- sr-users - kamailio.org

[Serusers] general sip problem
by arda balkanay 07 Feb '05

07 Feb '05

hi i have a sip network and when trying to make a call, caller can receive called party's voice but called guy cannot hear any voice. When called guy tries to make a call he can hear called party's voice and similarly new called guy cannot hear any voice. i wanna ask that what can be the reason about my problem Thanks best regards Arda Balkanay

2 1

Re: [Serusers] Modifying "From" header field of an INVITE message
by Andr�s Parra L. 07 Feb '05

07 Feb '05

Don't hesitate anymore, use asterisk to do so with the SetCallerID command in extensions.conf. You should forward the call through Asterisk from SER. If you don't want to have all the media pass through you, and i beleive you don't, use canreinvite=yes in sip.conf, you should fight a little bit with the codecs with allow and disallow but it should be fine. The next links could help you with the reinvitation issue: http://www.voip-info.org/wiki-Asterisk+SIP+media+path http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Hope it helps. Andres __________________________________ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com

3 2

[Serusers] Trusted IP and security.
by Tom Lowe 07 Feb '05

07 Feb '05

Hi all. I have a "security" question regarding "trusted IP's". Is it possible for someone to SUCCESSFULLY spoof an IP and actually make working calls? For example, '10.10.10.10' sends calls to SER (or any other proxy server) at 20.20.20.20, but actually spoofs the IP by sending an IP address of 30.30.30.30, which happens to be trusted by the SER at 20.20.20.20. I ask because I'm having a discussion with a vendor who is trying to tell me that using trusted IP's for SIP validation is insecure and easily hacked. I don't think it is because when SER gets an INVITE from 30.30.30.30, it is going to send it's progress messages to 30.30.30.30, regardless of the contents of the SIP messages....so the spoofer at 10.10.10.10 won't get any of the progress messages, and more importantly won't be able to establish a talk path. I suspect he may still cause SER to initiate some brief outbound calls, but they should fail when the SIP protocol falls apart. Does anyone have any thoughts on this? Tom

5 4

[Serusers] SER : no voice between two local sip phones
by fauzi＠aplio.com.my 06 Feb '05

06 Feb '05

Dear all, I had installed the SER Server at our own LAN. I had two sip phones connected locally via LAN. Both units can dialed each other without any problems but there are no voice on these two phones. Can you advise me on this. Dialing using Ip without going to the server had no problem at all but not when it use the server. Regards, fauzi

3 3

[Serusers] Permanent Registration
by support＠cybertel.biz 06 Feb '05

06 Feb '05

Hi, I would like to ask whether it is possible to allow permanent registration with the server i.e. Once the SIP client registered successfully with the server, the registration record will not expire until we delete it manually. Best Regards, Thomas

3 2

[Serusers] External Calls Through Ser & Asterisk !!!
by Felipe Martins 06 Feb '05

06 Feb '05

Hi everyone, I'm trying to make a call, based on, when a certain user auths my SER by mysql, if he dials "1", SER forwards the INVITE to Asterisk, The user dials "1" to open external calls and then the wanted number, doing this Asterisk forwards again to Go2call VoIP Server with a certain username and password, in order to auth my user and close the Tunnel, making the Call possible. What I want to know is that if its possible, or am I crazy !! I thinking of doing this by the following way: 1. Configure a Certain [context] block with the lines below at extensions.conf: a) Context "Go2Call_Calls": Extensions Description 1 Mark b) exten => <external_phone_number>,1,Answer exten => <external_phone_number>,2,Playback(tt-weasels) exten => <external_phone_number>,3,VoiceMail(45) exten => <external_phone_number>,4,Hangup 2. Configure a peer at sip.conf in order to my asterisk can talk to my SER Sip Proxy: [<external_phone_number>] type=peer username=<external_phone_number> insecure=yes canreinvite=no context=Go2Call_Calls mailbox=<external_phone_number> host=dynamic Will it work ??? Does it work this way, or am I getting crazy ? Is there anyone who have already made that ? I'm gonna be doing this till I get our Telecom Sip Numbers to company from our Telecom Provider. Anyone ?? -- 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

2 1

[Serusers] Phone rings but no audio
by Luca Corti 06 Feb '05

06 Feb '05

Hello, I'm running ser 0.8.14 and rtpproxy CVS on debian sarge. You can find my ser.cfg. below. I'm forwarding my calls to a Cisco AS5350 whish serves as a PSTN gateway. I can place a call to a PSTN number and the phne rings, but then no audio is sent or received. I use kphone as a client behind NAT. thnaks Luca --- ser.cfg routing --- 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; }; # !! Nathelper # Special handling for NATed clients; first,NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling 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 }; # 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"); rewritehostport("$AS5350_IP:5060"); 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("my.domain", "subscriber")) { www_challenge("my.domain", "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); } # -- nathelper -- route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs 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(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 1

[Serusers] SER + asterisk
by Patrick Baker 06 Feb '05

06 Feb '05

Hello, I'm trying to get asterisk to work with SER - I have ser forwarding the sip connection to asterisk with the following routing logic. if (uri=~"^sip:8[0-9]{7}.*") { forward( 10.0.18.3, 5061 ); break; Asterisk accepts the connection on port 5061 - but I get the following once it connects to asterisk. 001 -- Executing Answer("SIP/ast.digicen.com-08289858", "") in new stack 002 -- Executing Dial("SIP/ast.digicen.com-08289858", "IAX2/kknott@NuFone/12039063173") in new stack 003 -- Called kknott@NuFone/12039063173 004 -- Call accepted by 66.225.202.72 (format gsm) 005 -- Format for call is gsm 006 -- Hungup 'IAX2/NuFone/1' 007 == No one is available to answer at this time (1:0/0/0) 008 == Auto fallthrough, channel 'SIP/ast.digicen.com-08289858' status is 'NOANSWER' 009 > cdr_odbc: Query Successful! 010 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 011 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 012 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 013 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 014 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP headers... 015 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 016 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 017 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 018 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 019 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 020 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 021 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 022 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 023 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 024 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 025 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 026 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 027 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 028 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 029 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 030 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 031 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 032 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 033 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 034 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 035 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 036 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 037 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 038 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 039 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 040 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 041 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 042 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 043 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 044 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 045 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 046 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 047 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 048 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 049 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 050 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 051 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 052 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 053 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 054 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 055 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum retries exceeded on call 5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 52428 (Non-critical Response) 056 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum retries exceeded on call 5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 102 (Non-critical Request) 057 Jan 30 21:41:50 WARNING[9829]: chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' 058 059 *CLI> 060 *CLI> Asterisk will usually core dump after that - I have no idea where to start to fix it - No machines are behind NAT. Thanks, Patrick

4 3

[Serusers] is_from_local() definition
by Chris 06 Feb '05

06 Feb '05

Please can anyone tell me where I can get a complete and full definition of the criteria to fulfil function is_from_local() Thanks Chris -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.5 - Release Date: 03/02/2005

1 0

[Serusers] RTPPROXY in another PC
by Gustavo Villegas 06 Feb '05

06 Feb '05

dear users can i get the RTPPROXY runing in another PC, maybe in a CO-located PC in another country. all for the expensive cost of internet in mine. or maybe i need to move all , the server and the rtpproxy module to another PC ??? if yes, how can be done ?? some examples will be apreciates. Regards Gustavo Villegas

3 4

[Serusers] Multi server environment
by RICARDO AUGUSTO POPPI MARTINS 06 Feb '05

06 Feb '05

Hello all!! We are implementing a voip network in Brasil with 2 or more servers using asterisk and SER. We are still studying the configuration for it but it would be nice to know about similar implementations around the world. Is there anybody using a SER environment where a registered SIP client from sever A can conect directly (peer to peer) to a user registered in a server B? Thanks in advance for this case report. Rgds, Ricardo Poppi ______________________________________________________________________ Giro, a Internet rápida que acompanha você. http://www.giro.com.br

1 0

[Serusers] ALIAS not online after ser restart
by Stefano 05 Feb '05

05 Feb '05

Hi all, i have a little problem with aliases, that is: I have created some numeric aliases for some users with serctl in this way: # serctl alias add 1101 sip:user@mydomain.foo the command goes right, i can reach the correct users using aliases and i can see that records are correctly inserted in the "aliases" table in the mysql database. The problem is that when i reboot ser aliases don't work even if they are correctly in the aliases tables; it seems they are not online or not registered. The routing script correctly looks for users in the "aliases", but not after restart! How can i fix this problem? Thanks for your help! Stefano __________________________________________________________________ Tiscali Adsl 3 Mega Flat, 3 MESI GRATIS! Con Tiscali Adsl 3 Mega Flat navighi in Rete alla supervelocita' a soli 29.95 euro al mese senza limiti di tempo. Attivati entro il 15 Febbraio 2005, 3 MESI sono GRATIS Scopri come http://abbonati.tiscali.it/adsl/sa/2flat_tc/

2 1

Re: [Serusers] bug in auth db
by info＠beeplove.com 05 Feb '05

05 Feb '05

Didn't get any response about this bug ! :-(( MOhammad info(a)beeplove.com wrote on 02/04/2005, 12:32:02 AM: > > My mysql database server is different computer than my ser server. > ser was not able to connect to database. > > I put some debug code in authdb_mod.c > > static int child_init(int rank) > { > db_handle = db_init(db_url); > LOG(L_ERR, "authdb_mod.c#child_init(): db_url: %s\n", db_url); > if (!db_handle) { > LOG(L_ERR, "auth_db:init_child(): Unable to connect > database\n"); > return -1; > } > return 0; > > > } > > I have found db_url is not getting the value from my ser.cft. > Its getting the default value from config.h that was defined before > compiled. > > > I compiled my ser after editing config.h with mysql param that match my > situation and my ser is up !! > > Thanks, > > -- > Mohammad Khan > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers -- Mohammad Khan sip:beeplove@iptel.org

1 0

[Serusers] proxy don't query the location table?
by szj 05 Feb '05

05 Feb '05

Hi, I put the proxy and registar at the two machine. when proxy deal with calling, I found that proxy don't query the callee information from the database. I sniffed the data traffic with ethereal. the Ser log said in the registrar/lookup.c lines 81 that: "lookup(): '8644' Not found in usrloc". What I want to know when the proxy and registrar are located on the different devices. how can the registrar tell the proxy the new registered sip ua if the proxy didn't look the callee in the location table. If I am not wrong, the usrloc function get_urecord() did not query the table for every coming call. It attempts to find callee in a hash table. Can that guarantee to find the newest information in the database??? Thanks for your instructions. Best Regards. Zongjun

1 0

[Serusers] use_domain is 0!
by szj 05 Feb '05

05 Feb '05

Hi, Daniel: Thanks for your kind help! but I check it. Right, use_domain is 0. Do you have any further advance? Thanks again. Best Regards. Sun Zongjun

1 0

[Serusers] SER stateful
by Vitor Brasileiro 04 Feb '05

04 Feb '05

Hi list, I'm in a doubt about stateful vs. stateless ser operation and accounting issues. Using stateless mode, accounting informations became different? There's other important thing about it? What kind of method should I use? Thanks, Vitor Brasileiro.

3 2

[Serusers] rtpproxy mess
by Christian de Balorre 04 Feb '05

04 Feb '05

Hello, I am struggling with rtpproxy and I would like some help, because the documentation is rather scarce to be honest. I am running ser on an OpenBSD 3.3 box, with one internal interface (192.168.x.x) and a public interface (62.x.x.x). I am not sure wether I have to force rtpproxy to listen to a specific interface or not. I launch rtpproxy rather basically with ./rtpproxy -s /var/run/rtpproxy.sock Then I launch ser, it spawns some process and then run into race condition (see ps -aux below) Here is log file : Feb 2 15:34:28 cdg8-ser1 ser: WARNING: could not read from /dev/random (5) Feb 2 15:34:28 cdg8-ser1 ./ser[20970]: ERROR: send_rtpp_command: can't connect to RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[32404]: ERROR: send_rtpp_command: can't connect to RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[32404]: WARNING: rtpp_test: can't get version of the RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[32404]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Feb 2 15:34:28 cdg8-ser1 ./ser[20970]: WARNING: rtpp_test: can't get version of the RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[20970]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Feb 2 15:34:28 cdg8-ser1 ./ser[147]: ERROR: send_rtpp_command: can't connect to RTP proxy ... And the ps USER PID %CPU %MEM VSZ RSS TT STAT STARTED TIME COMMAND ser 16586 99.0 0.5 1744 2672 ?? R 3:34PM 6:15.11 ./ser -P /var/run/ser.pid root 16613 0.0 0.1 752 272 ?? Ss 3:33PM 0:00.01 ./rtpproxy -s /var/run/rtpproxy.sock #./rtpproxy -v 20040107 # ./ser -V version: 0.8.14 (i386/openbsd) Nathelper config : #-------------- nathelper parameters modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") Could it be that ser running as ser user cannot read the rtpproxy socket (running as root) ? Thanks Christian

4 7

[Serusers] Problems Compiling SEMS on Solaris 10.
by Sergio Armando Gutierrez Betancur 04 Feb '05

04 Feb '05

Good afternoon Currently, we are triyng to compile SEMS on a Sun FIRE 220R with Solaris 10 build 72, but we are having a kind of conflict with the definition of a structure in the file AmApi.cpp We are using gcc 3.3.2, and gnu make. The error is the following. -------------- Begin compiling output --------- # make make[1]: Entering directory `/export/home/operador/answer_machine' ...... Some output ommited: g++ -c -o AmApi.o AmApi.cpp -Wall -Wno-reorder -D_DEBUG -g -DTHREAD_SAVE -D_REENTRANT -D_THREAD_SAFE In file included from /usr/include/netinet/in.h:62, from AmRtpPacketTracer.h:7, from AmRtpStream.h:34, from AmSession.h:31, from AmApi.h:32, from AmApi.cpp:28: /usr/include/sys/stream.h:67: error: `struct queue' redeclared as different kind of symbol /usr/local/include/c++/3.3.2/bits/stl_queue.h:71: error: previous declaration of `template<class _Tp, class _Sequence> class std::queue' make[1]: *** [AmApi.o] Error 1 make[1]: Leaving directory `/export/home/operador/answer_machine' make: [all] Error 2 (ignored) -------------- End compiling output --------- Does somebody have any hint about this? Thanks in advance for your help. Sergio Armando Gutiérrez Betancur Equipo GSC Área Operación Nuevos Servicios Subgerencia DSS Telco. Gerencia Telco Empresas Publicas de Medellín E.S.P. * (574)4150540 * sgutierb(a)eeppm.com -----Mensaje original----- De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] En nombre de serusers-request(a)lists.iptel.org Enviado el: Viernes, 04 de Febrero de 2005 04:20 p.m. Para: Sergio Armando Gutierrez Betancur Asunto: Welcome to the "Serusers" mailing list Welcome to the serusers(a)lists.iptel.org mailing list! To post to this list, send your email to: serusers(a)lists.iptel.org General information about the mailing list is at: http://lists.iptel.org/mailman/listinfo/serusers If you ever want to unsubscribe or change your options (eg, switch to or from digest mode, change your password, etc.), visit your subscription page at: http://lists.iptel.org/mailman/options/serusers/sgutierb%40eeppm.com You can also make such adjustments via email by sending a message to: serusers-request(a)lists.iptel.org with the word `help' in the subject or body (don't include the quotes), and you will get back a message with instructions. You must know your password to change your options (including changing the password, itself) or to unsubscribe. It is: sagutier Normally, Mailman will remind you of your iptel.org mailing list passwords once every month, although you can disable this if you prefer. This reminder will also include instructions on how to unsubscribe or change your account options. There is also a button on your options page that will email your current password to you. "Este mensaje y/o sus anexos son para uso exclusivo de su destinatario intencional y puede contener información legalmente protegida por ser privilegiada o confidencial. Si usted no es el destinatario intencional del mensaje por favor infórmenos de inmediato y elimine el mensaje y sus anexos de su computador y sistema de comunicaciones. Igualmente, le comunicamos que cualquier retención, revisión no autorizada, distribución, divulgación, reenvío, copia, impresión, reproducción, o uso indebido de este mensaje y/o sus anexos, está estrictamente prohibida y sancionada legalmente. Agradecemos su atención. Empresas Públicas de Medellin E.S.P."

1 0

[Serusers] Radius Authentication Problem
by M.V. Jaga Mohan 04 Feb '05

04 Feb '05

Hi List, I have downloaded freeradius 0.9.1 and SER-0.8.14 source and radiusclient library version 0.4.2. compiled all of them and my radius is working well as I have tested with Radtest. I have compiled SER with radius module and I have got auth_radius.so also. But when I am trying to run ser I am getting error called : init_mod() : Error while initializing module. Kindly help me in this regard. Its very urgent. Below I have provided the debug log. 0(3672) WARNING: could not rev. resolve 192.168.5.1 Listening on 127.0.0.1 [127.0.0.1]:5060 192.168.5.1 [192.168.5.1]:5060 Aliases: localhost:5060 localhost.localdomain:5060 WARNING: no fork mode and more than one listen address found (will use only the the first one) 0(3672) DEBUG: init_mod: sl_module stateless - initializing 0(3672) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(3672) DEBUG: MD5 calculated: b27e1a1d33761e85846fc98f5f3a7e58 0(3672) DEBUG: init_mod: tm 0(3672) TM - initializing... 0(3672) Call-ID initialization: '7fbeafda' 0(3672) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(3672) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(3672) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(3672) DEBUG: lock_initialize: lock initialization started 0(3672) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(3672) DEBUG: MD5 calculated: 533cb9e91f4b999cf76861cbb9ed54ed 0(3672) DEBUG: MD5 calculated: a6a1c5f60faecf035a1ae5b6e96e979a 0(3672) DEBUG: init_mod: rr 0(3672) rr - initializing 0(3672) DEBUG: init_mod: maxfwd_module Maxfwd module- initializing 0(3672) DEBUG: init_mod: usrloc 0(3672) usrloc - initializing 0(3672) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(3672) DEBUG: init_mod: registrar 0(3672) registrar - initializing 0(3672) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(3672) find_export: found <ul_bind_usrloc> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_register_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_get_all_ucontacts> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_insert_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_delete_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_get_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_lock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_unlock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_release_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_insert_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_delete_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_get_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_update_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) DEBUG: init_mod: auth 0(3672) auth module - initializing 0(3672) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(3672) DEBUG: init_mod: auth_radius 0(3672) auth_radius - Initializing 0(3672) auth_radius: Error opening configuration file 0(3672) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules 0(3672) DEBUG: tm_shutdown : start 0(3672) DEBUG: tm_shutdown : empting DELETE list 0(3672) DEBUG: tm_shutdown : empting hash table 0(3672) DEBUG: tm_shutdown: releasing timers 0(3672) DEBUG: tm_shutdown : removing semaphores 0(3672) DEBUG: tm_shutdown : done 0(3672) shm_mem_destroy 0(3672) destroying the shared memory lock and My ser.conf file is like : # ----------- 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=9 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" # ------------------ 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" # 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" loadmodule "/usr/local/lib/ser/modules/auth_radius.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") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("auth_radius", "service_type", 15); # -- 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==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("192.168.5.1")) { www_challenge("192.168.5.1","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(); }; } =====

2 1

[Serusers] How to make precall announcements?
by Hauke Zuehl 04 Feb '05

04 Feb '05

Hello all :) I have a problem which I do think about for some days now. The situation is: I have one server running SER and another one with Asterisk. Some users (which are authenticated by SER) want to have an announcement before they do a call. An example: A user wants to know how much he has to pay for a call. So he piks up the phone, logs into SER and dials a number. Now a voice (spoken via Asterisk) should tell him: "Hey, your call willl cost 2 Cents per minute". Then the call should be given back to SER by Asterisk and SER routes the call. Another user does not want to hear any information so all calls will be routed directly. I thought about a script that looks into a database which user has booked this neat feature. That works! On Asterisk I have created an extension called "precall_<user>" (<user> is - of course - user's id). Now I want SER to rewirte <user> to precall_<user>, jump to Asterisk, afterwards SER snips precall_ and does the call (oh dear...so much "call"-ing *g*). I have following part of ser.cfg: if(method==INVITE) { # # some other stuff # ..... rewriteuser("precall_test_hauke"); rewritehost("asterisk.mydomain.de"); log(1,"MYLOG: Forward to precall\n"); forward(uri:host, uri:port); strip(8); log(1,"MYLOG: Do call\n"); forward(uri:host, uri:port); }; Unfortunately SER seems to try to do a new INVITE while forwarding!? But great, I see in Asterisk the incoming call to precall_test_hauke :) But then the call hangs somewhere :( Would be great if anyone has a hint or a link with an example for me :) Thanks and kind regards, Hauke This is done in ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

2 1

[Serusers] Dynamic Redirect Issue about SER!
by George Lee 04 Feb '05

04 Feb '05

SER gurus: I am an newbie to SER usuage. At present a dynamic redirect issue happens to me. But it is very pity for SER not to support this case!! Problem Description: ==================== When configuring SER as only redirect server, config file regarding redirect routing section is partly writen as follows: #-----------------------request routing logic------------------------- #main routing logic if (method=="INVITE") { #rewrite current URI, which is always part of destination ser rewriteuri("sip:80000000@192.168.0.191:5060"); #redirect now sl_send_reply("301", "Redirect"); break; } #--------------------------------------------------------------------- where, redirecting destination URI is staticly configed into SER, but dynamic destination URI is what our IP-PBX product(RTCCP) want to hope. The following call flow diagram helps to understand above mentioned scenario. GrandStream HandyTone486 SER Our B2BUA IP PBX(RTCCP) IP: 192.168.0.253 IP: 192.168.0.252 IP: 192.168.0.191 +++++++++++++++++++++++++++++++++++++ First Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F1(INVITE)---------->| |<-----------------F2(302)------------>| |-----------------F3(ACK)------------->| |------------------------------------F4(INVITE)------------------------>| ( The subsequent call flow is omitted ) +++++++++++++++++++++++++++++++++++++ Second Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F5(INVITE)---------->| |<-----------------F6(302)------------>| |-----------------F7(ACK)------------->| |------------------------------------F8(INVITE)------------------------>| ( The subsequent call flow is omitted ) F1: INVITE sip:8000000@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=c526ecce44ec6d75 To: <sip:8000000@192.168.0.252;user=phone> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: d6169571fe4f59a2(a)192.168.0.233 CSeq: 50105 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F2: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=c526ecce44ec6d75 To: <sip:8000000@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed Call-ID: d6169571fe4f59a2(a)192.168.0.233 CSeq: 50105 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6155 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:8000000@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1" F4: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58 From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=1be4a5d7d169d8f1 To: <sip:80000000@192.168.0.191:5060> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: 0d325b0e778321ee(a)192.168.0.233 CSeq: 17809 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F5: INVITE sip:80000002@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4 To: <sip:80000002@192.168.0.252;user=phone> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: c755c6b0908a8ce9(a)192.168.0.233 CSeq: 25248 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F6: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4 To: <sip:80000002@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906 Call-ID: c755c6b0908a8ce9(a)192.168.0.233 CSeq: 25248 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6151 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:80000002@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1" F8: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=58a90f4229e70a80 To: <sip:80000000@192.168.0.191:5060> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: 2a79d9ac84e2f72e(a)192.168.0.233 CSeq: 62121 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 In fact, I think SER should retrieve user portion of header "to" URI of Invite message(for above example it should be 80000000 for first call,whereas 80000002 for second call) to construct userinfo of destination redirect uri for header contact of 3xx response(but hostinfo maybe get from SER config file). Meanwhile I study into sl_send_reply source code, but no progress to issue. So any suggestions or solutions are appreciated!! Thanks in advance!! George Lee(ShenZhen, CHINA)

3 5

[Serusers] Can't find the latest registered users continued
by szj 04 Feb '05

04 Feb '05

Hi, All: I forget to remember one thing, My proxy domain name is sip.voiptest.com, registrar is voiptest.com. All of them are translated by my DNS. so registered URI is: sip:861111@voiptest.com while INVITE URI is: 861111(a)sip.voiptest.com. Does that have influence on the Usrloc and registrar modules. Usrloc only uses the username as a index to lookup. Thanks a lots. Best Regards. Sun Zongjun

3 2

[Serusers] Success Report inside LAN
by info＠beeplove.com 04 Feb '05

04 Feb '05

Hello, I have succeessfuly installed SER and was able to register from Windows Messenger 5.1 in my LAN. I have two computers, one for SER and another for MySQL. Next, I will try to connect to SER from outside my LAN. All computers in my LAN have private IPs. I use NAT in my router. Please suggest me, what documents/readme I would read. Thanks, -- Mohammad Khan sip:beeplove@iptel.org

1 0

[Serusers] new MediaProxy release 1.2.1
by Adrian Georgescu 04 Feb '05

04 Feb '05

Hello, this is a bug fixing release for the media relay part, the obvious improvement is the relaying of video streams, SER module remains unchanged. Changes from version 1.2.0 to 1.2.1 ------------------------------------------------- - Fixed payload=101 (Telephone event) as being wrongly interpreted as codec - Fixed memory leak caused by cyclic references between objects - Fixed memory leak caused by exception tracing - Fixed typo in variable name for counting idle time - Fixed issue with jerky video To download: http://mediaproxy.ag-projects.com/mediaproxy-1.2.1.tar.gz To upgrade, unpack over your existing installation and restart mediaproxy. Regards, Adrian Georgescu

3 4

[Serusers] CVS Serweb works with which versions ?
by mr. barker 04 Feb '05

04 Feb '05

Does the CVS head of Serweb work with which versions of Serweb ? 0.8.14 ? 0.9.0 ? If not which ver of Serweb does work with each ? I have tried the one that comes with 0.8.14 and can not get that version to work at all. I have followed all the read me's and looked in the archieves . I have come to the conclusion that this version is in need of a lot of patching or something. I am able to get the interface up .. however it continually comes up with the wrong host name no matter what I do . I manally enter into the http.conf file but still does not access the db correctly and comes up with invalid user name etc .. For some reason it keeps calling up the machines name that I gave to it when I installed Linux Fedora Core 3 Other then that the rest of the system runs fine Ser is good but SerWeb seems to need work. The CVS head of SerWeb seems to quite a bit more flexable .. in the configuration files. I can get that up and going also however when it logs in I get absolutly no tabs showing up at all .. no error messages either. Racking my brain out here any help would be appreciated.. thanks

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users