sr-users July 2004

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178 participants
449 discussions

[Serusers] Mediaproxy Problems
by Adrian Georgescu 08 Jul '04

08 Jul '04

Could you please upgrade to latest 1.1.0 (cvs update and copy the mediaproxy directory to stable tree and dload latest mediaproxy server software from http://mediaproxy.ag-projects.com/mediaproxy-1.1.0.tar.gz. Several bugs have been fixed in this release, your problem might be among those fixed. Adrian >> Hello List, I have the following setup - kphoneA------------- [SER A] ---------------- [SER B] ---------- kphoneB I am forcing both SER A and B to use mediaproxy regardless of whether the UA's are natted or not (ser.cfg file attached). When kphoneA makes a call to kphoneB, use_media_proxy() works fine on SER A. But on SER B, it fails (I can see an error - mediaproxy: Call-Id not found - but ethereal traffic dump shows that the Call-Id is present in the SIP message). So I get a call between khponeA---serA---kphoneB. When I initiate the call from kphoneB, the whole thing reverses and it becomes kphoneB---serB----kphoneA. I am using ser-0.8.12 (cannot use cvs version at the moment due to some code changed we have done) and mediaproxy-1.0.1. Anyone got any clue where I am wrong? ser.cfg file attached. Any help will be greately appreciated. Regards, Dhiraj

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[Serusers] bandwidth allocation
by Jesus Bernardo Ruiz Flores 08 Jul '04

08 Jul '04

Is there a way to reserve xxx KB for sip telephony. Thanks

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[Serusers] Mediaproxy Problems
by dhiraj.2.bhuyan＠bt.com 08 Jul '04

08 Jul '04

Hello List, I have the following setup - kphoneA------------- [SER A] ---------------- [SER B] ---------- kphoneB I am forcing both SER A and B to use mediaproxy regardless of whether the UA's are natted or not (ser.cfg file attached). When kphoneA makes a call to kphoneB, use_media_proxy() works fine on SER A. But on SER B, it fails (I can see an error - mediaproxy: Call-Id not found - but ethereal traffic dump shows that the Call-Id is present in the SIP message). So I get a call between khponeA---serA---kphoneB. When I initiate the call from kphoneB, the whole thing reverses and it becomes kphoneB---serB----kphoneA. I am using ser-0.8.12 (cannot use cvs version at the moment due to some code changed we have done) and mediaproxy-1.0.1. Anyone got any clue where I am wrong? ser.cfg file attached. Any help will be greately appreciated. Regards, Dhiraj ---------------------- ser.cfg ---------------------------------- ------------------- # # $Id$ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes 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=2 fifo="/tmp/ser_fifo" sip_warning=yes # ------------------ module loading ---------------------------------- # set the proper path to your modules modules # if you did "make install" it must be, for example # loadmodule "/usr/local/lib/ser/modules/sl.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/textops.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 to use SQL database #loadmodule "../sip_router/modules/mysql/mysql.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # modparam("usrloc", "db_mode", 2) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line # modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # -- auth params -- # -- mediaproxy params -- modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/ser/sip-asymmetrics-clients") modparam("mediaproxy", "rtp_asymmetrics", "/etc/ser/rtp-asymmetrics-clients") modparam("mediaproxy", "natping_interval", 20) modparam("domain", "db_url", "/usr/local/ser/domaintables") modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # ------------------------- 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; }; record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; if (client_nat_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { fix_contact(); force_rport(); }; }; if (uri==myself) { if (method=="REGISTER") { log("REGISTER"); save("location"); break; }; lookup("location"); 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; }; }; if (method=="INVITE") { use_media_proxy(); t_on_reply("1"); }; if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status=~"(183)|2[0-9][0-9]") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; if (status=~"[3-4]0[0-9]") { end_media_session(); break; }; }

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[Serusers] Proxy Configuration
by Ing. Francesco Russo - NoCable S.p.A. 08 Jul '04

08 Jul '04

Hi all, i have configured SER to manage internal SIP calls between users with same realm (nocable.it). To allow users to do normal calls to fixed phones in Italy, i have to configure the SER in proxy mode to route the calls to the PSTN gateway. How can i do it? Thanx a lot Francesco Russo

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[Serusers] Grandstream BT 100 firmware (1.0.5.3=
by Arnd Vehling 08 Jul '04

08 Jul '04

Hi all, i know this a bit OT on this list but nevertheless: Has anyone experience with the new "1.0.5.3" firmware release for the Grandstream BT Line fone types? We have a kind of "rapid registration" problem with a certain firmware release and so far the actuall firmware doesnt seem to have this bug. As i am usually using my trusty sipura adapter and not a GS fone i wanted to ask if someone has made any bad experiences with the current firmware release before upgrading a batch of those thingies to the current release. regards, :: arnd ::

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[Serusers] Strange behavior of Radius-Authentification with Usernames
by Kai Militzer 08 Jul '04

08 Jul '04

Hi list! I just came across something very strange when using the radius-modules and wonder if it is a wanted feature, a bug or simply me being stupid (which I guess will be the case). The thing is the following. My ser.cfg has the following in it when an UA registers: if (method=="REGISTER") { if (!radius_proxy_authorize("XXX.XXX.XXX.XXX")) { proxy_challenge("XXX.XXX.XXX.XXX", "0"); break; }; log(1,"Registered"); save("location"); break; }; This works fine, means the user get's registered, if it is known to Radius and not registered in the opposite case. Now to the strange thing. In most UAs you can enter different user-parts of the URI and Authentication-Users. I used kphone for this test and entered a valid username as authentication username and some random number (or word, that doesn't matter) as "User part of SIP URL". What happens then is, that the user can register and gets a URI different from the authenticated username. With this behavior every user would be able to "hijack" connections from other user. How can I tell SER to not allow this? Has it something to do with the SIP-Rpid argument in Radius? Ser seems to ignore it. Any hints, or RTFMs to get me looking in the right direction to solve this problem would be very kind. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-11 km(a)westend.com D-52064 Aachen Fax 0241/911879

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[Serusers] SERWEB configuration problem
by Ing. Francesco Russo - NoCable s.p.a. 08 Jul '04

08 Jul '04

Dear Friends, i have installed SER correctly, it works well and registers users present in mysql database. After SER installation i have installed SERWEB following step by step Dan Austin's HOW-TO. I have encountered this problems: 1) The web interface don't displays correclty...i don't see the background images, but the buttons are visible. 2) When a user logs in, on the missed_calls tab, i see the message "error in SQL query, line: 77" 3) When i click on email confirmation link....i see on the web page the message "400 Table "aliases" not in memory, please run save("aliases") or lookup("aliases") in the confirmation script first" Can you help me to solve this problem, please? Thanks all in advance Francesco Russo

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[Serusers] Fwd: problem wirh SIMPLE2Jabber
by Jiri Kuthan 08 Jul '04

08 Jul '04

>X-From_: mailman-bounces(a)mail.iptel.org Thu Jul 8 02:52:28 2004 >X-Original-To: list-admins(a)iptel.org >Delivered-To: jiri(a)fox.iptel.org >X-Original-To: serusers-owner(a)fox.iptel.org >Delivered-To: serusers-owner(a)fox.iptel.org >Date: Thu, 8 Jul 2004 09:00:55 +0800 >From: "=?gb2312?B?1cXJq9Xk?=" <zhangsezhen(a)sict.ac.cn> >To: "serusers-owner" <serusers-owner(a)lists.iptel.org> >Subject: problem wirh SIMPLE2Jabber >X-mailer: Foxmail 5.0 [cn] >X-Spam-Checker-Version: SpamAssassin 2.63 (2004-01-11) on fox >X-Spam-Level: >X-Spam-Status: No, hits=0.1 required=5.0 tests=BAYES_30,MIME_BASE64_TEXT > autolearn=no version=2.63 >Sender: mailman-bounces(a)mail.iptel.org > >serusers-owner£¬ÄúºÃ£¡ > After I installed jabber modules,sip client <-->msn,jabber client<-->psi.Ican send messgae from sip to jabber,but I cann't send message from jabber to sip ,when I capture message with ethereal,I cann't f receive a reply from jabber server:<?xml version='1.0'?> <stream:stream xmlns:stream='http://etherx.jabber.org/streams' id='some-random-id' xmlns='jabber:client' from='localhost'> ,what should I do? please help me! > > > >¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡zhangsezhen(a)sict.ac.cn >¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡2004-07-08 -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Initiate a call from SER
by GR S 08 Jul '04

08 Jul '04

Hello, --- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Hi, <snip> > Summarizing, only softphones and end-point applications will initiate > requests, that will be forwarded by servers within the "backbone" to the > appropriate end-point. > > I hope it is more clear now, Very clear! Thanks a lot for your reply... > Samuel. Regards, ===== Girish Gopinath <gr_sh2003(a)yahoo.com> __________________________________ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail

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Re: [Serusers] Initiate a call from SER
by Samuel Osorio Calvo 08 Jul '04

08 Jul '04

Hi, SIP's main entity are User Agents (UAs), which can be devided into UAC (User Agent Client) and UAS (User Agent Server), depending on the *transaction* they are involved into. UAC are transaction-level entities initiating requests and accepting responses. On the other side, UAS are transaction-level entities accepting requests and issuing responses. Thus, a normal SIP softphone instantiates (depending on the implementation may differ the exact way) a UAC whenever sending an INVITE and the *same* UAC waits for the response (usually an OK). If acting as receiving point, it will create a UAS instantiation as soon as it receives an incoming INVITE and will issue the appropriate answer. The point is that a server is mainly a UAS waiting for incoming requests BUT it will need to instantiate a UAC to send requests (just as forwarding INVITES, ACK, CANCEL,etc...) only after receiving incoming requests. By itself, a server MUST NOT send a request except to forward incoming ones. Next diagram (I expect it arrives with the right alignment) tries to explain the transaction-level entities involved in a generic SIP transaction. (INVITE+OK). I hope it will be useful to clarify how SIP works. softphone proxy server softphone UAC----------------------------------------->UAS + UAC----------------------------------------->UAS Thus a SIP application starting a request (initiating a transaction) will create a UAC transaction and will expect a UAS to receive and process it. I expect not to confuse you with the words UAS and UAC with the usual meaning of UA (=softphone). Summarizing, only softphones and end-point applications will initiate requests, that will be forwarded by servers within the "backbone" to the appropriate end-point. I hope it is more clear now, Samuel. P.S. This is the normal SIP behaviour........I'm afraid some situations might be different (NOTIFY requests). Unclassified >>> GR S <gr_sh2003(a)yahoo.com> 07/08/04 09:00AM >>> Hi, --- Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > So maybe you should start with your intention - what feature are you > trying to setup? > ser can create requests ( transaction = request+response(s) ) and you > can use this feature via the FIFO. Thanks for the reply. Sorry, I misnderstood the meaning of transaction. I was only thinking about SIP requests, not responses. But i am still confused about call initiating. Please correct me if i am wrong on this, the original poster was asking about initiating calls from SER and AFAIK proxies dont initiate calls. Can they initiate calls? If yes, on what situations? When you say SER can create requests, do you mean methods like CANCEL? > klaus TIA, ===== Girish Gopinath <gr_sh2003(a)yahoo.com> __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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sr-users July 2004