- sr-users - kamailio.org

[Serusers] RPID question
by Java Rockx 24 Jan '05

24 Jan '05

Hi All. It seems that append_rpid_hf() in ser-0.9 will not insert the Remote-Party-ID header unless proxy_authorize() is called first. Is this normal behaviour? Regards, Paul __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

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RE: [Serusers] Initialization PDT module problem.
by Ricardo Martinez 24 Jan '05

24 Jan '05

Bogdan, Thanks for the patch. It seems to work ok now. I have a question about the module. I hope that someone can help me. When i receive an INVITE like this : INVITE sip:005622408196@xx.xx.xx.232:5060 the debug is sending : 5(17457) PDT: get_domain_from_hash: required code 56224 is not allocated yet and the a final "Too many hops" answer... mysql domain table shows: mysql> describe domains; +--------+--------------+------+-----+---------+-------+ | Field | Type | Null | Key | Default | Extra | +--------+--------------+------+-----+---------+-------+ | code | int(11) | | PRI | 0 | | | domain | varchar(255) | | | | | +--------+--------------+------+-----+---------+-------+ 2 rows in set (0.00 sec) mysql> select * from domains; +------+---------------+ | code | domain | +------+---------------+ | 562 | xx.xx.xxx.234 | +------+---------------+ 1 row in set (0.01 sec) mysql> What am i doing wrong? Thanks Ricardo.- -----Mensaje original----- De: Bogdan-Andrei IANCU [mailto:iancu@fokus.fraunhofer.de] Enviado el: Viernes, 21 de Enero de 2005 19:19 Para: Ricardo Martinez CC: 'serusers(a)lists.iptel.org' Asunto: Re: [Serusers] Initialization PDT module problem. Richardo, I fixed on cvs both bugs, in pdt and mysql module. please update and re-try. bogdan Ricardo Martinez wrote: >Hello. > I now using the mysql.so module, but now i'm getting this error >(even with a core.dump). > > >

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[Serusers] mediaproxy problem
by Martin Bangiev 24 Jan '05

24 Jan '05

Hi guys, I'm trying to make X-lite to register to the ser server and the ser server to pass the invites to an asterisk box. The asterisk has only to play some sound sample in responce. And I want the RTP to pass through a mediaproxy. My X-lite is behind masquerading. I installed all the software needed for the tests and used the ser.cfg located in the mediaproxy.so module directory. But here is what appears in the ser's console: 0(25804) error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy 0(25804) error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy 0(25804) ERROR: on_reply processing failed And the conversation starts without using the mediaproxy. And when I hang up comes this: 0(25804) error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Where the problem could be (the mediaproxy is running)? And are there any examples about the domain.so module? I'm not sure if I configured it correctly becouse of insufficient documentation I found. I made this run with the rtpproxy, but I want to make it with mediaproxy too. Thanks in advance.

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[Serusers] Ser and Asterisk Integrated
by Felipe Martins 24 Jan '05

24 Jan '05

Hi everybody, I'm trying to integrate SER and Asterisk, put them to work together, ser ending SIP connections, and Asterisk acting as a REGISTRAR, logging the login attemps, passwords, and all the rest, in order to make a CDR and billing systems at asterisk. As i'm very new to VoIP and mainly to Asterisk and SER, i'm reading all the documentation I've found but I couldn't find any dox telling how to integrate them, with example config files and some graphs or something. Is there a link or Has anyone got dox on it ? It will be very aprecciated to have a link where I could learn it ... you know ... I'm already under pressure from the Forces ... Tnx a Lot in advance. -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

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[Serusers] SER and 2 domains register
by Welesley Sibelson Dias 24 Jan '05

24 Jan '05

I need register 2 domains in same SER using radius same part of script is but dont work, i have include into domain table both domains if (!radius_www_authorize( "192.168.0.216" /* realm */ /* table name */ )) { # challenge if none or invalid credentials www_challenge( "192.168.0.216" , "0") } else { if (!radius_www_authorize( "sipproxy.redevox.com.br")) /* table name */ )) { # challenge if none or invalid credentials www_challenge( "sipproxy.redevox.com.br" /* realm */, "0" ) break; } } thanks for you attention

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[Serusers] msilo can't dump stored message to different proxy?
by szj 24 Jan '05

24 Jan '05

Hi, all: I use two SER, one as registar, tho other as proxy. Both of them connected to the same database which located on the registar server. My one UA registered on the registar send message to another UA which registered on the same registar. when UA recieving message is offline and online later. It can't recieve the stored message through registar. When I trace the log of ser, I found that the UA registered to registar with To:<registar IP>. But Message's To header is in <proxy ip>, the msilo module don't think the message for the same UA with the rules of comparing the both To headers. How to cope with such situations? Thank you for your help and instructions. Best Regards. Sun Zongjun

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Re: [Serusers] Embedded OS
by Andrea Federico 24 Jan '05

24 Jan '05

I run ser-0.8.14 in linux (debian) with kernel 2.4.xx in an embedded system with an ARM processor. 1) is important you cross compile flex and bison for your architecture 2) you must use dbtext module instead of mysql module. bye. > > Hi All > > I am using SER from last one year and found it really interesting > and easy to use. I have one query now. Can we port SER on some > embedded system also. Can anyone guide me for that. > > Regards > > _________________________________________________________________ > Try the all-new MSN Search! Find exactly what you want. > http://search.msn.co.in Get more value for your time. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

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[Serusers] OPTIONS loop
by Ser Users 24 Jan '05

24 Jan '05

hello I have a problem with a sip phone witch is sending an OPTIONS message to the ser server itself. ser server is starting to send that message back to itself in a continuous loop adding a Record-Route and Via field every time. where is the problem ? do I have a broken client or a bad configuration for ser ? thanks Razvan Radu

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RE: [Serusers] Embedded OS
by Andrea Federico 24 Jan '05

24 Jan '05

> Is there any reason that one can't use mysql module? I thought that you only > need mysql client and then can have ser access a remote mysql server. > > Richard yes, it's correct! When I think about an embedded system, my idea is a small system with few hardware/software capabilities and in particular a stand alone system. Use Mysql server in an embedded system with for example 16/32Mbytes RAM, or a cpu with 66/100Mhz and a flash disk for me it's not a great idea. Best Regards Andrea Federico -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

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RE: [Serusers] b2bua + SER + RADIUS
by serdiehard 24 Jan '05

24 Jan '05

Dear Lucas, which radius server you are using ? iam using the free radius server - 0.9.3 and how are you maintaining the billing server my radius database radacct is haveing the following coloumns +-------------------------+----------------------+------+-+ | Field | Type | Null | Key | Default | Extra | +-------------------------+----------------------+------+-+ | RadAcctId | bigint(21) | | PRI | NULL | auto_increment | | AcctSessionId | varchar(255) | | MUL | | | | AcctUniqueId | varchar(255) | | MUL | | | | UserName | varchar(64) | | MUL | | | | Realm | varchar(64) | YES | | | | | NASIPAddress | varchar(15) | | MUL | | | | NASPortId | varchar(50) | | | | | | NASPortType | varchar(255) | | | | | | AcctStartTime | datetime | | MUL | 0000-00-00 00:00:00 | | | AcctStopTime | datetime | | MUL | 0000-00-00 00:00:00 | | | AcctSessionTime | int(12) | YES | | NULL | | | AcctAuthentic | varchar(32) | YES | | NULL | | | ConnectInfo_start | varchar(32) | YES | | NULL | | | ConnectInfo_stop | varchar(32) | YES | | NULL | | | AcctInputOctets | bigint(12) | YES | | NULL | | | AcctOutputOctets | bigint(12) | YES | | NULL | | | CalledStationId | varchar(50) | | | | | | CallingStationId | varchar(50) | | | | | | AcctTerminateCause | varchar(32) | | | | | | ServiceType | varchar(32) | YES | | NULL | | | FramedProtocol | varchar(32) | YES | | NULL | | | FramedIPAddress | varchar(15) | | MUL | | | | AcctStartDelay | int(12) | YES | | NULL | | | AcctStopDelay | int(12) | YES | | NULL | | | SipMethod | varchar(50) | | | | | | SipResponseCode | smallint(5) unsigned | | | 0 | | | SipToTag | varchar(255) | | | | | | SipFromTag | varchar(128) | | | | | | SipTranslatedRequestURI | varchar(255) | | | | | | SipUserAgents | varchar(128) | | | | | | SipApplicationType | varchar(255) | | | | | | SipCodecs | varchar(255) | | | | | | Timestamp | bigint(20) unsigned | | | 0 | | | DestinationId | varchar(255) | | | | | | Price | varchar(255) | | | | | | Rate | text | | | | | | Normalized | enum('0','1') did you able get sucess on the prepaid issue. if yes what approach you had to do this? with regards --- Lucas Aimaretto <lucas(a)cyneric.com> wrote: > > may i know which b2bua you are using? > > 1.5.0 (the one from www.vovida.org) > > > and how did you configured b2bua with the radius > and the ser exactly? > > this is configuration. > > B2BUA: 0.0.0.0:5065 > SER: 192.165.1.252:5060 > SER: 127.0.0.1:5060 > RADIUS: 0.0.0.0:1812 > RADIUS: 0.0.0.0:1813 > TEL SIP: 192.168.1.113:11001 > > - b2bConfig.xml > > <B2BUA_Configuration> > <SIP> > > <Local> > <Port>5065</Port> > <Transport>UDP</Transport> > </Local> > > // - Outbound Proxy Server > > <Proxy_Server> > <Address>192.168.1.252</Address> > <Port>5060</Port> > </Proxy_Server> > > <Registration> > <Register>no</Register> > <Address>192.168.1.252</Address> > <Port>5060</Port> > <Expires>600</Expires> > </Registration> > > </SIP> > > <RADIUS> > > <Local> > > <Authentication_Port>1812</Authentication_Port> > <Accounting_Port>1812</Accounting_Port> > </Local> > > <Billing_Server> > <Address>192.168.1.253</Address> > > <Authentication_Port>1813</Authentication_Port> > <Accounting_Port>1813</Accounting_Port> > <Password>rootpass</Password> > </Billing_Server> > > </RADIUS> > > <PrePaid> > > <Billing> > <Option>mandatory</Option> > <Refresh_Time>60</Refresh_Time> > > <Extract_User_Id_From>Proxy-Authorization</Extract_User_Id_From> > > <User_Id_Decode_Scheme>Basic</User_Id_Decode_Scheme> > </Billing> > > <Use_SIP_INFO>no</Use_SIP_INFO> > <Use_HTTP>no</Use_HTTP> > > </PrePaid> > > <Redundancy /> > > </B2BUA_Configuration> > > - ser.cfg > > [...] > # -- RADIUS AUTH -- > modparam("auth_radius", "radius_config", > "/etc/radiusclient.conf") > modparam("auth_radius", "service_type", 15) > # -- RADIUS ACC -- > modparam("acc", "radius_config", > "/etc/radiusclient.conf") > modparam("acc", "radius_flag", 1) > modparam("acc", "radius_missed_flag", 3) > modparam("acc", "log_level", 2) > [...] > route{ > setflag(1); > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long > requests > if (!mf_process_maxfwd_header("10")) > { > sl_send_reply("483","Too Many > Hops"); > break; > }; > if ( msg:len > max_len ) > { > sl_send_reply("513", "Message too > big"); > break; > }; > > # we record-route all messages -- to make > sure that > # subsequent messages will go through our > proxy; that's > # particularly good if upstream and > downstream entities > # use different transport protocol > record_route(); > # loose-route processing > > if (loose_route()) > { > t_relay(); > break; > }; > # if the request is for other domain use > UsrLoc > # (in case, it does not work, use the > following command > # with proper names and addresses in it) > > if (uri==myself) > { > > if (method=="REGISTER") > { > log(1, "REGISTER message > received\n"); > # Lo siguiente es para auth > con RADIUS > if > (!radius_www_authorize("")) > { > www_challenge("", > "0"); > break; > }; > save("location"); > break; > }; > > if (method=="INVITE") > { > forward(192.168.1.253,5065); > > sl_send_reply("181","Forwarding the call to > B2BUA"); > > }; > > # native SIP destinations are > handled using our USRLOC > DB > if (!lookup("location")) > { > sl_send_reply("404", "Not > Found"); > break; > }; > }; > > t_relay(); > } > > > could you able to establish the communication > properly? > > REGISTER: > > No problems here, because the telephone talks to > SER, SER talks to > RADIUS (see ser.cfg) and the phone is registered. > Easy. > When TEL sends and INVITE, SER forwards the INVITE > to B2BUA, which sends > an Authorization Request to RADIUS. OK here, because > B2BUA sends and > auth req with calling-station-id and > called-station-id, which is good, > with both acct-start and acct-stop. Now, upon the > BYE message, B2BUA is > not sending the Acct-Session-Time for doing billing. > Which is not good. > Any ideas ? > > Now, it this the correct way of doing this ? > > Thanx. > === message truncated === __________________________________ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com

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Re: [Serusers] FW: SER and NAT
by Ashling O'Driscoll 22 Jan '05

22 Jan '05

Hi all, I would like to thank everyone for their help to date answering my questions (particularly Giovanni). The clients now register with SER despite the NAT issues. I have yet to test a call so I may be back with more queries if the voice doesnt transmit!! I used the default NAT script provided on the voip inof wiki. However I want my clients to authenticate before they are allowed to register. If I enable authentication in the ser.cfg script, registration fails. I do not understand why as the mysql database is set up and I have created user accounts. I have included my ser.cfg script below with the authentication part and the mysql load module part commented out. Users only authenticate at the moment if these parts are commented. Any help would be greatly appreciated. Also on a slightly unrelated note, I am trying to test this system with Grandstream Budgetone 100 hardphones (Its so far tested with XLite)I plug these hones into a hub which is in turn plugged into a router,however the phone wont obtain an IP address through dhcp or take its statically assigned one. Shouldn't an ip phone act like any other IP device on a network i.e. a pc etc?? Thanks as always, Aisling. # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" alias=84.203.148.14 # ------------------ 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/nathelper.so" #loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/maxfwd.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) #!!Nathelper #modparam("registrar","nat_flag",6) #modparam("nathelper","natping_interval",30) #Ping intervals 30 seconds #modparam("nathelper","ping_nated_only",1) #Ping only clinets behind NAT # -------------------------request routing logic------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; #############Aisling Insert################ # #!Nat Insert # #the below line tests if the IP of the received packet is different from the IP in the via header and also # #sees if the IP address in the contact header is private # if (nat_uac_test("3")){ # if (method == "REGISTER" || ! search("^Record-Route:")){ # log("Log: Someone trying to register from private IP,rewriting\n"); # # fixed_nated_contact(); #Rewrite contact with source IP # if (method == "INVITE"){ # fix_nated_sdp("1"); #Add direction=active to SDP # }; # force_rport(); # Add rport parameter to topmost Via # setflag(6); # Mark as Nated # }; # }; ###################End##################### # 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(); # 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("84.203.148.14", "subscriber")) { # www_challenge("84.203.148.14", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #inserted by klaus if (method == "INVITE"){ record_route(); force_rtp_proxy(); /* set up reply processing*/ t_on_reply("1"); }; # 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(); }; } #insert by klaus onreply_route[1]{ if (status=~"[12][0-9][0-9]") force_rtp_proxy(); } #route[1] #{ #if client or server know to be behind NAT, enable relay # if (isflagset(6)){ # force_rtp_proxy(); # }; # # #NAT processing of replies; apply to all transaction (for example, # #reinvites from public to private UA are hard to identify as # #Nated at the moment of request processing); look at replies # t_on_reply("1"); # # #send it out now; use stateful forwarding as it works reliably # #even for UDP2TCP # if(!t_relay()){ # sl_reply_error(); # }; #} #!!NatHelper #onreply_route[1]{ #Nated Transaction?? #if (isflagset(6) && status =~ "(183)\2[0-9][0-9]"){ # #fixed_nated_contact(); # force_rtp_proxy(); # } #else if (nat_uac_test("1")){ # fix_nated_contact(); # }; #} ---- Original Message ---- From: jev(a)emmplus.ie To: ashling.odriscoll(a)cit.ie Subject: Re: [Serusers] FW: SER and NAT Date: Fri, 21 Jan 2005 13:47:47 -0800 >Hi Ashling, > >Running ser behind NAT is a big pain in the ass, and I would suggest >you >do what ever you can to avoid that situation. I have played a little >with such a set up, and it's just a crap situation. > >Best thing to do is get your ser on a public IP, or at least a IP >that >is routable through out the entire CIT campus. > >What type of router have you in front of ser? If it's some linux host > >doing routing, maybe you could put ser right on it. If it's a cisco >or >something, then maybe not ;) > >Let me know how it goes.... > >Good luck, >-Jev > >Ashling O'Driscoll wrote: >> Just to give more information on the below problem...This is what I >> saw in the xlite diagnostic log...Does the warning at the timeoue >> message mean something to anyone?? >> >> Thanks again, >> Aisling. >> >> >> SEND TIME: 325416584 >> SEND >> 84.203.148.14:5060 >> REGISTER sip:84.203.148.14 SIP/2.0 >> Via: SIP/2.0/UDP >> >157.190.70.231:5061;rport;branch=z9hG4bK7BFAD25591E34485B5C67454E24B9 >B >> EB >> From: Aisling O' Driscoll <sip:2000@84.203.148.14>;tag=1976120825 >> To: Aisling O' Driscoll <sip:2000@84.203.148.14> >> Contact: "Aisling O' Driscoll" <sip:2000@157.190.70.231:5061> >> Call-ID: 2B12EE2F1DA64C11984134EF0CE89DC1(a)84.203.148.14 >> CSeq: 5729 REGISTER >> Expires: 1800 >> Max-Forwards: 70 >> User-Agent: X-Lite release 1103m >> Content-Length: 0 >> >> >> RECEIVE TIME: 325434169 >> RECEIVE << 84.203.148.14:5060 >> SIP/2.0 408 Request Timeout >> Via: SIP/2.0/UDP >> >157.190.70.231:5061;rport=5061;branch=z9hG4bKFDD40D52818B4721AF5F83D3 >0 >> A7D526F >> From: Aisling O' Driscoll <sip:2000@84.203.148.14>;tag=3316651270 >> To: Aisling O' Driscoll >> <sip:2000@84.203.148.14>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-a63b >> Call-ID: E8E25C47488A4E958F4BE8293AF7E1F3(a)84.203.148.14 >> CSeq: 16015 REGISTER >> Server: Sip EXpress router (0.8.14 (i386/linux)) >> Content-Length: 0 >> Warning: 392 172.16.3.15:5060 "Noisy feedback tells: pid=19819 >> req_src_ip=157.190.70.231 req_src_port=5061 >in_uri=sip:84.203.148.14 >> out_uri=sip:84.203.148.14 via_cnt==0" >> >> ---- Original Message ---- >> From: ashling.odriscoll(a)cit.ie >> To: serusers(a)iptel.org >> Subject: FW: SER and NAT >> Date: Fri, 21 Jan 2005 09:49:56 -0000 >> >> Hi, >> >> Hope someone can help me. I have a fairly bad NAT situation. >> >> Clients are behind NAT and SER is behind NAT. The idea is that the >> clients will register to the public address which is assigned to my >> router and I have enable port-forwarding to send the packets onto >the >> pc running SER listening on 5060. This situation worked when >clients >> registered with Asterisk but I would rather them to register with >SER >> and just use Asterisk for voicemail etc. >> >> My config is like the default except the nathelper and mysql >modules >> are loaded. I have looked at the following websites but all the >> configs are very different and use different rtp proxys..I am also >> confused as to how much configuration is required to accommodate >nat >> >> >http://lists.cs.columbia.edu/pipermail/sip-implementors/2004-February >/ >> 006179.html >> http://voip-info.org/wiki-SER+example+NAThelper >> >> Could someone give me an idea of how the script should be modified, >> what else must be installed (rtp proxy wise)or any ideas as to why >my >> clients cant register?....Im presuming its nat... >> >> Thanks a million, >> 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)iptel.org >> http://mail.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.

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[Serusers] Seeking Advise With Caller ID Block
by Java Rockx 22 Jan '05

22 Jan '05

Hi All. I'd like to implement caller ID blocking at the SIP proxy. Can I simply use textops replace() to rewrite the "From:" header to something like the following without breaking the SIP transaction? From: "Anonymous" <sip:my.domain.com> Using this in my ser.cfg: replace("^From:(.*)sip:(.*)@" , "From: \"Anonymous\" <sip:"); Regards, Paul __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

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RE: [Serusers] STOP NOTIFY
by Matt Schulte 22 Jan '05

22 Jan '05

Well, if you don't think you need NOTIFY's you could do: if (method == NOTIFY) { sl_send_reply("200", "blah blah") break; } You'd want to put that before your accounting setflag. -----Original Message----- From: Ronald Ramos [mailto:ron@silverbackasp.com] Sent: Friday, January 21, 2005 6:13 PM To: serusers(a)lists.iptel.org Subject: [Serusers] STOP NOTIFY Hi All, How can I configure ser to stop sending NOTIFY on the table acc? It's flooding my database, it inserts row like 800+ every minute. Hope anyone can help me. Thank you Regards, Ron _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] STOP NOTIFY
by Ronald Ramos 22 Jan '05

22 Jan '05

Hi All, How can I configure ser to stop sending NOTIFY on the table acc? It's flooding my database, it inserts row like 800+ every minute. Hope anyone can help me. Thank you Regards, Ron

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RE: [Serusers] Initialization PDT module problem.
by Ricardo Martinez 21 Jan '05

21 Jan '05

Hello. I now using the mysql.so module, but now i'm getting this error (even with a core.dump). 0(0) DEBUG: init_mod: pdt 0(0) PDT: initializing... 0(0) qm_malloc(0x4212f000, 4) called from pdt.c: mod_init(228) 0(0) qm_malloc(0x4212f000, 4) returns address 0x422e6098 frag. 0x422e6080 (size=4) on 1 -th hit 0(0) qm_malloc(0x80fc3c0, 16) called from fifo_server.c: register_fifo_cmd(133) 0(0) qm_malloc(0x80fc3c0, 16) returns address 0x81333fc frag. 0x81333e4 (size=16) on 1 -th hit 0(0) DEBUG: register_fifo_cmd: new command (get_domainprefix) registered 0(0) qm_malloc(0x80fc3c0, 16) called from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_malloc(0x80fc3c0, 16) returns address 0x813343c frag. 0x8133424 (size=16) on 1 -th hit 0(0) unixsock_register_cmd: New command (get_domainprefix) registered 0(0) qm_malloc(0x80fc3c0, 6) called from db/db.c: bind_dbmod(69) 0(0) qm_malloc(0x80fc3c0, 8) returns address 0x813347c frag. 0x8133464 (size=8) on 1 -th hit 0(0) find_mod_export: found <db_use_table> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_init> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_close> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_raw_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_free_result> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) qm_malloc(0x80fc3c0, 12) called from dbase.c: db_init(253) 0(0) qm_malloc(0x80fc3c0, 12) returns address 0x81334b4 frag. 0x813349c (size=12) on 1 -th hit 0(0) qm_malloc(0x80fc3c0, 27) called from my_id.c: new_my_id(53) 0(0) qm_malloc(0x80fc3c0, 28) returns address 0x81334f0 frag. 0x81334d8 (size=28) on 1 -th hit 0(0) qm_malloc(0x80fc3c0, 36) called from my_id.c: new_my_id(60) 0(0) qm_malloc(0x80fc3c0, 36) returns address 0x813353c frag. 0x8133524 (size=36) on 1 -th hit 0(0) qm_malloc(0x80fc3c0, 5) called from my_id.c: new_my_id(73) 0(0) qm_malloc(0x80fc3c0, 8) returns address 0x8133590 frag. 0x8133578 (size=8) on 1 -th hit Segmentation fault (core dumped) This is my ser.cfg # # $Id: pdt.cfg,v 1.3 2003/11/03 13:03:58 janakj Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes# (cmd line: -E) 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" listen=xxx.xxx.xxx.232 # ------------------ 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" loadmodule "/usr/local/lib/ser/modules/pdt.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.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", "secret", "alsdkhglaksdhfkloiwr") #modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth", "password_column", "password") modparam("pdt", "db_url", "mysql://root@127.0.0.1/pdt") modparam("pdt", "db_table", "domains") modparam("pdt", "prefix", "00") #modparam("pdt", "start_range", 2000) #modparam("pdt", "hsize_2pow", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; loose_route(); prefix2domain(); record_route(); # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } Can someone help me? Thanks in advance. Ricardo.- -----Mensaje original----- De: Ricardo Martinez [mailto:rmartinez@redvoiss.net] Enviado el: Viernes, 21 de Enero de 2005 17:14 Para: 'serusers(a)lists.iptel.org' Asunto: [Serusers] Initialization PDT module problem. Hello. I was trying to start SER with the PDT module and i got this error message. Can somone help me ? 0(0) PDT: initializing... 0(0) qm_malloc(0x4212f000, 4) called from pdt.c: mod_init(228) 0(0) qm_malloc(0x4212f000, 4) returns address 0x422e6098 frag. 0x422e6080 (size=4) on 1 -th hit 0(0) qm_malloc(0x80fc3c0, 16) called from fifo_server.c: register_fifo_cmd(133) 0(0) qm_malloc(0x80fc3c0, 16) returns address 0x81337cc frag. 0x81337b4 (size=16) on 1 -th hit 0(0) DEBUG: register_fifo_cmd: new command (get_domainprefix) registered 0(0) qm_malloc(0x80fc3c0, 16) called from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_malloc(0x80fc3c0, 16) returns address 0x813380c frag. 0x81337f4 (size=16) on 1 -th hit 0(0) unixsock_register_cmd: New command (get_domainprefix) registered 0(0) qm_malloc(0x80fc3c0, 6) called from db/db.c: bind_dbmod(69) 0(0) qm_malloc(0x80fc3c0, 8) returns address 0x813384c frag. 0x8133834 (size=8) on 1 -th hit 0(0) find_mod_export: <db_use_table> in module mysql not found 0(0) bind_dbmod: Module mysql does not export db_use_table function 0(0) qm_free(0x80fc3c0, 0x813384c), called from db/db.c: bind_dbmod(142) 0(0) qm_free: freeing frag. 0x8133834 alloc'ed from db/db.c: bind_dbmod(69) 0(0) PDT: mod_init: Database module not found 0(0) qm_free(0x4212f000, 0x422e6098), called from pdt.c: mod_init(356) 0(0) qm_free: freeing frag. 0x422e6080 alloc'ed from pdt.c: mod_init(228) 0(0) init_mod(): Error while initializing module pdt ERROR: error while initializing modules 0(0) qm_free(0x80fc3c0, 0x813131c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8131304 alloc'ed from sr_module.c: register_module(136) 0(0) PDT: mod_destroy : Cleaning up 0(0) qm_free(0x4212f000, 0x422e6098), called from pdt.c: mod_destroy(545) 0(0) BUG: qm_free: freeing already freed pointer, first free: pdt.c: mod_init(356) - aborting According to this the PDT module can't find the database. But this is the "show databases" from mysql:mysql> show databases; +------------+ | Database | +------------+ | mysql | | pdt | | test | +------------+ 5 rows in set (0.00 sec) mysql> Anyone? Thanks ! Ricardo.- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] rewriteuser
by Mik Cheez 21 Jan '05

21 Jan '05

I would like to be able to route calls based on the user part of the URI, but with a wrinkle; Assuming the name was a 10 digit number (ie. 5551230000), I'd like to be able to use only the first six digits to route a call. Is there currently a way to strip off the last four digits just until I find the user group they belong to? Best regards, Mik

1 0

[Serusers] Initialization PDT module problem.
by Ricardo Martinez 21 Jan '05

21 Jan '05

Hello. I was trying to start SER with the PDT module and i got this error message. Can somone help me ? 0(0) PDT: initializing... 0(0) qm_malloc(0x4212f000, 4) called from pdt.c: mod_init(228) 0(0) qm_malloc(0x4212f000, 4) returns address 0x422e6098 frag. 0x422e6080 (size=4) on 1 -th hit 0(0) qm_malloc(0x80fc3c0, 16) called from fifo_server.c: register_fifo_cmd(133) 0(0) qm_malloc(0x80fc3c0, 16) returns address 0x81337cc frag. 0x81337b4 (size=16) on 1 -th hit 0(0) DEBUG: register_fifo_cmd: new command (get_domainprefix) registered 0(0) qm_malloc(0x80fc3c0, 16) called from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_malloc(0x80fc3c0, 16) returns address 0x813380c frag. 0x81337f4 (size=16) on 1 -th hit 0(0) unixsock_register_cmd: New command (get_domainprefix) registered 0(0) qm_malloc(0x80fc3c0, 6) called from db/db.c: bind_dbmod(69) 0(0) qm_malloc(0x80fc3c0, 8) returns address 0x813384c frag. 0x8133834 (size=8) on 1 -th hit 0(0) find_mod_export: <db_use_table> in module mysql not found 0(0) bind_dbmod: Module mysql does not export db_use_table function 0(0) qm_free(0x80fc3c0, 0x813384c), called from db/db.c: bind_dbmod(142) 0(0) qm_free: freeing frag. 0x8133834 alloc'ed from db/db.c: bind_dbmod(69) 0(0) PDT: mod_init: Database module not found 0(0) qm_free(0x4212f000, 0x422e6098), called from pdt.c: mod_init(356) 0(0) qm_free: freeing frag. 0x422e6080 alloc'ed from pdt.c: mod_init(228) 0(0) init_mod(): Error while initializing module pdt ERROR: error while initializing modules 0(0) qm_free(0x80fc3c0, 0x813131c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8131304 alloc'ed from sr_module.c: register_module(136) 0(0) PDT: mod_destroy : Cleaning up 0(0) qm_free(0x4212f000, 0x422e6098), called from pdt.c: mod_destroy(545) 0(0) BUG: qm_free: freeing already freed pointer, first free: pdt.c: mod_init(356) - aborting According to this the PDT module can't find the database. But this is the "show databases" from mysql:mysql> show databases; +------------+ | Database | +------------+ | mysql | | pdt | | test | +------------+ 5 rows in set (0.00 sec) mysql> Anyone? Thanks ! Ricardo.-

1 0

[Serusers] missing bye on acc
by sendman 21 Jan '05

21 Jan '05

Hi folks! I have a problem, on my sip proxy, some UA (i think the problem - linksys) don't send bye to proxy server, when the call was terminated! Regards.

1 0

[Serusers] ser-0.9 Bug with BYE messages/Record-Route not being honored??
by Java Rockx 21 Jan '05

21 Jan '05

Hi All. I think I've identified a bug with how ser-0.9 handles BYE messages. It seems that ser-0.9 is not honoring Record-Route headers when sending BYE messages. Below is a complete SIP conversation between my GrandStream BT100 phone and a PSTN phone. We use a 3rd party for PSTN access and they utilize a Sonus box for SIP-to-TDM and a seperate SIP proxy. The actual problem is this; when a SIP phone hangs up **FIRST** the BYE message is sent to their Sonus Box directly rather than to their SIP proxy. Their Sonus box is listed as the "Contact" field in the SDP of the INVITE. The original INVITE was Record-Routed so shouldn't the BYE be sent to this IP rather than to the other end-point directly? When the phones go off-hook, our PSTN provider sends us a 200 OK which we ACK correctly, but then the BYE is wrong. Can anyone look at this conversation and tell me why the BYE is sent to 216.229.118.76 rather than 216.229.127.60? Regards, Paul IP LEGEND --------------------------------------------------- 68.84.100.10 - PUBLIC SIDE OF MY LAN FIREWALL 68.84.199.18 - IP ADDRESS OF MY SER SIP PROXY 216.229.127.60 - IP ADDRESS OF 3RD PARTY SIP PROXY 216.229.118.76 - IP ADDRESS OF 3RD PARTY SONUS BOX U 2005/01/21 11:38:14.259621 68.84.100.10:63689 -> 68.84.199.18:5060 INVITE sip:3219392336@mycompany.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK67f005c3c771821e. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>. Contact: <sip:3212514231@192.168.0.83:63689;user=phone>. Supported: replaces. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27735 INVITE. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 411. . v=0. o=3212514231 0 8000 IN IP4 192.168.0.83. s=SIP Call. c=IN IP4 192.168.0.83. t=0 0. m=audio 39738 RTP/AVP 98 18 4 15 2 9 0 8 101. a=sendrecv. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:15 G728/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=ptime:40. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. # U 2005/01/21 11:38:14.264900 68.84.199.18:5060 -> 68.84.100.10:63689 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK67f005c3c771821e;rport=63689;received=68.84.100.10. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=bf952ed189d8425c881b09485aa0b6f1.954a. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27735 INVITE. Proxy-Authenticate: Digest realm="mycompany.com", nonce="41f13122784651d60bfa0cd3d5778587c1105f35". Server: Sip eXpress Router (0.9.0 (i386/linux)). Content-Length: 0. . # U 2005/01/21 11:38:14.270933 68.84.100.10:63689 -> 68.84.199.18:5060 ACK sip:3219392336@mycompany.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK67f005c3c771821e. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=bf952ed189d8425c881b09485aa0b6f1.954a. Contact: <sip:3212514231@192.168.0.83:63689;user=phone>. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27735 ACK. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 11:38:14.288516 68.84.100.10:63689 -> 68.84.199.18:5060 INVITE sip:3219392336@mycompany.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK7779e54c00e334a6. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>. Contact: <sip:3212514231@192.168.0.83:63689;user=phone>. Supported: replaces. Proxy-Authorization: DIGEST username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:3219392336@mycompany.com;user=phone", nonce="41f13122784651d60bfa0cd3d5778587c1105f35", response="2463041979c1bb36367b0d9d06bfa966". Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 411. . v=0. o=3212514231 1 8000 IN IP4 192.168.0.83. s=SIP Call. c=IN IP4 192.168.0.83. t=0 0. m=audio 39738 RTP/AVP 98 18 4 15 2 9 0 8 101. a=sendrecv. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:15 G728/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=ptime:40. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. # U 2005/01/21 11:38:14.304423 68.84.199.18:5060 -> 68.84.100.10:63689 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK7779e54c00e334a6;rport=63689;received=68.84.100.10. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. Server: Sip eXpress Router (0.9.0 (i386/linux)). Content-Length: 0. . # U 2005/01/21 11:38:14.304581 68.84.199.18:5060 -> 216.229.127.60:5060 INVITE sip:3219392336@216.229.127.60;user=phone SIP/2.0. Record-Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKd55a.8a20c236.0. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>. Contact: <sip:3212514231@68.84.100.10:63689;user=phone>. Supported: replaces. Proxy-Authorization: DIGEST username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:3219392336@mycompany.com;user=phone", nonce="41f13122784651d60bfa0cd3d5778587c1105f35", response="2463041979c1bb36367b0d9d06bfa966". Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 412. . v=0. o=3212514231 1 8000 IN IP4 192.168.0.83. s=SIP Call. c=IN IP4 68.84.199.18. t=0 0. m=audio 36096 RTP/AVP 98 18 4 15 2 9 0 8 101. a=sendrecv. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:15 G728/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=ptime:40. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. # U 2005/01/21 11:38:14.370297 216.229.127.60:5060 -> 68.84.199.18:5060 SIP/2.0 100 Try. Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKd55a.8a20c236.0. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6. To: <sip:3219392336@mycompany.com;user=phone>. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. Supported: replaces. Proxy-Authorization: Digest username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:3219392336@mycompany.com;user=phone", nonce="41f13122784651d60bfa0cd3d5778587c1105f35", response="2463041979c1bb36367b0d9d06bfa966". User-Agent: Grandstream BT100 1.0.5.16 . Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Content-Length: 0. . # U 2005/01/21 11:38:15.561196 216.229.127.60:5060 -> 68.84.199.18:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKd55a.8a20c236.0. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. Contact: sip:3219392336@216.229.118.76:4060. Record-Route: <sip:216.229.127.60:5060;lr>. Record-Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 249. . v=0. o=Sonus_UAC 31122 22717 IN IP4 216.229.118.76. s=SIP Media Capabilities. c=IN IP4 216.229.118.100. t=0 0. m=audio 14202 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. # U 2005/01/21 11:38:15.565333 68.84.199.18:5060 -> 68.84.100.10:63689 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. Contact: sip:3219392336@216.229.118.76:4060. Record-Route: <sip:216.229.127.60:5060;lr>. Record-Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 247. . v=0. o=Sonus_UAC 31122 22717 IN IP4 216.229.118.76. s=SIP Media Capabilities. c=IN IP4 68.84.199.18. t=0 0. m=audio 36096 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. ## U 2005/01/21 11:38:20.788205 216.229.127.60:5060 -> 68.84.199.18:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKd55a.8a20c236.0. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. Contact: sip:3219392336@216.229.118.76:4060. Record-Route: <sip:216.229.127.60:5060;lr>. Record-Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay. Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO. Supported: timer. Content-Disposition: session;handling=required. Content-Type: application/sdp. Session-Expires: 240;refresher=uas. Content-Length: 249. . v=0. o=Sonus_UAC 31122 22717 IN IP4 216.229.118.76. s=SIP Media Capabilities. c=IN IP4 216.229.118.100. t=0 0. m=audio 14202 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. # U 2005/01/21 11:38:20.791744 68.84.199.18:5060 -> 68.84.100.10:63689 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 INVITE. Contact: sip:3219392336@216.229.118.76:4060. Record-Route: <sip:216.229.127.60:5060;lr>. Record-Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay. Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO. Supported: timer. Content-Disposition: session;handling=required. Content-Type: application/sdp. Session-Expires: 240;refresher=uas. Content-Length: 247. . v=0. o=Sonus_UAC 31122 22717 IN IP4 216.229.118.76. s=SIP Media Capabilities. c=IN IP4 68.84.199.18. t=0 0. m=audio 36096 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. # U 2005/01/21 11:38:20.815426 68.84.100.10:63689 -> 68.84.199.18:5060 ACK sip:3219392336@216.229.118.76:4060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bKe4c4477775398df9. Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Route: <sip:216.229.127.60:5060;lr>. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. Contact: <sip:3212514231@192.168.0.83:63689;user=phone>. Proxy-Authorization: DIGEST username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:3219392336@216.229.118.76:4060", nonce="41f13122784651d60bfa0cd3d5778587c1105f35", response="4160207afa0e806b3e1050a73ca1a852". Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 ACK. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 11:38:20.816100 68.84.199.18:5060 -> 216.229.127.60:5060 ACK sip:3219392336@216.229.118.76:4060 SIP/2.0. Record-Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Via: SIP/2.0/UDP 68.84.199.18;branch=0. Via: SIP/2.0/UDP 192.168.0.83:63689;received=68.84.100.10;branch=z9hG4bKe4c4477775398df9. Route: <sip:216.229.127.60:5060;lr>. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. Contact: <sip:3212514231@192.168.0.83:63689;user=phone>. Proxy-Authorization: DIGEST username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:3219392336@216.229.118.76:4060", nonce="41f13122784651d60bfa0cd3d5778587c1105f35", response="4160207afa0e806b3e1050a73ca1a852". Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27736 ACK. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . ##################### U 2005/01/21 11:38:31.289799 68.84.100.10:63689 -> 68.84.199.18:5060 BYE sip:3219392336@216.229.118.76:4060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK41337bf509b31050. Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Route: <sip:216.229.127.60:5060;lr>. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. Proxy-Authorization: DIGEST username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:3219392336@216.229.118.76:4060", nonce="41f13122784651d60bfa0cd3d5778587c1105f35", response="85f6afb2964895599d68eac1f6812fb5". Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27737 BYE. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 11:38:31.293894 68.84.199.18:5060 -> 216.229.118.76:4060 BYE sip:3219392336@216.229.118.76:4060 SIP/2.0. Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKe55a.7940d157.0. Via: SIP/2.0/UDP 192.168.0.83:63689;received=68.84.100.10;branch=z9hG4bK41337bf509b31050. Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>. Route: <sip:216.229.127.60:5060;lr>. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. Proxy-Authorization: DIGEST username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:3219392336@216.229.118.76:4060", nonce="41f13122784651d60bfa0cd3d5778587c1105f35", response="85f6afb2964895599d68eac1f6812fb5". Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27737 BYE. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 11:38:31.364179 216.229.118.76:4060 -> 68.84.199.18:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKe55a.7940d157.0. Via: SIP/2.0/UDP 192.168.0.83:63689;received=68.84.100.10;branch=z9hG4bK41337bf509b31050. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27737 BYE. Content-Length: 0. . # U 2005/01/21 11:38:31.364640 68.84.199.18:5060 -> 68.84.100.10:63689 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.0.83:63689;received=68.84.100.10;branch=z9hG4bK41337bf509b31050. From: "Paul Hazlett" <sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c. To: <sip:3219392336@mycompany.com;user=phone>;tag=0ceeba5a. Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83. CSeq: 27737 BYE. Content-Length: 0. . __________________________________ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250

1 1

[Serusers] v0.9 change log
by Steve Blair 21 Jan '05

21 Jan '05

I upgraded to 0.9 and my current configuration file no longer loads. Before I poke around and ask alot of the same questions others have asked I was wondering if there is a change log for this release? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

2 1

[Serusers] FW: radius accounting problem.
by CM Rahman Jr. 21 Jan '05

21 Jan '05

Anybody here solved the radius accounting problem with ser? Please post if you are using radius with ser for accounting. Thanks &*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. IT Manager CCNP, MCSE Security "Secure your self by securing your System" CompTI Security Plus Certified CCS Internet <http://www.ccsi.com/> http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 _____

3 2

[Serusers] newbie - Handle incoming calls?
by Tim Pushor 21 Jan '05

21 Jan '05

Hi All, I'm new to all this stuff, so apologize if these are dumb questions. I have an account with a local/ld voip provider. They provide me a SIP connection to/from the pstn. I have this working with sjphone and xten. I also have ser running (very basically) and handling/routing calls for local softphones registering with it. I really want to be able to receive incoming pstn calls and route them internally. Here's where my ignorance comes in. The sjphone and xten register at the vendors registrar as sip:nnnn@vendor.com. I havn't seen an example or read anything how I could configure ser to do that. Also, the SIP line is authenticated. How do I tell ser to use a set of credentials when sending/receiving SIP messages to my vendor? Thanks! Tim

1 0

[Serusers] HELP: 487 Request Terminated Being Retransmitted
by Java Rockx 21 Jan '05

21 Jan '05

Hi all. Below is the tail end of a SIP conversation whereby I dialed a PSTN telephone number and let it ring but hung up before the PSTN phone was answered. The SER sent the PSTN gateway a CANCEL message and it replied and all seemed well until the very end of the conversation where there are multiple "487 Request Terminated" messages transmitted from SER to the IP phone. Can anyone tell me if this is "normal" or if I need to worry about this? Regards, Paul IP Legend: ------------------------------------- 68.84.10.10 - PUBLIC SIDE OF FIREWALL 68.84.10.20 - IP ADDRESS OF SER PROXY 68.84.10.30 - IP ADDRESS OF PSTN GATEWAY U 2005/01/21 10:43:36.874272 68.84.10.10:63689 -> 68.84.10.20:5060 CANCEL sip:4075551212@mycompany.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK868145b476db8dae. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. To: <sip:4075551212@mycompany.com;user=phone>. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 CANCEL. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 10:43:36.877795 68.84.10.20:5060 -> 68.84.10.30:5060 CANCEL sip:4075551212@68.84.10.30;user=phone SIP/2.0. Via: SIP/2.0/UDP 68.84.10.20;branch=z9hG4bKd753.8322b441.0. Via: SIP/2.0/UDP 192.168.0.83:63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. To: <sip:4075551212@mycompany.com;user=phone>. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 CANCEL. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 10:43:36.877884 68.84.10.20:5060 -> 68.84.10.10:63689 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK868145b476db8dae;received=68.84.10.10. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. To: <sip:4075551212@mycompany.com;user=phone>;tag=8dfa8202461711560b5fb8c33916f682-1b42. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 CANCEL. Content-Length: 0. . # U 2005/01/21 10:43:36.934535 68.84.10.30:5060 -> 68.84.10.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.84.10.20;branch=z9hG4bKd753.8322b441.0. Via: SIP/2.0/UDP 192.168.0.83:63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. To: <sip:4075551212@mycompany.com;user=phone>. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 CANCEL. User-Agent: Grandstream BT100 1.0.5.16 . Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 10:43:36.955761 68.84.10.30:5060 -> 68.84.10.20:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 68.84.10.20;branch=z9hG4bKd753.8322b441.0. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 INVITE. Record-Route: <sip:68.84.10.30:5060;lr>. Record-Route: <sip:68.84.10.20;ftag=16ec22902b85a039;lr=on>. Content-Length: 0. . # U 2005/01/21 10:43:36.955950 68.84.10.20:5060 -> 68.84.10.30:5060 ACK sip:4075551212@68.84.10.30;user=phone SIP/2.0. Via: SIP/2.0/UDP 68.84.10.20;branch=z9hG4bKd753.8322b441.0. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. CSeq: 30333 ACK. Content-Length: 0. . # U 2005/01/21 10:43:36.956037 68.84.10.20:5060 -> 68.84.10.10:63689 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 INVITE. Record-Route: <sip:68.84.10.30:5060;lr>. Record-Route: <sip:68.84.10.20;ftag=16ec22902b85a039;lr=on>. Content-Length: 0. . # U 2005/01/21 10:43:36.976267 68.84.10.10:63689 -> 68.84.10.20:5060 ACK sip:4075551212@mycompany.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK868145b476db8dae. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. Contact: <sip:3212514231@192.168.0.83:63689;user=phone>. Proxy-Authorization: DIGEST username="3212514231", realm="mycompany.com", algorithm=MD5, uri="sip:4075551212@mycompany.com;user=phone", nonce="41f12450f15d3b4a9fb6cd05506a6030c7165718", response="d9a2404031e7404ad89df3bef3d5dfd4". Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 ACK. User-Agent: Grandstream BT100 1.0.5.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 2005/01/21 10:43:37.152117 68.84.10.20:5060 -> 68.84.10.10:63689 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 INVITE. Record-Route: <sip:68.84.10.30:5060;lr>. Record-Route: <sip:68.84.10.20;ftag=16ec22902b85a039;lr=on>. Content-Length: 0. . ## U 2005/01/21 10:43:39.172156 68.84.10.20:5060 -> 68.84.10.10:63689 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 INVITE. Record-Route: <sip:68.84.10.30:5060;lr>. Record-Route: <sip:68.84.10.20;ftag=16ec22902b85a039;lr=on>. Content-Length: 0. . # U 2005/01/21 10:43:43.212328 68.84.10.20:5060 -> 68.84.10.10:63689 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 INVITE. Record-Route: <sip:68.84.10.30:5060;lr>. Record-Route: <sip:68.84.10.20;ftag=16ec22902b85a039;lr=on>. Content-Length: 0. . ########### U 2005/01/21 10:43:47.252508 68.84.10.20:5060 -> 68.84.10.10:63689 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 192.168.0.83:63689;rport=63689;received=68.84.10.10;branch=z9hG4bK868145b476db8dae. To: <sip:4075551212@mycompany.com;user=phone>;tag=06b020d6. From: "Paul" <sip:3212514231@mycompany.com;user=phone>;tag=16ec22902b85a039. Call-ID: 42c962bf359504ab(a)192.168.0.83. CSeq: 30333 INVITE. Record-Route: <sip:68.84.10.30:5060;lr>. Record-Route: <sip:68.84.10.20;ftag=16ec22902b85a039;lr=on>. Content-Length: 0. __________________________________ Do you Yahoo!? All your favorites on one personal page � Try My Yahoo! http://my.yahoo.com

1 0

[Serusers] SOP and Application Agent info
by Welesley Sibelson Dias 21 Jan '05

21 Jan '05

i wanna find more information about SOP (SIP Operator Pack (SOP)): Application Agent, but don't find nothing i send email to info(a)iptel.org but no receive response. somebody know this products from iptel. ,i wanna buy . thks a lot Welesley Sibelson Dias

1 0

[Serusers] forward on busy for certain subscribers
by Maria Yndefors 21 Jan '05

21 Jan '05

Hi, Is it possible to configure SER to forward calls that are busy only for subscribers that have ordered the service? I can't see that subscriber's forwarding information is stored in the database. Also does SER support privacy standards: RFC 3323, RFC 3325? Regards /Maria

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