694542016. > 2.Add that value into the lcr table. > 3.Making a PSTN call, using xlog to capture %ru > > Result: > Debian configuration will show abc(a)192.168.0.101 while MacOS will show > abc(a)101.0.168.192 > > I can get the desired effect if I convert the IP address in reverse order > for MacOS (192.168.0.101 <http://192.168.0.101>) to (3232235621). But, I > think this is not how it was supposed to be. > > Any idea? > > Regards. >

1 0

[Serusers] Acc table fields
by Federico Giannici 20 Sep '05

20 Sep '05

I cannot understad the meaning and the purpose of the two fields "caller_deleted" and "callee_deleted" in the "acc" SQL table. I found no reference to them in the entire SER sources (0.9.4)... Anybody knows what is their purpose? As we are talking about accounting... I'd like if it was possible to register the source IP of the request messages and the session media info (if present): used codec, media type (audio, video...), etc... Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 2

RE: [Serusers] Handoff to PSTN with a set prefix
by Colin Jordan 20 Sep '05

20 Sep '05

I need to send a prefix of 666# followed by full intl number without leading 0's. eg. 666#442071016211 The PSTN are connecting me to a Cisco SIP Proxy server, which uses Tech-prefixes eg 123# to separate and authenticate the calls into different domains. Can you please help me? -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Tuesday, September 20, 2005 10:50 AM To: Colin Jordan Cc: serusers(a)lists.iptel.org; serusers-request(a)lists.iptel.org Subject: Re: [Serusers] Handoff to PSTN with a set prefix Colin Jordan wrote: > # attemt handoff to PSTN > if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { > log ("Forwarding to PSTN\n"); > t_relay_to_udp("217.15.11.87","5060"); > t_relay_to_tcp("217.15.11.87","5060"); > break; > }; Hm? Both UDP and TCP? Try: if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); prefix("666#"); t_relay_to_udp("217.15.11.87","5060"); break; }; I'm not sure about the "#" and if it's parsed correctly or if it's treated as start of a comment, haven't ever tried that. Andy

1 0

[Serusers] Handoff to PSTN with a set prefix
by Colin Jordan 20 Sep '05

20 Sep '05

I need to set a fixed prefix of, 666#, to enable me to break out to the PSTN that I am sending my traffic. Will you please help me with the needed commands and where to place it? Please find below the ser.cfg file: ________________________________ # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) # new insterted statement listen=66.8.91.200 listen=127.0.0.1 # hostname matching an allias will satify the condition uri-myself alias=gaco.co.za alias=66.8.91.200 /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) # this was commented port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # database location - NEW #modparam("auth_db, "db_uri, "sql://ser:>heslo>@localhost/ser") # 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"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("gaco.co.za", "subscriber")) { www_challenge("gaco.co.za", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # attemt handoff to PSTN if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); t_relay_to_udp("217.15.11.87","5060"); t_relay_to_tcp("217.15.11.87","5060"); break; }; } ________________________________ 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 2

[Serusers] Enabling Radius communication between SER and Alepo RBS
by Thomas Tomter 20 Sep '05

20 Sep '05

Hello, We are working on establishing Radius-communication between SER and the billing system Alepo RBS version 4.2, but we are experiencing some trouble with this. We think the problem might be caused by the way we have defined SER as a network component in the Alepo RBS configuration file called 'NasCfgDbs' and maybe how the SER dictionary is built up. Does anyone have any experience with integration of SER with Alepo RBS? We would appreciate any kind of help or tip. Thanks in advance. Best regards, Thomas

1 0

[Serusers] SER with asterisk as voicemail problem... please help!
by Ryan Pagquil 20 Sep '05

20 Sep '05

Hi, I configured SER to forward certain calls to asterisk for the voicemail functionality. But when I forward the SIP request to Asterisk my xlite softphone prompts "403", "Forbidden". This is my ser.cfg part where i forward the certain uri to the asterisk. if (uri=~"^sip:1[0-9]{3}@sip.philonline.com.*$") { strip(1); rewritehostport("202.84.24.47:5060"); t_relay(); break; }; Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] 408 to Caller UA when CANCEL to Callee
by Tulika Pradhan 20 Sep '05

20 Sep '05

hi, i am facing the following situation. UA1 calls a user(UA2) who does not answer. the control comes to failure_route where i try another UA (UA3). but as UA3 rings, SER sends 408 Request timeout to UA1 and call gets disconnected. this is the SIP message flow. UA1 SER UA2 UA3 INVITE----------------> INVITE--------------> <----------------TRYING <----------------RINGING <------------------RINGING CANCEL--------------> <---------------------408 INVITE----------------------------------------> <---------------------487 ACK-------------------> <-----------------------OK <-------------------------------------------TRYING <--------------------------------------------RINGING (but UA already has got the busy tone) and does not hear this ringing. if 408 was not sent to UA1, then the call could have been established. what is going wrong, regards, tulika

4 10

[Serusers] SER+MEDIAPROXY+ B2BUA
by Amit.Chawre＠relianceinfo.com 20 Sep '05

20 Sep '05

Hello I want to test a scenario with SER, MEDIAPROXY and Vovida B2BUA, but I'm having some problems. If someone has tested this before, I would like to ask you the config files to help me. When NATedUA1 calls NATedUA2, invite reaches NATedUA2. NATedUA2 sends responses back to SER (100 trying 180 ringing) but SER doesn’t forward it further to NATedUA1.I'm having problems when NatedUA2 send '100 Trying' and '180 ringing' to SER, it forwards it to B2BUA, but then there is no flow of messages. NATedUA2 ^ | 4.INVITE | 1.INVITE 2.INVITE NATedUA1 ----> SER+MEDIAPROXY ------> B2BUA <----- 3. INVITE ser.cfg is: # Example ser.cfg for mediaproxy functionality check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=72.21.40.107 port=5060 children=4 fifo="/tmp/ser_fifo" # fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" modparam("usrloc", "db_mode", 0) #modparam("usrloc", "db_url","mysql://ser:heslo@localhost/ser") modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; if (method=="REGISTER") { if (is_from_local()) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!radius_www_authorize("72.21.40.107")) { www_challenge("72.21.40.107", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; #if (method=="INVITE") { t_on_reply("1"); #}; if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if(!(src_port==5065 && src_ip==72.21.40.107)){ if (method=="INVITE" || method=="BYE") { log("amit: Forwarding to B2bua\n"); forward(72.21.40.107,5065); break; }; }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { #if (status=~"(183)|(2[0-9][0-9])") { if (status=~"[1-2][0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } Thanks for your help amit

1 0

[Serusers] SER + MediaProxy or rtpproxy
by Frank Kostin 20 Sep '05

20 Sep '05

Hi All, Using Onsip SER with Mediaproxy --no-fork I can see caller/called/relayed. This is OK for local calls only. With or without mediaproxy I have audio both directions. When I call to /from another network the phone rigning but no audio. Can I relay /forward by IP address using mediaproxy and eventually redirect media streams throught a 3rd party ? Thanks, Frank --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

1 0

[Users] request service to openser
by Loide Mara 20 Sep '05

20 Sep '05

How the UA requests a service for proxy (openser)? Is possible a direct communication between the UA and proxy? For example... How the UA requests the blacklist's service? Somebody has some suggestion? -------------------------------------------------- Loide Mara Verdes de Jesus 6º período - Ciência da Computação / UFAM Laboratório VoIP - UFAM

2 1

[Users] Bug on pseudevariables in failure_route
by Franz Edler 20 Sep '05

20 Sep '05

Hi, I am actually testing with openser 0.10.0-dev12-tls and detected the following problem: When I use pseudo-variables $rs and $rr in a failure_route I get <null> instead of reply status and reply reason in case of a failure response. I think this is a bug. Regards Franz

2 1

[Serusers] Instant Messaging cannot work with Windows Messenger 5.0
by kokmeng＠moij.biz 20 Sep '05

20 Sep '05

Hi, I have two PCs running Windows Messenger 5.0 and SER as registrar. I am able to get Voice, Video, App Sharing, Whiteboard, Presence (peer to peer only) working. However, I cannot get Instant Messaging to work. When I look at my Ethereal capture, I see that with 5.0, INVITE is used to start an IM session. The other client responds with TRYING, then OK. However, it also sends a BYE immediately after sending the OK. Subsequently when the originating client sends the MESSAGE, the other client responds with an Error because "Call Leg does not exist". Has anyone encountered this before? I've seen this problem being mentioned in Google searches, but no answer so far. Regards, KokMeng Loh

3 3

[Serusers] DNS SRV for load balancing
by Peter 19 Sep '05

19 Sep '05

We are trying to use DNS SRV records to load balance outbound transactions across 2 or more servers. Something like this: UA => SER => (Some processing logic) => Back to SER after a 300 Redirect which contains a Contact phone number and IP/DNS at which to contact the contact => DNS SRV lookup for the contact IP and send it to one of the two servers returned by the DNS lookup. So this last leg is where we have to load balance multiple contact IP servers and we were trying to use DNS SRV for that. We haven't seen any success in this, it does load balance when both are up. But as soon as only one server is up, SER keeps trying to reach the first one which is down and never tries to reach the second one. Ultimately the transaction times out. Has anybody had success in implementing this kind of scenario for load balancing with DNS SRV? Is anything else besides using DNS SRV an option? Thanks in advance!! __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

RE: [Serusers] SER Load balancing
by Rose, John 19 Sep '05

19 Sep '05

Marcello, Here is one method that I have been looking at - has anyone tried something similar? # --- Rewrite 50/50 -------------------------------- if (uri=~"test") { if (is_gflag("1")) { reset_gflag("1"); rewriteuri("sip:test@10.1.1.1"); } else { set_gflag("1"); rewriteuri("sip:vsd@10.1.1.2"); }; t_on_failure("1"); } # -------------------------------------------------- Thanks, John Rose -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marcello Lupo Sent: Saturday, September 17, 2005 7:38 AM To: serusers(a)lists.iptel.org Subject: [Serusers] SER Load balancing Hi to all, did you have suggestions how to accomplish to load balancing load of SER between 2 machine running both SER and DB to check routing entries?? I was thinking to use LVM but i don't know how to let the same call messages to go to the same machine. There should be something line an LVM module able to check the CallID field and keep track of it. May be some other ways?? Using DNS records or other? Another question is if there is a searchable archive of this Mailing list. Thanks, Bye, Marcello _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] MySQL socket file
by John, Paul 19 Sep '05

19 Sep '05

How can SER be made to look for the MySQL socket file in /var/run/mysql/mysql.sock instead of /var/lib/mysql/mysql.sock? Paul

1 0

[Serusers] 5060
by Sebastian Kühner 19 Sep '05

19 Sep '05

Hi, I notice that ser ignores all "REGISTER"s that come from a special server with the port 5060. All other ports are accepted. What could be the problem? Sebastian

1 0

[Serusers] MySQL 4.1.10a
by John, Paul 19 Sep '05

19 Sep '05

I'm having trouble with MySQL starting when the socket file is located in /var/run/mysql/mysql.sock instead of the default location /var/lib/mysql/mysql.sock. I have changed the owner and permission of the new directory, but I still get the following error in the log. I don't have another mysqld server running on this machine. 050919 12:24:11 [ERROR] Can't start server : Bind on unix socket: Permission denied 050919 12:24:11 [ERROR] Do you already have another mysqld server running on socket: /var/run/mysql/mysql.sock ? 050919 12:24:11 [ERROR] Aborting Paul

1 0

[Serusers] ForwardNoAnswer
by Ramin Nikaeen 19 Sep '05

19 Sep '05

Dear Friends, Has anyone attempted implementing the "Forward On No Answer" service with SER who can give me some pointers. I greatly appreciate your help Thanks Ramin rnikaeen(a)goldline.net

2 1

[Serusers] avpops and register
by Rodrigue ESTEVE 19 Sep '05

19 Sep '05

Hi, I'm trying to do the following things using the avpops: When I receive a Register Message from uri 666@.., I'd like to register the uri 1000@... and 2000@... at the same time. I'd like to do this also for uri 777, to register 778 and 779 when 777 triggers a Register Message. So I decided to use the SQL Table below to achieve this: Initnumber: Varchar(64) Attribute: Varchar(64) Type: INT (default to 0) Associated: Varchar(64) Initnumber attribute type associated 666 register 0 1000 666 register 0 2000 777 register 0 778 777 register 0 779 In Ser.cfg, I put this: modparam("avpops","avp_url","mysql://ser:heslo@IP/telephone") modparam("avpops", "avp_table", "associate") modparam("avpops","username_column","initnumber") modparam("avpops","attribute_column","attribute") modparam("avpops","type_column","type") modparam("avpops","value_column","associated") if (method=="REGISTER") { # Authentication Challenge Challenge #Test of AVPs log(1, "LOG: AVP REGISTRATION"); avp_db_load("$from ","s:register"); #to display avp in memory avp_print(); #fork registration using new avp value to register avp_pushto("$from","s:register/g"); save("location"); break; }; What I expect this to do is to get the values "associated" with "Initnumber", using avp_db_load and to fork the registration process using avp_pushto to register the numbers in the associated column. I use avp_print to see the avp loaded in memory, but I see nothing in the syslog. Moreover nothing happens, ie only the uri which initiates the Register message is registered on SER :'( Can somebody help me please, Rodrigue

1 0

R: [Serusers] fields To/From
by Piccari Gianluca 19 Sep '05

19 Sep '05

ok ): i'll performed that ugly thing named "changing the source code". Good for me (; -----Messaggio originale----- Da: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]Per conto di Piccari Gianluca Inviato: venerdì 16 settembre 2005 13.10 A: serusers(a)lists.iptel.org Oggetto: [Serusers] fields To/From Hi all, i have a big problem... i need to parser information in Header field From() and eventually add or delete part of info in Header field TO. For istance: if i(n From field there is '55') add ('55' in To field). I don't have familiar to SER (i have started to study it last week (: ) and i likr e to know if it is possible to do it using SER's method without change the source code .... thanks for your help _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] cisco dial-peer
by Nhadie 19 Sep '05

19 Sep '05

Hi List, Don't know if this is the right list for this but I hope someone can help me: I have a SIP server when it starts on a certain string e.g. 123 it will forward it to gateway else it will forward to gateway 2. Gateway 1 will be a cisco 1760 with and FXO port, so whe I dial 12345678 it should forwar it to the Cisco 1760, using ngep I can see that it's sending the invite to the CISCO 1760 but I don't know what typre of dial-pper to define in order for th 1760 to respond, also what typre of dial-peer would forward it to the FXO port and eventually call PSTN. TIA Regards, nhadie

1 0

[Serusers] Error when make mysql in ser-0.9.3
by Alberto 19 Sep '05

19 Sep '05

Have you resolve your problem?

1 0

Re: [Serusers] Speeddial
by Aisling O'Driscoll 19 Sep '05

19 Sep '05

Thanks Ramona. > >---- Original Message ---- >From: ramona(a)voice-system.ro >To: ashling.odriscoll(a)cit.ie >Subject: Re: [Serusers] Speeddial >Date: Thu, 15 Sep 2005 20:10:56 +0200 > >>Aisling wrote: >> >>> Hello. >>> >>> I would really appreciate if someone could show me an example of >how a >>> speed-dialing entry should be added to the ser mysql database? >>> >>> E.g. If user 2092 dials 3 it rings user 3500. >>> >>> I tried : insert into usr_preferences (username, domain, >sd_username, >>> sd_domain, new_uri) values (2092, x.x.x.x, 3500, x.x.x.x, >3) >>> >>Hello Aisling, >> >>the entry should be (2092, x.x.x.x, 3, x.x.x.x, >sip:3500@x.x.x.x). >> >>Regards, >>Ramona >> >>-------------------Legal >Disclaimer--------------------------------------- >> >>The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >> >> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

4 4

[Serusers] PSTN Gateway and caller-id
by Luca Corti 19 Sep '05

19 Sep '05

Hello, I've setup a ser proxy/registrar which forwards PSTN calls to an AS5350. The AS5350 also redirects PSTN calls to the ser box, which associates incoming called numbers with sip accounts through associations in the SQL DB. My problem is that the AS doesn't know about these associations when forwarding calls from sip to PSTN. I'd like to set the callerid on the outgoing calls to be the PSTN number associated with the sip account for incoming calls. thanks -- Luca Corti PGP Key ID 1F38C091

3 2

[Serusers] serweb. DB Error: connect failed
by Alberto 19 Sep '05

19 Sep '05

Hello, When I try to connect with admin user or other user account I get follow error: DB Error: connect failed Bad username or password I can connect with MySQL Administrator (windows) to server (freeBSD). The server is a freeBSD 5.4 + MySQL 4.1.14 + apache 2.0.54 + PHP 4.4.0 + ser-0.9.3 + serweb-0.9.3 For example, when I try to connect with administrator account I have tried to enter: username: admin(a)192.168.0.2 or admin or admin@sip or admin(a)teching.net password: heslo I do not understand what she is happening. In the 'subscriber' table: +----------+-----------+-------------+ | username | password | domain | +----------+-----------+-------------+ | admin | heslo | sip | | 1000 | 123454321 | teching.net | | 2000 | 12345 | teching.net | | 3000 | 12345 | teching.net | | 4000 | 12345 | teching.net | | 5000 | 12345 | teching.net | | 4000 | 12345 | 192.168.0.2 | +----------+-----------+-------------+ 7 rows in set (0.00 sec) and 'domain' table: +-------------+---------------------+ | domain | last_modified | +-------------+---------------------+ | teching.net | 2005-09-16 11:01:21 | | 192.168.0.2 | 2005-09-16 11:01:21 | | sip | 2005-09-19 13:55:21 | +-------------+---------------------+ 3 rows in set (0.00 sec) thanks, -- Alberto

1 0

[Serusers] send receive sms via http
by saiful bahri 19 Sep '05

19 Sep '05

hi all, is there any plan or work carry out to implement send/receive sms via http interface instead only via gsm modem? regards, saifulbahri.

1 0

Re: [Serusers] SER Load balancing
by Samuel Osorio Calvo 19 Sep '05

19 Sep '05

www.google.com might help you....;) site:iptel.org LVM "load balancing" Samuel. Unclassified. >>> "Marcello Lupo" <lupo(a)itspecialist.it> 09/17/05 03:37PM >>> Hi to all, did you have suggestions how to accomplish to load balancing load of SER between 2 machine running both SER and DB to check routing entries?? I was thinking to use LVM but i don't know how to let the same call messages to go to the same machine. There should be something line an LVM module able to check the CallID field and keep track of it. May be some other ways?? Using DNS records or other? Another question is if there is a searchable archive of this Mailing list. Thanks, Bye, Marcello _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] LDAP authentication (without Radius)
by Samuel Osorio Calvo 19 Sep '05

19 Sep '05

If I don't recall wrong, there was an LDAP module for SER developed in some university/research center from Switzerland, and I am pretty sure you could dopwnload it....just try to google, because I haven't got internet access right now... Samuel. Unclassified. >>> "Arek Bekiersz" <sip(a)perceval.net> 09/19/05 10:17AM >>> Hi Atle, all, It is not only my decision, as I did it as part of my normal job. I will ask management, let you know a.s.a.p. Arek, ----- Original Message ----- From: "Atle Samuelsen" <clona(a)camaro.no> To: "Arek Bekiersz" <sip(a)perceval.net> Cc: <serusers(a)lists.iptel.org> Sent: Monday, September 19, 2005 10:03 AM Subject: Re: [Serusers] LDAP authentication (without Radius) > Hi Arek, > > Why dont you give the LDAP module back to the comunity, and get it into > the exprimental branch. > > -Atle > > * Arek Bekiersz <sip(a)perceval.net> [050919 09:51]: >> Hello, >> >> >> I went through the same problem. Used Radius + LDAP for a while. >> >> Then only option for me was to write my own SER module, to perform digest >> authentication, lookup, group membership checking and so on, in LDAP. It >> is >> not so complicated though, can be really done in few days (ser 0.8.12) >> >> Works fine since then. Now I'm building more and more solutions for SER >> (including lcr and sdp manipulation, according to user profiles) based on >> LDAP. I use MySQL only for ser location register - that's all. >> >> >> Arek, >> >> >> >> ----- Original Message ----- >> From: "Felipe Louback" <louback(a)gmail.com> >> To: <serusers(a)lists.iptel.org> >> Sent: Thursday, September 15, 2005 7:59 PM >> Subject: [Serusers] LDAP authentication (without Radius) >> >> >> >I'm working in a VoIP solution here in my university and we have a >> >wide LDAP base and I am thinking about using it for >> > >> >I found on the internet Ldap/Ldaps modules for SER >> >(http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldaps), >> >but I am afraid that I could only use them for lookups and not direct >> >authentication. >> > >> >I guess I could use radius for authentication but I dont know if >> >installing a radius server "just" for this would be worth the effort. >> > >> >Does anyone know if there is a way to authenticate users in a LDAP >> >server without using Radius? I mean, could I use SER directly with >> >LDAP without using RADIUS? >> > >> >Regards, >> > >> >Felipe _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] Handoff to PSTN
by Iqbal 19 Sep '05

19 Sep '05

Hi I need to see the ser.cfg, or to make life easier use one from onsip.org, which is a good little get started guide, and will connect to pstn, u just pretty much cut and paste the config make a few changes and away you go iqbal Colin Jordan wrote: >I do not too much about this, can please let me know where and in what part >of the ser.cfg it will be? > > >-----Original Message----- >From: Iqbal [mailto:iqbal@gigo.co.uk] >Sent: Friday, September 16, 2005 12:07 PM >To: cjordan(a)econetwireless.com; olivier.taylor(a)gmail.com >Cc: serusers(a)lists.iptel.org >Subject: RE: [Serusers] Handoff to PSTN > > >where u route your INVITE > >Iqbal > >On 9/16/2005, "Colin Jordan" <cjordan(a)econetwireless.com> wrote: > > > >>Where should I insert it in the code below? >> >> >> >># attempt handoff to PSTN >> >>if (uri=~“^sip:9[0-9]*@mydomain.com”) { >> >>log(“Forwarding to PSTN\n”); >> >>t_relay_to( “192.168.0.2”, “5060”); >> >>break; >> >>}; >> >> >> >> >> >> _____ >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 4:00 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>prefix("666") >> >>-----Message d'origine----- >>De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la >>part de Colin Jordan >>Envoyé : jeudi 15 septembre 2005 15:19 >>À : 'Olivier Taylor' >>Cc : serusers(a)lists.iptel.org >>Objet : RE: [Serusers] Handoff to PSTN >> >>The PSTN has given me a prefix of 666# to dial before the number, where can >>I insert this so that it will allow me to make a call? >> >> >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 2:58 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>module tm >> >>-----Message d'origine----- >>De : Colin Jordan [mailto:cjordan@econetwireless.com] >>Envoyé : jeudi 15 septembre 2005 14:55 >>À : 'Olivier Taylor' >>Cc : serusers(a)lists.iptel.org >>Objet : RE: [Serusers] Handoff to PSTN >> >>t_relay_to( “192.168.0.2”, “5060”); >> >> >> >> >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 2:50 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>what's on the line 143??? >> >>-----Message d'origine----- >>De : Colin Jordan [mailto:cjordan@econetwireless.com] >>Envoyé : jeudi 15 septembre 2005 14:46 >>À : 'Olivier Taylor' >>Cc : serusers(a)lists.iptel.org >>Objet : RE: [Serusers] Handoff to PSTN >> >>The line 143 points to the script below. What modules will it be looking >>for? >> >> >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 2:33 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>please reply to all >> >> >> >>u have a module not loaded, the one involved on line 143 >> >> >> >> >> >>Olivier >> >> >> >>-----Message d'origine----- >>De : Colin Jordan [mailto:cjordan@econetwireless.com] >>Envoyé : jeudi 15 septembre 2005 14:28 >>À : 'Olivier Taylor' >>Objet : RE: [Serusers] Handoff to PSTN >> >>It gives me the following message: >> >>_______________________________________________________ >> >> >> >>Parse error (143,40-41): unknown command missing loadmodule? >> >>_______________________________________________________ >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 1:21 PM >>To: 'Colin Jordan'; serusers-request(a)lists.iptel.org >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>just have a look at /var/log/messages u will find the line number where >> >> >the > > >>error occurs >> >> >> >>Olivier >> >>-----Message d'origine----- >>De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la >>part de Colin Jordan >>Envoyé : jeudi 15 septembre 2005 12:33 >>À : serusers-request(a)lists.iptel.org >>Cc : serusers(a)lists.iptel.org >>Objet : [Serusers] Handoff to PSTN >> >> >> >>I am not able to restart ser if I put he code below in the ser.cfg to >>handoff calls to PSTN. >> >> >> >># attempt handoff to PSTN >> >>if (uri=~“^sip:9[0-9]*@mydomain.com”) { >> >>log(“Forwarding to PSTN\n”); >> >>t_relay_to( “192.168.0.2”, “5060”); >> >>break; >> >>}; >> >> >> >> >> >>Please help!!!! >> >>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. >> >>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. >> >>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. >> >>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. >> >>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. >> >> >> >> >> >> >> > > >. > > >

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Re: [Serusers] No VM module and IVR plugin
by Nicky 19 Sep '05

19 Sep '05

Hi, thank you for your reply. as I need to have multiple ppl to access this sems and i understand from the readme that it will be better to use socket over fifo for more users. The ser.cfg file does not take the variable "socket", I have tested on the unix_socket, it works. Is this the correct vairable to assign? Do you know how can I alllow sems to have multiple users to access as I am having this error:~ Error : stop (AmThread.cpp:99) : pthread_detach failed with code ENINVAL : thread already in detached state. Please help. regards, nicky ----- Original Message ----- From: Gareth C. Fowler To: Nicky Sent: Friday, September 16, 2005 5:46 PM Subject: RE: [Serusers] No VM module and IVR plugin It should auto-create this file, check the socket file location in the ser configuration. You could also try the command: # touch /tmp/ser_socket Any particular reason for using the socket over fifo? Hope this helps. Rgds. G. -----Original Message----- From: Nicky [mailto:nicky@caliber.com.sg] Sent: 16 September 2005 09:34 To: Gareth C. Fowler Subject: Re: [Serusers] No VM module and IVR plugin Hi G. Thank you so much for your helps. I have another problem which I would like to use the ser_socket but in my /tmp there isnt any related ser_socket file. Do you know how can i get this file? Please help...thanks... Regards, Nicky ----- Original Message ----- From: Gareth C. Fowler To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Thursday, September 15, 2005 10:24 PM Subject: RE: [Serusers] No VM module and IVR plugin The IVR module is only available in the SEMS development version, and can be downloaded @ http://www.iptel.org/sems/sems_devel_snapshot_190505.tar.bz2 Sadly they took out the VM module in version 0.9.0 onwards, and it's now in the tm module. Instead of using vm you have to use t_write_req (fifo), and t_write_unix (socket). Hope this helps! G. -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Nicky Sent: 15 September 2005 13:19 To: sems(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: [Serusers] No VM module and IVR plugin Dear all, Please help. I am trying to download Sip_router_rel_0_8_12 and Sems version ser_rel_0_8_12 from the CVS but in the sems there is missing module for IVR. Then I try to download 0.9.0 version for both but there is missing VM module in Ser which is required to link to Sems. Please advice if I would like to use 0.9.0 version, without VM module how can link to SEMS and why in the sems version ser_rel_0_8_12 the IVR plugin is missing and where can I find it. Thanks in advance.... Regards, Nicky -------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Can connect to server but can't call
by Alessio Rosi 18 Sep '05

18 Sep '05

Hi, I'm stting up ser for my home (local network). I'm using MySql registration I can login whit a softphone (Xlite) but I'm not able to call. The answer is : Call failed: 404 Not Found This is my ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ alias=sip.eraclito.it alias=eraclito.it alias=10.10.10.7 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("sip.eraclito.it", "subscriber")) { # www_challenge("sip.eraclito.it", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ********************************** This is wath ngrep catch (note that 81.208.34.102 is the public IP of my router/firewall) U 10.10.10.5:5060 -> 10.10.10.7:5060 INVITE sip:admin@sip.eraclito.it SIP/2.0..Via: SIP/2.0/UDP 81.208.34.102:50 60..From: arfio <sip:arfiodefarro@sip.eraclito.it>;tag=139597290..To: <sip: admin(a)sip.eraclito.it>..Contact: <sip:arfiodefarro@81.208.34.102:5060>..Cal l-ID: 89ABA3C9-01C8-4B06-8847-4C322A9E27FA@10.10.10.5..CSeq: 11480 INVITE.. Content-Type: application/sdp..Content-Length: 272....v=0..o=arfiodefarro 7 787500 7787500 IN IP4 81.208.34.102..s=X-Lite..c=IN IP4 81.208.34.102..t=0 0..m=audio 8000 RTP/AVP 4 0 8 3 101..a=rtpmap:4 G723/8000..a=rtpmap:0 pcmu/ 8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:101 telephone-eve nt/8000..a=fmtp:101 0-15.. # U 10.10.10.7:5060 -> 10.10.10.5:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 81.208.34.102:5060;received=10.1 0.10.5..From: arfio <sip:arfiodefarro@sip.eraclito.it>;tag=139597290..To: < sip:admin@sip.eraclito.it>;tag=b27e1a1d33761e85846fc98f5f3a7e58.05fe..Ca ll- ID: 89ABA3C9-01C8-4B06-8847-4C322A9E27FA@10.10.10.5..CSeq: 11480 INVITE..Se rver: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0..Warning: 392 10.10.10.7:5060 "Noisy feedback tells: pid=23852 req_src_ip=10.10.10.5 req_src_port=5060 in_uri=sip:admin@sip.eraclito.it out_uri=sip:admin@sip.e raclito.it via_cnt==1".... # U 10.10.10.5:5060 -> 10.10.10.7:5060 ACK sip:admin@sip.eraclito.it SIP/2.0..Via: SIP/2.0/UDP 81.208.34.102:5060; received=10.10.10.5..From: arfio <sip:arfiodefarro@sip.eraclito.it>;tag=139 597290..To: <sip:admin@sip.eraclito.it>;tag=b27e1a1d33761e85846fc98f5f3a7e5 8.05fe..Contact: <sip:arfiodefarro@81.208.34.102:5060>..Call-ID: 89ABA3C9-0 1C8-4B06-8847-4C322A9E27FA@10.10.10.5..CSeq: 11480 ACK..Max-Forwards: 70..C ontent-Length: 0....

1 0

[Serusers] From header
by Iqbal 18 Sep '05

18 Sep '05

Hi The From request in REGISTER and INVITE, what should be in the domain segment of it, should it be, the IP address of the UA, the proxy or what, I seem to be getting conflicting results with a quintum, and wanted to double check Iqbal

5 9

Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses?
by Charles Wang 18 Sep '05

18 Sep '05

Mike: Before I test these two interfaces(Private and Public), I have only a public interface on my SER proxy. My NATed clients are XLite or any SIP IP phones or SIP gateways. They work fine with SER and each other. When I try to make these clients register from private interface, the problem happens. So I don't think it is the problem of NAT functions at XLite or Gateways. On 9/18/05, Mike Williams <mwilliams(a)etc1.net> wrote: > I was having problems with NAT myself, and found this. I thought it sounded like your problem. > > http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.h… > > ---Mike > > 5.1. SIP with NAT or Firewalls ( Back to Tutorials Page ) > > 1.1. Problem Description: > > > Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. > > The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange both call legs (incoming and outgoing audio). > > Most firewalls/NATs are unable to link the signalling protocol packets with the audio packets and are in some cases unable to tell where to send the audio to. > > When making a call, everything will seem to go normal, caller id will get passed, ringing will start, you can pick up and hangup the call, but no audio in one or both directions. > > > > -----Original Message----- > From: Charles Wang [mailto:lazy.charles@gmail.com] > Sent: Sat 9/17/2005 2:40 AM > To: Mike Williams; serusers(a)iptel.org > Subject: Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses? > > Hi, Mike: > the default gateway is 192.168.11.254 not 192.168.11.1. > So i dont think the flow is 192.168.11.2 to 192.168.11.1. > > On 9/17/05, Mike Williams <mwilliams(a)etc1.net> wrote: > > I believe your problem is simple. With the SIP protocol, you are sending > > the streams like this: > > > > 192.168.11.2 -> 192.162.11.1 -> 221.21.X.X > > > > After you answer, the clients negotiate for RTP traffic, and try to send > > data directly from 192.168.11.2 to 221.21.X.X, not using the SIP server. > > You are probably having problems actually routing the data (trying > > pinging the 221.21.X.X box from your 192.168.11.2 client) or you're > > having NAT issues. Is far as I know, you must have a direct route from > > the caller to the callee to pass RTP streams; you can't proxy them > > through the SIP server. > > > > Good luck, and let me know if you have any more questions. > > > > Mike Williams (mwilliams(a)etc1.net) > > > > Charles Wang wrote: > > > > >On 9/13/05, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > > > > >>Hi, ALL: > > >> > > >>I use ser + mediaproxy + PSTN support, and my ser with two interfaces. > > >>One is public IP address such as 211.21.xxx.xxx. > > >>Another one is private IP address such as 192.168.11.1. > > >> > > >>And I use XLite (192.168.11.2) register to SER's private interface(via > > >>HUB only). > > >>It can register sucessfully. > > >> > > >>But when I make a call to PSTN with this XLite, the callee rings and I > > >>answer it. > > >>I can not hear any sounds from each side. > > >> > > >>I try to register another XLite(192.168.11.3), and make a call to > > >>another private XLite(192.168.11.2). I can hear rings but it is still > > >>no any sounds from each side. > > >> > > >>Can anyone tell me what it happens? > > >> > > >>Best Regard > > >>Charles > > >> > > >> > > >> > > > > > > > > > > > > > > > > > -- > > Best Regards > Charles > > > -- Best Regards Charles -- Best Regards Charles

1 0

[Serusers] SER Load balancing
by Marcello Lupo 17 Sep '05

17 Sep '05

Hi to all, did you have suggestions how to accomplish to load balancing load of SER between 2 machine running both SER and DB to check routing entries?? I was thinking to use LVM but i don't know how to let the same call messages to go to the same machine. There should be something line an LVM module able to check the CallID field and keep track of it. May be some other ways?? Using DNS records or other? Another question is if there is a searchable archive of this Mailing list. Thanks, Bye, Marcello

1 0

Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses?
by Charles Wang 17 Sep '05

17 Sep '05

Hi, Mike: the default gateway is 192.168.11.254 not 192.168.11.1. So i dont think the flow is 192.168.11.2 to 192.168.11.1. On 9/17/05, Mike Williams <mwilliams(a)etc1.net> wrote: > I believe your problem is simple. With the SIP protocol, you are sending > the streams like this: > > 192.168.11.2 -> 192.162.11.1 -> 221.21.X.X > > After you answer, the clients negotiate for RTP traffic, and try to send > data directly from 192.168.11.2 to 221.21.X.X, not using the SIP server. > You are probably having problems actually routing the data (trying > pinging the 221.21.X.X box from your 192.168.11.2 client) or you're > having NAT issues. Is far as I know, you must have a direct route from > the caller to the callee to pass RTP streams; you can't proxy them > through the SIP server. > > Good luck, and let me know if you have any more questions. > > Mike Williams (mwilliams(a)etc1.net) > > Charles Wang wrote: > > >On 9/13/05, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > >>Hi, ALL: > >> > >>I use ser + mediaproxy + PSTN support, and my ser with two interfaces. > >>One is public IP address such as 211.21.xxx.xxx. > >>Another one is private IP address such as 192.168.11.1. > >> > >>And I use XLite (192.168.11.2) register to SER's private interface(via > >>HUB only). > >>It can register sucessfully. > >> > >>But when I make a call to PSTN with this XLite, the callee rings and I > >>answer it. > >>I can not hear any sounds from each side. > >> > >>I try to register another XLite(192.168.11.3), and make a call to > >>another private XLite(192.168.11.2). I can hear rings but it is still > >>no any sounds from each side. > >> > >>Can anyone tell me what it happens? > >> > >>Best Regard > >>Charles > >> > >> > >> > > > > > > > > > -- Best Regards Charles

1 0

[Serusers] SER proxy and SER+SEMS media server
by Teemu Harju 17 Sep '05

17 Sep '05

Hi, I'm implementing following kind of system. I have two separate machines where I've installed SER on. The other machine has also SEMS installed. Like you probably already figured out other is used as a plain proxy/registrar and the other as a sort of UAS serving voicemail, conference etc with the help of SEMS. Now my question is, what is the best way to forward messages from proxy/registrar to the media server? I've done the forwarding according to the examples I've found and it uses rewritehostport() to forward messages to voicemail. Voicemail needs to get the email address from mysql database located in proxy. This is no problem connecting to the database, but isn't it so that when I've rewritten the Request-URI that the voicemail cannot find the email from database beacause it's rewritten and has the media servers host part in the URI. How have you guys implemented this? I would also need other SEMS plug-ins like conference, so I would also need to differentiate those requests also. Could I rewerite the Request-URI like voicemail@mediaserver? Can the voicemail still find out who was I actually trying to call to since the URI is rewritten? Is there difference in implementing the media server with asterisk? Wouldn't I need then two different databases? One in asterisk and one in SER. Thanks... -- Teemu Harju http://www.teemuharju.net

2 2

[Serusers] Building SEMS on Mac OS..share your experience..
by Amos Nungu 16 Sep '05

16 Sep '05

Hi, I am Installing SER/SEMS on Mac. SER is running fine Building SEMS give some error but it does build to the end, and I can run "make Install" BUT can't start SEMS.. error below ser:~ root# sems bash: /usr/local/sbin/sems: cannot execute binary file and compilation error is.. #make all cd gsm-1.0-pl10; make ./lib/libgsm.a make[5]: `lib/libgsm.a' is up to date. cc -o gsm.so gsm.o gsm-1.0-pl10/lib/libgsm.a -shared powerpc-apple-darwin8-gcc-4.0.0: unrecognized option `-shared' /usr/bin/ld: gsm.o bad magic number (not a Mach-O file) /usr/bin/ld: warning empty table of contents: gsm-1.0-pl10/lib/libgsm.a (can't load from it) collect2: ld returned 1 exit status make[4]: *** [gsm.so] Error 1 make[3]: [all] Error 2 (ignored) make[4]: Nothing to be done for `deps'. make[5]: Nothing to be done for `all'. make[4]: Nothing to be done for `deps'. make[4]: `../../lib/apps/isdngw.so' is up to date. make[4]: Nothing to be done for `deps'. g++ -o ivr.so Ivr.o IvrDtmfDetector.o IvrEvents.o IvrMediaHandler.o IvrPython.o -lm -ldl -lsocket -lnsl -lpthread -shared -ldl -lpthread -lutil -lm -Xlinker --export-dynamic -L/usr/lib/python2.3/config -lpython2.3 /usr/lib/python2.3/lib-dynload/time.so powerpc-apple-darwin8-g++-4.0.0: unrecognized option `-shared' /usr/bin/ld: unknown flag: --export-dynamic collect2: ld returned 1 exit status make[4]: *** [ivr.so] Error 1 make[3]: [all] Error 2 (ignored) make[4]: Nothing to be done for `deps'. cc -fPIC -I ../../amci -Wall -I./lame-3.96/include -c mp3.c -o mp3.o mp3.c:37:18: error: lame.h: No such file or directory mp3.c: In function 'MP3_create': mp3.c:88: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:88: error: (Each undeclared identifier is reported only once mp3.c:88: error: for each function it appears in.) mp3.c:88: error: 'gfp' undeclared (first use in this function) mp3.c:93: warning: implicit declaration of function 'lame_init' mp3.c:98: warning: implicit declaration of function 'lame_set_errorf' mp3.c:99: warning: implicit declaration of function 'lame_set_debugf' mp3.c:100: warning: implicit declaration of function 'lame_set_msgf' mp3.c:102: warning: implicit declaration of function 'lame_set_num_channels' mp3.c:103: warning: implicit declaration of function 'lame_set_in_samplerate' mp3.c:104: warning: implicit declaration of function 'lame_set_brate' mp3.c:105: warning: implicit declaration of function 'lame_set_mode' mp3.c:106: warning: implicit declaration of function 'lame_set_quality' mp3.c:108: warning: implicit declaration of function 'id3tag_init' mp3.c:109: warning: implicit declaration of function 'id3tag_set_title' mp3.c:110: warning: implicit declaration of function 'lame_init_params' mp3.c:118: warning: control reaches end of non-void function mp3.c: In function 'MP3_destroy': mp3.c:122: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:122: error: parse error before ')' token mp3.c: In function 'Pcm16_2_MP3': mp3.c:138: warning: implicit declaration of function 'lame_encode_buffer' mp3.c:138: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:138: error: parse error before ')' token mp3.c:143: warning: control reaches end of non-void function mp3.c: In function 'MP3_close': mp3.c:224: warning: implicit declaration of function 'lame_encode_flush' mp3.c:224: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:224: error: parse error before ')' token mp3.c:228: warning: implicit declaration of function 'lame_mp3_tags_fid' mp3.c:228: error: parse error before ')' token mp3.c:221: warning: unused variable 'mp3buffer' mp3.c:229: warning: control reaches end of non-void function make[4]: *** [mp3.o] Error 1 make[3]: [all] Error 2 (ignored) . . . make[4]: Nothing to be done for `deps'. make[4]: `../../lib/apps/voicemail.so' is up to date. make[4]: Nothing to be done for `deps'. make[4]: `../../lib/audio/wav.so' is up to date. Regards

1 0

[Serusers] Authentication
by Sebastian Kühner 16 Sep '05

16 Sep '05

Hello, I want to authenticate the users that wants to make a PSTN call on every INVITE. I have one user that can't make PSTN calls, because the message "407 Proxy Authentication Required" goes back to the wrong port (the IP is correct). I checked it with ngrep. Has anyone of you an idea how could I fix that problem? Thanks! Sebastian

2 1

[Serusers] Re: 400 Bad Request
by Alan Chiu 16 Sep '05

16 Sep '05

This INVITE came from a client device but not sure why SER 0.8.14 (i386/linux) would reject it with 400 Bad Request and there's no reason given. Any insights would help. ***************************************************** Session Initiation Protocol Request line: INVITE sip:4519@10.100.0.111:5060;transport=udp SIP/2.0 Message Header To: <sip:4519@10.100.0.222> From: "PHONE TEST"<sip:1111@172.31.1.1>;tag=5b4ae231 Via: SIP/2.0/UDP 172.31.1.20:5060;branch=z9hG4bK-d87543-ffe1a8337da8f24e89ef-1-cHA5ZGU5Mz E0ZWU3YTE3YjUzNjgxYg..-d87543- Call-ID: 4132a9661e815dd517bec43512d032db CSeq: 267297881 INVITE Contact: <sip:4AmIBQqlz52H8OKUlL_xUG_l--zuAAngOdmZbN4wa1rlMGPDB0hIkU7H3mat-JyuSGS Vy4.@172.31.1.20:5060> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Date: Thu, 15 Sep 2005 16:34:32 GMT User-Agent: Server Authorization: Digest username="testuser(a)domains.com",realm="companyX.com",algorithm="MD5",uri ="sip:4519@10.100.0.222",no nce="432993119128c2b1a32ca43b9436e2fe0bca62f0",response="e42bc4a17da0282 552265396c0eb461e",opaque="" Content-Length: 243 Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 27663 1963582334 IN IP4 172.31.1.20 Owner Username: root Session ID: 27663 Session Version: 1963582334 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 172.31.1.20 Session Name (s): session Connection Information (c): IN IP4 172.31.1.20 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 172.31.1.20 Time Description, active time (t): 0 0 Session Start Time: 0 Session Start Time: 0 Media Description, name and address (m): audio 10492 RTP/AVP 0 8 101 Media Type: audio Media Port: 10492 Media Proto: RTP/AVP Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Attribute Value: 101 0-16 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 101 telephone-event/8000 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Session Initiation Protocol Status line: SIP/2.0 400 Bad Request Message Header To: <sip:4519@10.100.0.222>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6a43 From: "PHONE TEST"<sip:1111@172.31.1.1>;tag=5b4ae231 Via: SIP/2.0/UDP 172.31.1.20:5060;branch=z9hG4bK-d87543-ffe1a8337da8f24e89ef-1-cHA5ZGU5Mz E0ZWU3YTE3YjUzNjgxYg..-d87543- Call-ID: 4132a9661e815dd517bec43512d032db CSeq: 267297881 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 10.100.0.111:5060 "Noisy feedback tells: pid=6024 req_src_ip=172.31.1.20 req_src_port=5060 in_uri=sip:4519@10. 100.0.111:5060;transport=udp out_uri=sip:4519@10.100.0.111:5060;transport=udp via_cnt==1" *****************************************************

1 0

[Serusers] Does SER works on TWO network interfaces with public and private IP addresses?
by Charles Wang 16 Sep '05

16 Sep '05

Hi, ALL: I use ser + mediaproxy + PSTN support, and my ser with two interfaces. One is public IP address such as 211.21.xxx.xxx. Another one is private IP address such as 192.168.11.1. And I use XLite (192.168.11.2) register to SER's private interface(via HUB only). It can register sucessfully. But when I make a call to PSTN with this XLite, the callee rings and I answer it. I can not hear any sounds from each side. I try to register another XLite(192.168.11.3), and make a call to another private XLite(192.168.11.2). I can hear rings but it is still no any sounds from each side. Can anyone tell me what it happens? Best Regard Charles

1 1

[Serusers] SIP VXML
by Abdul Lateef 16 Sep '05

16 Sep '05

Hi all, Can any suggest me to developing IVR using vxml for SIP Gatekeeper, Is there any SIP Vxml open source available? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

FW: [Serusers] Call forwarding Question (following issue 5)
by Aisling 16 Sep '05

16 Sep '05

Sorry I never attached the messages... -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 16 September 2005 11:31 To: 'Greger V. Teigre'; 'serusers(a)lists.iptel.org' Subject: RE: [Serusers] Call forwarding Question (following issue 5) I have attached a complete ngrep of the messages. There were no errors in the var/log/messages file. Originally 3500 and 5000 were on the same machine so I changed the scenario so that 3500 (Windows messenger) was calling 2092 (BT100 - which is busy) and should be forwarded to 2009 (KPhone, same pc as SER). However it still doesn't forward the call.....Its the last two messages confuse me...I don't understand why a 407 Proxy Authentication required would be sent back to original caller.... My rule in the usr_preferences table of the mysql database is: Username (2092) Attribute (fwdbusy) Value (sip:2009@serveraddress) Any ideas? Many thanks, Aisling. -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: 15 September 2005 18:43 To: Aisling; serusers(a)lists.iptel.org Subject: Re: [Serusers] Call forwarding Question (following issue 5) Yes, we have verified that the configs lack the append_branch you mention. The fix has been done (https://siprouter.onsip.org/trac/changeset/15) and new configs are available through the regular downloads at onsip.org. To your other problem: No clue, Aisling. Sounds very strange indeed. The config should only trigger proxy authentication on an INVITE (not REGISTER). Maybe what you are experiencing is Messenger sending a new INVITE. (It should not, as it should receive a 100 Trying from ser). Only a *complete* ngrep trace will help in understanding what's happening (and any error messages in /var/log/messages with the timestamp for matching) g-) Aisling wrote: > Hi, > > I am testing the call forwarding features demonstrated in Issue 5 of > the > onsip getting started document. I found that blind call transfer > worked > perfectly but fwdbusy & fwdnoanswer gave me errors: > > ERROR: t_forward_nonack: non branched for forwarding > ERROR: w_t_relay(failure mode): forwarding failed > ERROR: sl_reply_error used: I'm terribly sorry, server error occurred. > > On the onsip site I noted that someone else had this problem and it > was > solved by putting append_branch in the fwdbusy and fwdnoanswer > sections in > the failure route. > > Thankfully that fixed those errors. However when I went to test > fwdbusy > again, it doesn't give any errors but still doesn't work. The call > scenario > was as follows: > > Windows Messenger client 3500 ring BT100 2092. 2092 is off the hook > (thereby > sending a 486 busy message) and the call should be forwarded to xlite > client > 5000. > > i.e. 3500 -> 2092(busy) -> 5000 > > The message sequence showed that everything was correct up to 2092 > sending > the 486 busy to SER and then SER sending an ACK back to 2092. But > then SER > sends a 407 proxy authentication required to 3500 which replies with > an > ACK....and that's it... > > Can someone explain why SER would send a 407 Proxy authentication to > the > original caller?...I thought this should only be in response to a > register > message?.... > > Any help appreciated, > Thanks, > Aisling. > > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of this > message is strictly prohibited. The Institute does not guarantee the > security of any information electronically transmitted and is not > liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the > sender nor for any delay in its receipt. > > > > >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 2

Re: [Serusers] SER proxy and SER+SEMS media server
by Samuel Osorio Calvo 16 Sep '05

16 Sep '05

You might try to just forward without modifying the Req-URI with the config file options: t_relay_to_udp(ip, port); OR t_relay_to_tcp(ip, port OR forward(ip,port); where the tuple ip,port corresponds to the SER instance acting as voicemail/conference server. The last option does not require the tm module and does not involve any transaction, Depending on your performance requirements, it should be better to use one or another... Hope it helps, Samuel. Unclassified. >>> Teemu Harju <teemu.harju(a)gmail.com> 09/16/05 03:11PM >>> Hi, I'm implementing following kind of system. I have two separate machines where I've installed SER on. The other machine has also SEMS installed. Like you probably already figured out other is used as a plain proxy/registrar and the other as a sort of UAS serving voicemail, conference etc with the help of SEMS. Now my question is, what is the best way to forward messages from proxy/registrar to the media server? I've done the forwarding according to the examples I've found and it uses rewritehostport() to forward messages to voicemail. Voicemail needs to get the email address from mysql database located in proxy. This is no problem connecting to the database, but isn't it so that when I've rewritten the Request-URI that the voicemail cannot find the email from database beacause it's rewritten and has the media servers host part in the URI. How have you guys implemented this? I would also need other SEMS plug-ins like conference, so I would also need to differentiate those requests also. Could I rewerite the Request-URI like voicemail@mediaserver? Can the voicemail still find out who was I actually trying to call to since the URI is rewritten? Is there difference in implementing the media server with asterisk? Wouldn't I need then two different databases? One in asterisk and one in SER. Thanks... -- Teemu Harju http://www.teemuharju.net

1 0

[Serusers] DTMF
by Iqbal 16 Sep '05

16 Sep '05

Hi I am guessing the ser does nothing with dtmf coming from the pstn side, since it's all in the rtp stream, if is there anyway of "seeing" what ser is doing with it, or what is being sent, I can see the dtmf in asterisk being passed from ser, but if I use another remote gateway, I wanted to be able to see if possible, even using ethreal or whatever Iqbal

1 0

[Serusers] [about SER] 483 too many hops problem
by 송 미진 16 Sep '05

16 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 dear, I'm working on a project and i'm using ser as a sip server. actually i have this problem: every time i send an instant message to a subscriber i get this error message: 483 too many hops i tried to fix it but i couldn't and i don't know what to do ! (My computer connect many computers. Is it reason?!) if any one can help me i'll be very grateful thank you in advance i'm using ser 0.9.3 & here's my code: # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:vlftmd@localhost/ser" alias=sam.cse # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc","db_url","mysql://root:vlftmd@localhost/ser") #modparam("usrloc", "db_mode", 0) modparam("registrar","use_domain",1) # 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","secret","123123123") #modparam("auth_db","db_url","mysql://root:vlftmd@localhost/ser") #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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sam.cse", "subscriber")) { www_challenge("sam.cse", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _________________________________________________________________ 상큼한 만남과 따뜻한 공동체 생활... 지금 MSN 커뮤니티에서 시작하세요! http://groups.msn.com/?pgmarket=ko-kr

2 1

[Serusers] fields To/From
by Piccari Gianluca 16 Sep '05

16 Sep '05

Hi all, i have a big problem... i need to parser information in Header field From() and eventually add or delete part of info in Header field TO. For istance: if i(n From field there is '55') add ('55' in To field). I don't have familiar to SER (i have started to study it last week (: ) and i likr e to know if it is possible to do it using SER's method without change the source code .... thanks for your help

1 0

RE : [Serusers] Handoff to PSTN
by Olivier Taylor 16 Sep '05

16 Sep '05

module tm -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:55 À : 'Olivier Taylor' Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] Handoff to PSTN t_relay_to( 192.168.0.2, 5060); _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 2:50 PM To: 'Colin Jordan' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN what's on the line 143??? -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:46 À : 'Olivier Taylor' Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] Handoff to PSTN The line 143 points to the script below. What modules will it be looking for? _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 2:33 PM To: 'Colin Jordan' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN please reply to all u have a module not loaded, the one involved on line 143 Olivier -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:28 À : 'Olivier Taylor' Objet : RE: [Serusers] Handoff to PSTN It gives me the following message: _______________________________________________________ Parse error (143,40-41): unknown command missing loadmodule? _______________________________________________________ _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 1:21 PM To: 'Colin Jordan'; serusers-request(a)lists.iptel.org Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN just have a look at /var/log/messages u will find the line number where the error occurs Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Colin Jordan Envoyé : jeudi 15 septembre 2005 12:33 À : serusers-request(a)lists.iptel.org Cc : serusers(a)lists.iptel.org Objet : [Serusers] Handoff to PSTN I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~^sip:9[0-9]*@mydomain.com) { log(Forwarding to PSTN\n); t_relay_to( 192.168.0.2, 5060); break; }; Please help!!!! 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. 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. 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. 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.

3 6

[Serusers] Call forwarding Question (following issue 5)
by Aisling 16 Sep '05

16 Sep '05

Hi, I am testing the call forwarding features demonstrated in Issue 5 of the onsip getting started document. I found that blind call transfer worked perfectly but fwdbusy & fwdnoanswer gave me errors: ERROR: t_forward_nonack: non branched for forwarding ERROR: w_t_relay(failure mode): forwarding failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred. On the onsip site I noted that someone else had this problem and it was solved by putting append_branch in the fwdbusy and fwdnoanswer sections in the failure route. Thankfully that fixed those errors. However when I went to test fwdbusy again, it doesn't give any errors but still doesn't work. The call scenario was as follows: Windows Messenger client 3500 ring BT100 2092. 2092 is off the hook (thereby sending a 486 busy message) and the call should be forwarded to xlite client 5000. i.e. 3500 -> 2092(busy) -> 5000 The message sequence showed that everything was correct up to 2092 sending the 486 busy to SER and then SER sending an ACK back to 2092. But then SER sends a 407 proxy authentication required to 3500 which replies with an ACK....and that's it... Can someone explain why SER would send a 407 Proxy authentication to the original caller?...I thought this should only be in response to a register message?.... Any help appreciated, Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 2

[Serusers] ceratins users to one gateway others to another
by Nhadie 16 Sep '05

16 Sep '05

Hi All, Can I have some extensions use gateway 1 and some can use gateway 2? I want to do this for clients that pay premium they have unlimited calls for free on a certain onuntry and others will still go to the per minute basis. Can I set this up on groups? Hope anyone can help me. Thank You Regards, Nhadie

3 4

[Serusers] ser.cfg with PSTN gateway connectivity
by Alberto 16 Sep '05

16 Sep '05

Hi, I've read the SER Getting Started form http://www.onsip.org . When I configure the ser with PSTN Gateway (subject 8 of the guide) I can't register any client (403, Forbidden). I think, the problem is the next line of ser.cfg: if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403","Forbidden"); }; break; }; I'd copied the permissions.allow and permissions.deny to /usr/local/etc/ser and in the .allow file I'd uncomment the last line (ALL:ALL) Any idea? Thanks. -- Alberto

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Re: [Serusers] Distributing "OSP enable SER" binary file w/ source
by Dmitry Isakbayev 16 Sep '05

16 Sep '05

Hi Jan, I am following up on the conversation that you and Vikrant had a short while ago. The OSP interface was implemented as a SER module. It does not require any changes to the core SER or any of its modules. We have been hosting the project @ http://developer.berlios.de/projects/osp-module We would like to move the project to the experimental branch in your CVS. How should we proceed? > > In what form does the come ? Did you implement it as a SER module > > ? If so then we can put it in the experimental branch in CVS. If you > > made some changes to other modules or the core then it would be great if > > you could generate patches against the latest development version. Best Regards, Dmitry > > ----- Original Message ----- > > From: "Jan Janak" <jan(a)iptel.org> > > To: <vmathur(a)transnexus.com> > > Cc: <serusers(a)lists.iptel.org> > > Sent: Friday, August 12, 2005 2:35 PM > > Subject: Re: [Serusers] Distributing "OSP enable SER" binary file w/ > > source > > code > > > > > >> On 12-08-2005 18:18, vmathur(a)transnexus.com wrote: > >> > Hi Jan, > >> > > >> > The code has been written as a SER module. The details of the code are > > available at: > >> > http://developer.berlios.de/projects/osp-module > >> > > >> > Right now the process of compilation requires that you download the OSP > > toolkit source code and complie it. This would build the OSP client > > library > > that the SER would use. You should then download the SER and the OSP > > modules > > and build them. The process is rather lengthy and we were wondering if we > > come up with an easier way of doing this. > >> > > >> > What exactly is the experimental branch in CVS ? > > > >> SER CVS repository contains experimental directory. This is the > >> directory where anyone can put their contribution. It stays there for > >> a while and if it proves to be useful and supported by the author then > >> we move it to the main source tree, giving the author write access to > >> the main tree. > >> > >> Jan. > >> > >> > > > > In what form does the come come ? Did you implement it as a SER module > > ? If so then we can put it in the experimental branch in CVS. If you > > made some changes to other modules or the core then it would be great if > > you could generate patches against the latest development version. > > > > Jan.

2 1

[Serusers] Making SER implement Jabber
by Saleh, Saleh 15 Sep '05

15 Sep '05

Hello Everyone, Can anyone tell me what I need to do next in order for SER version 0.9.3 to use the Jabber Module correctly? This is what I have done so far (on a Red Hat Linux 9 machine): Installed Jabberd 1.4.4 Installed GNU pth-2.0.4 Installed Expat 1.95.8 Installed Libidn ? I have Modified the config file for SER to load the Jabber.so module after the pa.so and tm.so (I'm sure this is where I am stuck, what to do with parameters in config file for SER) This is what I can/cannot do: I can run the jabber server I cannot run the SER with the config file which includes the Jabber.so module load command but I can run it without this command in the config file. I am using MySQL for SER What other files do I need to modify? What information do I need? I'm not very familiar with linux, I am a beginner... Any comments/suggestions/websites/help is GREATLY appreciated! Thanks, Saleh

1 0

[Serusers] ser 0.9.4 crash
by Tulika Pradhan 15 Sep '05

15 Sep '05

i am trying the 0.9.4 release and sometimes ser starts up ok. other times it crashes. here is the 'bt' results. Loaded symbols for /lib/libnss_dns.so.2 #0 0x00b4eec3 in preload_udomain (_c=0x810c898, _d=0xb6178ce8) at udomain.c:302 302 ua.len = strlen(ua.s); (gdb) bt #0 0x00b4eec3 in preload_udomain (_c=0x810c898, _d=0xb6178ce8) at udomain.c:302 #1 0x00b4ab76 in register_udomain (_n=0x0, _d=0xbfedf6ec) at dlist.c:249 #2 0x00133795 in domain_fixup (param=0x810d834, param_no=1) at reg_mod.c:207 #3 0x08070723 in fix_actions (a=0x0) at route.c:238 #4 0x08070845 in fix_actions (a=0x0) at route.c:225 #5 0x080744e4 in fix_rls () at route.c:726 #6 0x0805e33d in main (argc=2, argv=0xbfedf9c4) at main.c:1582 (gdb) any help would be greatly appreciated. regards, tulika

2 1

[Serusers] SER and RFC3325
by Gummadidala, Ravi 15 Sep '05

15 Sep '05

Does SER support RFC3325? Does it understand headers like "Privacy", "P-Preferred-Identity" and "P-Asserted-Identity"? If so, how can we specify the Spec(T) to control SER's behavior? Thanks, -Ravi.

2 1

[Users] how to modify record route header with advertised_address ?
by Gustavo García 15 Sep '05

15 Sep '05

Hi, Ive configured an address with advertised_address. I can see that its used in Via address but not in record route. What could be the reason? Thanks a lot

3 6

[Users] cvs is frozen
by Daniel-Constantin Mierla 15 Sep '05

15 Sep '05

Hello everybody, the cvs head is frozen in preparation for the new release. Please commit only fixes to the C code. Documentation updates are very welcome as well as adjacent tools upgrades. In about two weeks (more or less), a new branch will be created the the cvs head will be unfrozen and open for new features. Cheers, Daniel

3 6

[Serusers] call hunt
by Tulika Pradhan 15 Sep '05

15 Sep '05

i am tring to implement call hunt in ser 0.8.12 (no avpops). there is one problem. i am using exec_dset in the failure route as follows. failure_route[1] { exec_dset("/etc/ser/getnextnumber1 $SIP_OUSER; echo>/dev/null;"); append_branch(); t_on_failure("1"); xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> <%tu>\n"); t_relay(); } after the first failure, the SER sends 408 to the UA (caller) and it gets a busy tone. at this point, the second number of the call hunt group is ringing. but no voice can go through as the caller UA is showing 408 Request Timeout. please help, regards, tulika

1 0

[Serusers] Speeddial
by Aisling 15 Sep '05

15 Sep '05

Hello. I would really appreciate if someone could show me an example of how a speed-dialing entry should be added to the ser mysql database? E.g. If user 2092 dials 3 it rings user 3500. I tried : insert into usr_preferences (username, domain, sd_username, sd_domain, new_uri) values ("2092", "x.x.x.x", "3500", "x.x.x.x", "3") where x.x.x.x = ser address I have the following in my ser.cfg: loadmodule "/usr/local/lib/ser/modules/speeddial.so" modparam("speeddial", "db_url", "mysql://root:password@localhost/ser") modparam("speeddial", "user_column", "userid") modparam("speeddial", "sd_user_column", "short_user") modparam("speeddial", "sd_domain_column", "short_domain") modparam("speeddial", "new_uri_column", "real_uri") if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")){ sd_lookup("speed dial"); }; I got a 404 not found, but its probably because I didn't create the entry properly in the mysql database. If someone could show me an example that would be great. Many thanks. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

[Serusers] SER and rtpproxy in a DMZ
by Holger Moskopp 15 Sep '05

15 Sep '05

Hello, my name is Holger Moskopp and i'm Student at the FH Cologne. At the moment i'm working on my Thesis. I have to build a Firewall with DMZ and a SIP Expressrouter with RTPPROXY. This should look like this: -------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser1.jpg -------------------------------------------------------- I'm in a Subnet of the FH and got on the I' in a Subnetz of the school and have installed on the computer xxx.22 fwclient a SER Registrar with RTPproxy and a Kphone softphone. On the internal SER there is a Kphone registered. (holleinnen). In the DMZ is a SER with rtpproxy. In the FH-Net there is a SER with Radius authentification and two softphones. Phil(a)xxx.73 <mailto:Phil@xxx.73> is registerd at that SER. holleaussen is registerd on another registrar that is not on the picture. If I want to call from holleinnen to phil, everything functions marvelously. The SIP signaling and the RTP-Traffic runs throuhg the DMZ. -------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser2.jpg -------------------------------------------------------- Now to the problem: If I start a call from holleaussen to holleinnen the SIP phase , works perfectly thruh the DMZ. It rings inside and I can assume. After that nothing more happens. With tetereal and etheral I saw that the RTP traffic „wants“ to take the way directly from end to end. -------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser4.jpg -------------------------------------------------------- Do you have an idea what is going wrong? I attached the two ser.cfg files because i think it is a mistake there. I tryed to fix that since 3 days now - but with no success. -------------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser-dmz.txt http://www.ganeymed.de/pixx/fw_ids/ser-innen.txt --------------------------------------------------------------- Here is the relevant Firewallpart: $IPTABLES -N SIPLOG $IPTABLES -I FORWARD -p udp -i $DMZ_ETH --sport 1024:65535 -o $EXTERN_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I FORWARD -p udp -i $EXTERN_ETH --sport 1024:65535 -o $DMZ_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I FORWARD -p udp -i $INTERN_ETH --sport 1024:65535 -o $DMZ_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I FORWARD -p udp -i $DMZ_ETH --sport 1024:65535 -o $INTERN_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I SIPLOG -j LOG --log-prefix "SIPLOG: " $IPTABLES -A SIPLOG -j ACCEPT $IPTABLES -t nat -I PREROUTING -p udp -i $EXTERN_ETH --dport 5060:5062 -j DNAT --to $prox /(That changing only send the packets to prox but prox don#t take them: $IPTABLES -t nat -I PREROUTING -p udp -i $EXTERN_ETH --dport 1024:65535/ /-j DNAT --to prox) / I think a soulution could be to use the statefull iptabels filtering, but I don't like that solution. Thank you and best regards Holger Moskopp

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