- sr-users - kamailio.org

[Serusers] A-Z route
by mohammad 09 Jan '05

09 Jan '05

Hi ALL; A-Z route over H323 is available. 1) several backup routes per one destination 2) availability of test 3) technical support Plz contact me offlist for rate and further informations MSN:hezare3@hotmail.com Mohammad

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[Serusers] ser-0.9.0 and rpid
by Helge Waastad 09 Jan '05

09 Jan '05

Hello, is there a bug in append_rpid_hf in ser-0.9.0 ? I get ascii characters in my rpid even if MySQL version 4.1.8 I've added the INVITE from ser to the gateway (cisco) [removed some private information...] The Remote-Party address in subscriber table is 67512387 and comes out like 'edc allo'........... best regards, hw INVITE sip:<name>@<cisco>:5060 SIP/2.0 Record-Route: <sip:<ser>;ftag=1952781227;lr> Via: SIP/2.0/UDP <ser>;branch=z9hG4bKc863.7c873544.0 Via: SIP/2.0/UDP 10.0.0.4:5060;received=<client IP>;rport=64532;branch=z9hG4bK5497F64D2C5444D0BDF03D0F9B7E38E3 From: <name> <sip:<name>@<relam>;tag=1952781227 To: <sip:<destination>@<realm>> Contact: <sip:<name>@<client IP>:64532> Call-ID: 1E1BCEDE-DE4C-4B5D-A0F7-D6F3048427B1(a)10.0.0.4 CSeq: 59604 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 298 Remote-Party-ID: <sip:edc allo@<realm>;party=calling;screen=no;privacy=off <mailto:allo@<realm>;party=calling;screen=no;privacy=off> P-RTP-Proxy: YES P-NATed-Calee: YES P-hint: GATEWAY v=0 o=67830064 202717921 202717968 IN IP4 10.0.0.4 s=X-Lite c=IN IP4 <ser> t=0 0 m=audio 35706 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15

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[Serusers] DTMF problems
by Humberto Aicardi 09 Jan '05

09 Jan '05

Hi, I'm a newbie on SER so please be patient. I have been using * for quite a while and now decided to use SER as a front-end for my SIP users. I have everything setup including *, SER, RPTProxy, STUN and everything works except for DTMF tones, for example, when I access * voicemail through SER the tones do not get correctly recognized by *. Asterisk is configured to use RFC2833 as a transport for DTMF tones. Can anyone please help me solve this issue? Thanks in advance, Humberto

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RES: [Serusers] DTMF problems
by Humberto Aicardi 09 Jan '05

09 Jan '05

Hello, My current configuration is: RedHat 9.0 Asterisk HEAD version SER latest CVS version www.vovida.org STUN server eyeBeam XTen softphone When I dial from the softphone, which is connected to the SER server, to the * box (voicemail) it will request to input the mailbox number and password. When I try to do this the * box get some but not all digits I have pressed, so basically, it looses some digits. When I'm connected directly it works ok. Here is the context in * for the SER server: [300] type=peer host=xxx.xxx.xxx.xxx nat=no disallow=all allow=g729 allow=gsm allow=ulaw dtmfmode=rfc2833 qualify=yes context =from-ser ; Default for incoming calls insecure=very If you need any further any information please let me know. Thanks, Humberto -----Mensagem original----- De: Yair Hakak [mailto:yhakak@gmail.com] Enviada em: Saturday, January 08, 2005 1:37 PM Para: Humberto Aicardi Assunto: Re: [Serusers] DTMF problems Hello, what endpoints are you using? -yair p.s. not that it should have any relevance to the question, but what stun server are you using, and are you running it on the same machine as asterisk and/or SER? I am also running an asterisk/SER combination and I want to run my own STUN server as well. On Sat, 8 Jan 2005 10:23:17 -0200, Humberto Aicardi <humberto(a)aicardi.com.br> wrote: > > > Hi, > > > > I'm a newbie on SER so please be patient. I have been using * > for quite a while and now decided to use SER as a front-end for my SIP > users. I have everything setup including *, SER, RPTProxy, STUN and > everything works except for DTMF tones, for example, when I access * > voicemail through SER the tones do not get correctly recognized by *. > Asterisk is configured to use RFC2833 as a transport for DTMF tones. > > > > Can anyone please help me solve this issue? > > > > Thanks in advance, > > Humberto > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Serusers] SEMS Horoscope
by Ian Bonham 08 Jan '05

08 Jan '05

Come on guys, when I compile SEMS it says a thing about horoscopes! HOW do I get this workin? so 901 is Gemini, 902 is Taurus and so on! Cheers, Bon

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[Serusers] Using SER in conjunction with Asterisk questions
by Nate Kapi 08 Jan '05

08 Jan '05

Hello, I'm looking at setting up SER along with an Asterisk PBX installation and I have a few questions, as I've never used SER before. What is an ideal setup for SER? I was going to use Redhat 9 and have SER on one machine and Asterisk on another. Is this best or should they both be on one machine? How will SER handle the requests for the PSTN and internal extensions setup on the Asterisk machine? How should asterisk register with SER and SER with asterisk? Sorry if these questions are to broad. Any asterisk + SER installation tips or links are appreciated. Thanks in advance!

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[Serusers] my_account page never completes...
by Shavkat Berdiev 08 Jan '05

08 Jan '05

Hello, After logging in as admin, when I click on the account link at the end of the line, I'm taken to a new page but that page never completes loading. No text is ever displayed. It appears to be hanging at the connect_to_db() just inside the start of the while in my_account.php. -- Best regards, Shavkat Berdiev shavkat(a)2oxygen.uz

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[Serusers] (no subject)
by Ian Bonham 08 Jan '05

08 Jan '05

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[Serusers] Place call using ser
by Chris HARIGA 07 Jan '05

07 Jan '05

Hi, I install ser 0.8.14 on my server and I register 2 IP phones. I setup the nathelper and everything work fine. The problem is: how I can place calls between this 2 ip phones? One has username "phone1" and the second one "phone2". How I can make calls? I have Asterisk in the same server, running on 5070, can I use my * for dialplan? Best regards, Chris HARIGA

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[Serusers] SerWeb Registration
by Felipe Martins 07 Jan '05

07 Jan '05

Hi everyone, I've configured SERWEB, apache and mysql, everything seems to work fine. I enter the page from serweb, see the login and password fields, and also the SUBSCRIBE field, click at subscript, fill all information, but when I click REGISTER, all the screen becomes WHITE with nothing written on it ... When I try to make another user, with the same name it tells that the user already exists. So I tried to authenticate at home screen with login and password that I made at the later step, but it tells me "Bad username or password". Well, somehow it's working but I see nothing on the screen and i can't login either. For this problem I have some questions. Where are being saved the login and password information, i can't see neither of them at mysql. What am doing wrong ? -- 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 installation error
by Nguyen, Thuan (N-EKD Systems) 07 Jan '05

07 Jan '05

I was trying to create the SER database in MySQL. I ran the command: ser_mysql.sh create I entered the password and got this error message: ser_mysql.sh: line 1: gen_ha1: command not found HA1 calculation failed And yet the executable file gen_ha1 is present in the same folder as ser_mysql.sh Could anyone help? Thanks very much. Thuan

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[Serusers] softphones
by Joao Pereira 07 Jan '05

07 Jan '05

Hi, can someone tell be about some good and free softphones? Are they easy to use by non-tecnical users? Can someone share their experience about the implementation of VoIP softphones in a company?.... because usualy people dont want to make changes in the way they work.... I would like to know a way to convince peaple in my company to use them. Thanks Joao Pereira

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[Serusers] V_M and SMTP Auth
by Ian Bonham 07 Jan '05

07 Jan '05

Hi all, I'm sure someone will be able to advise me on this. I'm using SEMS as my Voicemail service, and in this case I have to relay the voicemail messages through an Authorisation SMTP server. I would use sendmail on localhost, but if I try to send a message to my own email address using localhost on 1&1's servers the message never gets to me :( Is there any way to edit the AmMail.cpp to allow it to handle auth'd SMTP? If so, can anyone let me know the procedure please? Then I can send my mail through my 1&1 SMTP server Many thanks,. Ian Bonham

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[Serusers] b2bua + SER + RADIUS
by Lucas Aimaretto 07 Jan '05

07 Jan '05

I've got some doubts with b2bua. If any body knows, I'd like to ask the following .. This is the configuration. SER: 192.168.1.252:5060 b2bua: 192.168.1.253:5065 Radius-auth: 192.168.1.253:1812 Radius-acct: 192.168.1.253:1813 Tel SIP: 192.168.1.178:11005 (Note: I've SER, RADIUS and B2B running on the same PC. To identify each service, my linux has 2 IP address, 192.168.1.253 and 192.168.1.252) I've sniffed the net ... and got this. Msg time ms sniffing 1 12:41:39 606789 192.168.1.178.11005 > 192.168.1.253.5065: udp 2 2 12:41:39 625476 192.168.1.178.11005 > 192.168.1.253.5065: udp 465 3 12:41:39 793503 192.168.1.252.5065 > 192.168.1.252.5060: udp 422 4 12:41:39 800937 192.168.1.252.5060 > 192.168.1.252.5065: udp 673 5 12:41:39 990283 192.168.1.253.5065 > 192.168.1.178.11005: udp 436 6 12:41:39 995814 192.168.1.178.11005 > 192.168.1.253.5065: udp 646 7 12:41:40 184280 192.168.1.252.5065 > 192.168.1.252.5060: udp 603 8 12:41:40 203387 192.168.1.253.41011 > 192.168.1.253.radius: rad-access-req 271 [id 5] 9 12:41:40 368387 192.168.1.253.radius > 192.168.1.253.41011: rad-access-accept 52 [id 5] 10 12:41:40 374554 192.168.1.252.5060 > 192.168.1.252.5065: udp 629 11 12:41:40 586069 192.168.1.253.5065 > 192.168.1.178.11005: udp 391 12 12:41:49 632154 192.168.1.178.11005 > 192.168.1.253.5065: udp 2 Which is, basically ... TEL SIP B2B SER RADIUS |------(1)----->| | | |------(2)----->|----(3)--->| | | |<---(4)----| | |<-----(5)------| | | |------(6)----->| | | | |----(7)--->| | | |-----------+--(8)------->| | |<----------+--(9)--------| | |<---(10)---| | |<----(11)------| | | |-----(12)----->| | | Well, the Tel SIP got registered at the SIP SERVER (SER), as usual, ... and I have authentication, with radius, configured in SER to be used at REGISTER time ... [...] 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; }; [...] QUESTION: Why is B2B talking to RADIUS instead of SER ??? Should not be SER the one talking to RADIUS ??? If it is really obvious, do not get mad at me ... ;-) Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005

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[Serusers] mediaproxy functions client_nat_test()
by Gurol Akman 07 Jan '05

07 Jan '05

take a look at the Readme file under ser-0.8.14/modules/mediaproxy dir. hope it helps -- GA

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[Serusers] softphones
by Gurol Akman 07 Jan '05

07 Jan '05

You can also try http://www.freeworldialup.com/ for dowloading FWD. Regards -- GA

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[Serusers] serweb pconnect failure
by Gurol Akman 07 Jan '05

07 Jan '05

Hi, I'm using ser-0.8.14 & serweb_2004-07-27 and have the following problem. When I try to access the http://host/.../admin/index.php page, I get the following error: Warning: mysql_pconnect(): Access denied for user: 'ser(a)host.private.com' (Using password: YES) in /usr/local/serweb/phplib/db_mysql.inc on line 73 Database error: pconnect(192.168.1.2, ser, $Password) failed. MySQL Error: () Session halted. Can anyone offer any advice? Thanks, GA

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[Serusers] mediaproxy functions client_nat_test()
by proson cheng 07 Jan '05

07 Jan '05

Hello all, I asked the questions but i still don't find an appropiate answer, in the fuctions. client_nat_test("3"); it takes an argument usually, i see a "3" for the arguments. what does that 3 mean? does it represent a type of NAT? If so, what others argument can I take? thanks proson .

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[Serusers] Ser Postgresql Problems
by RAceala andrei 07 Jan '05

07 Jan '05

I am trying to setup ser to use a posgresql databse. Everythig is ok until i try to register to SIP server with SjPhone when i get the following errors: PG[129] connect_db Error while parsing it Register PG[488] begin_transaction no connection, FATAL -3! get_ha1(): Error while querying database Any ideeas ? This is my ser.cfg # # $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=3 fork=no log_stderror=no listen=193.***.***.*** 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=test.ro # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/postgres.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" # 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_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 0) # -- 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") modparam ("auth_db","db_url", "sql://blahblah:blahblah@localhost:5432/ser") # -- 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 > 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=~test.ro) { log (1,"Cerere pt test.ro\n"); if (method=="REGISTER") { log (1,"Am primit Register\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("test.ro", "subscriber")) { www_challenge("test.ro", "0"); break; }; save("location"); log (1,"User Existent\n"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log (1,"User Inexistent\n"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }

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RE: [Serusers] Problem with ACK on Negative Reply
by PHAN, Quang-Minh (Docteur Ordinateur) 07 Jan '05

07 Jan '05

Hi Jiri, David, Thank you very much for your timely replies. I will check if the ACK hits t_relay or not in my configuration. I have skimmed through the transaction matching code in SER but I don't remember if it checks the FROM field or the FROM tag. I'll verify this when I can. Minh _____ De : Wearne, David W. [mailto:DWWearne@west.com] Envoyé : jeudi 6 janvier 2005 20:56 À : PHAN, Quang-Minh (Docteur Ordinateur); serusers(a)lists.iptel.org Objet : RE: [Serusers] Problem with ACK on Negative Reply I ran into exactly the same problem in December. There are several reasons why SER may not realize the ACK belongs to the transaction. Look carefully at the via: and from: headers in the INVITE and the ACK. The via: should have a branch parameter, and it should be the same in both. Some UAs do not put in the branch parameter. In this case SER should (according to RFC3261) be looking at the combination of the Request URI, To tag, From tag, Call-ID, Cseq, and top via header. In my case the problem was that SER compares the >From field rather than the From tag, and it does so incorrectly. I raised this issue both here and in the serdev list, (you can find the thread in the December archives), but the developers said they would not fix this because it is in support of an obsolete spec (RFC 2543). I recommend you check those headers to see if your situation actually matches what I ran into, and if so, raise the issue to the serdev list. If enough people complain, they might decide to provide better support of RFC3261, section 17.2.3 (Matching Requests to Server Transactions). Dave _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of PHAN, Quang-Minh (Docteur Ordinateur) Sent: Thursday, January 06, 2005 12:10 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Problem with ACK on Negative Reply Hi all, When I use SER in transaction stateful mode, I have the following problem: When A sends an "INVITE B" to SER, SER forwards the INVITE to B statefully. If B reply with "200 OK", everything goes fine as SER will delete the transaction from memory when it sees the 200 OK. However, if B send an "BUSY HERE" back to SER, SER will send an ACK to B right away and then forwards the "BUSY HERE" to A. A will then send an ACK to SER but SER won't realize that the ACK belong to the transaction and forward it to B. SER will then keep sending "BUSY HERE" to A until the transaction is timed out. I wonder if there is something to add in the configuration file to make SER understand that the ACK send from A should terminate the transaction. Any help will be highly appreciated! Minh

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[Serusers] nat
by kcassidy＠kakelma.mine.nu 07 Jan '05

07 Jan '05

Hi All, I've successfully installed media proxy and it's working with ser.0.8.14 just fine. I'm using the ser config that comes with media proxy module. The thing is I don't want all my RTP streams to pass thru the media proxy (e.g. Clients using public IP). I've tried the isflagset command but it doesn't seem to work. I've placed a isflagset(2) in my config below but all my traffic doesn't pass thru the media proxy. TIA. 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 (!www_authorize("", "subscriber")) { www_challenge("", "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(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") && (isflagset(2)) { 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; }; # 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") && (isflagset(2)) { use_media_proxy(); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; }

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[Serusers] how can i get the standard SIP CallFlow
by Joe 07 Jan '05

07 Jan '05

hello,all I am a newer in SIP.How Can i get SIP Standard CallFlows ! Thanks Best regards Joe

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[Serusers] how to compile SER + RADIUSCLIENT + RADIATOR (radius server, not freeradius)
by dangtien 07 Jan '05

07 Jan '05

Dear all, How to compile ser + radius RADIATOR and Cisco AS 5350 step by step Help me... Thank all

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[Serusers] how to compile SER + RADIUSCLIENT + RADIATOR (radius server, not freeradius)
by dangtien 07 Jan '05

07 Jan '05

Dear all, How to compile ser + radius RADIATOR and Cisco AS 5350 step by step Help me... Thank all

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[Serusers] RTP protocol with NAT
by Gustavo Villegas 06 Jan '05

06 Jan '05

Users the nat in Ser is managed by the SIP message, i was wondering if is theer a solution for managing the NAT in RTP protocol as the RTP is a trasport layer protocol All the best Gustavo Villegas

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