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37537 discussions

[Serusers] (no subject)
by info＠beeplove.com 22 Apr '05

22 Apr '05

Hello List: Are you guys taking any authentication measure to allow access on MediaProxy? Thanks, Mohammad -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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FW: [Serusers] calls between UA´s behind same NAT using nathelper/rtpproxy
by jimmy 22 Apr '05

22 Apr '05

-----Original Message----- From: jimmy [mailto:jimmy_huang@uni.com.tw] Sent: Tuesday, March 15, 2005 10:01 AM To: 'rrisco(a)millicom.net.pe' Subject: RE: [Serusers] calls between UA´s behind same NAT using nathelper/rtpproxy Hi Rafael: It seems has two way to meet your requirement (1)use external script http://lists.iptel.org/pipermail/serusers/2005-February/015477.html (2)use avpops module http://www.voice-system.ro/docs/avpops/ Hope it's useful ... Jimmy -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Rafael J. Risco G.V. Sent: Tuesday, March 15, 2005 2:22 AM To: serusers(a)lists.iptel.org Subject: [Serusers] calls between UA´s behind same NAT using nathelper/rtpproxy Hi I would like to know if its possible to comunicate SIP Clients betwen them when they are behind the same nat, I am using Nathelper/RTPProxy... thanks Rafael -- rrgv _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] Bug # 2925
by Jiri Kuthan 22 Apr '05

22 Apr '05

I received questions for more detailed explanation, so here it is: branch value serves as transaction id, which (among others) differentiates retransmission of the same request from new requests. From RFC3261's standpoint, ACK for a 200 (as opposed to a [3-6]xx ACK) is a separate transaction, i.e., a different branch value is adequate. In case people do keep asking: it is really so, and receivers MUST be able to deal with ACK branch different from branch in INVITE. Let's think of a proxy that does not record-route and introduced branch 1234 to "its" Via. ACK coming later directly to UAS includes UAC's branch which is different from proxy's branch. It may or may not be the same as UAC's INVITE branch but it is different from what UAS sees. -jiri At 12:44 AM 4/20/2005, Darryl H. Thomas wrote: >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Tuesday, April 19, 2005 2:23 PM >To: Darryl H. Thomas; serusers(a)lists.iptel.org >Subject: Re: [Serusers] Bug # 2925 > >The bug report is invalid, ignore it. branch=0 in ACK to a 200 >is perfectly valid according to RFC3261. Your termination provider >must be using a buggy devices. > >-jiri > >At 07:04 PM 4/19/2005, Darryl H. Thomas wrote: >>Hello everyone, >> >>I was wondering whether anyone knew what was going on with the >following >>bug: >>http://developer.berlios.de/bugs/?func=detailbug&bug_id=2925&group_id=4 >80 >> >>For those not wanting to follow the links, this is a bug submitted by >>someone back in December wherein ser puts 'branch=0' in the Via header >it >>creates for an ACK to a 200 OK Invite response. >> >>I haven't seen any activity related to this, and it's affecting >>interoperability with one of our termination providers. >> >>0.9.0 users, have you seen the same problem?? >> >>Cheers, >>Darryl >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >-- >Jiri Kuthan http://iptel.org/~jiri/ -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] ser 0.9.0+rpid
by raid＠gmx.net 22 Apr '05

22 Apr '05

hi list please can someone explain to use ser with rpid where is the best place to insert "append_rpid_hf("<", ">;party=calling;id-type=subscriber;screen=yes;privacy=off");" thanks regards raid -- +++ NEU: GMX DSL_Flatrate! Schon ab 14,99 EUR/Monat! +++ GMX Garantie: Surfen ohne Tempo-Limit! http://www.gmx.net/de/go/dsl

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[Serusers] PSTN => SER => PSTN
by Mik Cheez 22 Apr '05

22 Apr '05

I'm having an issue where my OUT PSTN is not getting a signal that a call has ended. My setup is as follows: IN OUT PSTN ==> SER ==> PSTN If I place a call and the person hangs up before the callee answers, the callee's phone just keeps ringing. I see a CANCEL event on the SER box, but nothing on the OUT PSTN. Here's a sample of my ser.cfg script: ------------------------------------ debug=3 fork=yes log_stderror=yes sip_warning=no check_via=no dns=no rev_dns=no port=5060 #children=4 fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/exec.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/xlog.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ 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; }; if (method!="REGISTER") { record_route(); }; xlog("L_INFO", "#INFO# %rm\n"); if (loose_route()) { t_relay(); break; }; setflag(1); if (method=="CANCEL"){ sl_send_reply("487", "Terminated"); break; }; if (method=="INVITE") { if (uri=~"sip:[0-9]+@.*") { xlog("L_INFO", "Original URI: %ru \n"); exec_dset("insert script here"); xlog("L_INFO", "Relay URI = %ru\n"); if (!t_relay()) { sl_reply_error(); break; }; }; }; }

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[Serusers] gw based on username
by Iqbal 22 Apr '05

22 Apr '05

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[Serusers] gw based of username
by Iqbal Gandham 22 Apr '05

22 Apr '05

Hi Currently I am using lcr to choose my gateway based upon prefix, now I wanted to choose a gateway based on username. I was thinking that if a user made a call to say 001xyz I could detect its username, see if in usr_prefences (using avp) it had a variable set eg:add_prefix, if so add a prefix of 2 to the number. Then in lcr add a false prefix of 2001 to the table with the new gateway, and at the gateway remove the prefix, before making the call, this way I dont have to ask users to dial any special prefixes. Any better ideas -- Regards Iqbal Gandham, CEO IPClouds Ltd (http://www.ipclouds.co.uk) Baylis House ,Stoke Poges Lane Slough,Berkshire SL1 3PB V: iqbal(a)sip.ipclouds.co.uk ( call me free !!) M: +44 7866318220 F: +44 (0) 870 7474784 W: http://www.ipclouds.co.uk ------------------------- b5 18 b5 07 1c cd 15 41 ba 23 ad 94 38 30 b8 df -------------------------

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[Serusers] AS 5350 + Ser Problem
by Razvan Nemesiu 22 Apr '05

22 Apr '05

Hi all, I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming pots dial-peer from PSTN (let's say that it matches 333XXX numbers).Then I have two outgoing voip dial-peers (both dial-peers match 333XXX numbers and are the outgoing routes for these numbers).One of these two outgoing dial-peers is set with a higher preference than the other in order to be the first choice for sending the calls.One of the dial-peers has session target one of the SER and the other has the other SER as session target.So, the main idea is that I want to do fallback between these two dial-peers: if the connection with the first SER (that is with higher preference) is down or there are network problems I want my CISCO to choose the other route to my second SER (and I want it to do that after a specified time).The question is how cand I do that ? And how can I set this timer ?(let's say 10 seconds -> after 10 seconds to choose the other route).What commands do I have to set on the dial-peers and what commands need to be set in the global configuration (timers,etc...)? Thank you very much and I appreciate any help.

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[Serusers] Help with authorization
by Steve Blair 22 Apr '05

22 Apr '05

I have a working 0.9.1 config to which I would like add server side features such as call forward all (cfwdall). I have a pretty good idea how to handle cfwdall using avp_ops however I'm stuck on the authentication. If a subscriber has local calling permissions (acl=local) and cfwdall their phone to a long distance number I need to respond with an informative response. In the INVITE processing in my config I have statements such as: if (is_user_in("credentials", "ld")) { setflag(11); }; These checks fail with the following errors: Apr 21 18:31:53 ser[498]: [SER]: AVP: Checking From gateway caller Apr 21 18:31:53 ser[498]: check_username(): No authorized credentials found (error in scripts) Apr 21 18:31:53 ser[498]: check_username(): Call {www,proxy}_authorize before calling check_* function ! Apr 21 18:31:53 ser[498]: [SER]: Flag for UMVM redirect successful. Apr 21 18:31:53 ser[498]: [SER]: AVP: Checking credentials Apr 21 18:31:53 ser[498]: is_user_in(): No authorized credentials found (error in scripts) I thought adding proxy_authorize("", "subscriber")), check_to and check_from calls prior to the is_user_in check would address these errors but that hasn't worked either. If I want to set a flag if the caller is an authorized subscriber, the callee is an authorized subscriber and then use "is_user_in" to determine if the called party has a particular credential what am I missing? Thanks,Steve

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[Serusers] serweb path
by Llanos Serna García-Conde 22 Apr '05

22 Apr '05

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