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[Users] more outbound proxies/openser & one register/*
by Peter Hudec 10 May '06

10 May '06

hi all, I haven't found so far any suitable solution for my problem. I want to have more outbound proxies and only one register. Outbound proxies should be also RTP proxy for clients /for all, or only behind NAT, it doesn't matter/ My idea is, that the REGISTER server see the IP of the outbound proxy, not the UA it self. Insipired by siporxd and milkfish project I wrote a patch for nathelper to rewrite the contacnat addres to IP of the proxy. If the proxy receives REGISTER, it save the contact and send new to to register with it IP. The signalinig is working good. But I have problem with RTP streams and the SDP part of the INVITE request. If you try to call between two proxies, the RTP goes PROXY1<->REGISTER<->PROXY2, instead of only between PROXY1<->PROXY2. Also if you made a call between 2 phone on the same proxy, the call passes through REGISTER ;( The example openser/nathelper configuration is working good, if the proxy is also register ;) I have atteached my own. Can anyone just a bit help me with configuration or give me a link to the another sip proxy project, which can satisfy my needs ? best regards Peter Hudec -- Things are easier than they appear; all you need is the right focus Leonardo Balliache # # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/openser/lib/openser/modules/sl.so" loadmodule "/usr/local/openser/lib/openser/modules/tm.so" loadmodule "/usr/local/openser/lib/openser/modules/rr.so" loadmodule "/usr/local/openser/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/openser/lib/openser/modules/usrloc.so" loadmodule "/usr/local/openser/lib/openser/modules/registrar.so" loadmodule "/usr/local/openser/lib/openser/modules/textops.so" loadmodule "/usr/local/openser/lib/openser/modules/nathelper.so" loadmodule "/usr/local/openser/lib/openser/modules/mysql.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("registrar", "sip_natping_flag", 7) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:22222") modparam("nathelper", "sipping_from", "sip:pinger@voip.swan.sk") modparam("nathelper", "sipping_method", "OPTIONS") # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return(0); }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return(0); }; if (method=="REGISTER") { if (uri==myself) { sl_send_reply("403", "This is PROXY, you can't register here!"); return(0); } else { if (nat_uac_test("3")) { setflag(6); fix_nated_register(); } setflag(7); force_rport(); save_noreply("location"); fix_nated_contact("62.152.225.27"); route(1); } return(0); } if (nat_uac_test("3")) { if (uri==myself) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); return(0); } } else { if (!lookup("location")) { fix_nated_contact("62.152.225.27"); if (method=="INVITE") { setflag(5); force_rtp_proxy(); } } else { fix_nated_contact("62.152.225.27"); if (method=="INVITE") { setflag(5); force_rtp_proxy(); } } force_rport(); t_on_reply("1"); } } else { if (uri==myself) { if (lookup("location")) { if (method=="INVITE") { setflag(5); force_rtp_proxy(); } force_rport(); t_on_reply("1"); } else { sl_send_reply("404", "Not Found"); return(0); } } } route(1); return(0); } route[1] { if (!t_relay()) { sl_reply_error(); }; return(0); } onreply_route[1] { if (nat_uac_test("3")) { fix_nated_contact("62.152.225.27"); if (isflagset(5) && status =~ "(183)|2[0-9][0-9]") { force_rtp_proxy(); } } else { if (isflagset(5) && status =~ "(183)|2[0-9][0-9]") { force_rtp_proxy(); } } force_rport(); }

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[Users] Problem with RPID
by Bogomolov Andrei 10 May '06

10 May '06

I use radius authorize reply to set AVP rpid value: 0(30405) DEBUG:auth_radius:radius_authorize_sterman: Success 0(30405) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(30405) DEBUG:auth_radius:extract_avp: string is <rpid:9284444> 0(30405) DEBUG:auth_radius:extract_avp: AVP name is <rpid> 0(30405) DEBUG:auth_radius:extract_avp: AVP val is <9284444> 0(30405) DEBUG:auth_radius:generate_avps: AVP 'rpid'/0='9284444'/0 has been added But when this caller is invating, openser writes to log: 0(30405) append_rpid_hf: No rpid AVP The question is: where are SIP-AVPs stored? Do I have to load auth_db in openser.cfg? Or avps could be stored in RAM without mysql database?

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[Users] asterisk and openser - CallerID
by Paul Cupis 10 May '06

10 May '06

Hi all, I'm having some trouble integrating openser with asterisk. I am able to make/receive calls PSTN<>asterisk<>OpenSER<>SIP-client fine, and the SIP client received the PSTN callers CallerID, but I cannot get the SIP-originated call to present a useful CLI to the PSTN recipient. FWIW, I have been able to set the CallerID manually from AGI scripts and the like, and it works from IAX<>asterisk<>PSTN. In openser.cfg I have: # Append Remote-Party-ID header field xlog( "L_INFO", "appending rpid" ); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); and I can see the RPID header in the SIP traffic. In sip.conf I have: [openser] type=friend context=from-openser host=<hidden> insecure=very trustrpid=yes amaflags=billing We are using asterisk 1.0.7 atm and openser 1.0.1 with MySQL backend. I've tried setting the rpid field in openser.subscriber to various settings but it is not making it to the PSTN recipient. Any ideas/gotchas with this sort of setup, or any further information required?

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[Users] Overwrite "From:" HF
by Bogomolov Andrei 10 May '06

10 May '06

Is it possible to overwrite "From:" HF on Invitation depending on RPID AVP? Our VOIP gateway does not support Remote-Party-ID header field.

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[Serusers] help me in serweb
by ravi reddy 10 May '06

10 May '06

Hello All, I have been trying to run this serweb since 4 days but i am getting different errors and i follwed so many e-mails from our community but still it doesnt work Iam using fedora 4 and ser 0.9.6 and i want to install serweb as followed by the docs but iam getting some errors like favicon.ico file missing some thing and i keep changing all the settings and at last now iam totally confused i installed my serweb in directory /var/www/ and in serweb there is one html directory and i moved it to the directory /var/www/html as told in the docs sip.aarnet.edu But when i typed this url:http://192.168.0.76/serweb Iam getting all list of directories and when i typed http://192.168.0.76/serweb/admin/index.php no message nothing a white blank page bullshit i dont understand what is happening here i changed my settings in /etc/httpd.conf ,and /etc/php.ini but,no use can any body please help me to get out of this work and to move forward thanks in advance :-) regards, Ravi.

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[Users] Overwrite "From:" header
by Bogomolov Andrei 10 May '06

10 May '06

Is it possible to overwrite "From:" header on Invitation depending on RPID AVP?

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[Serusers] SER Proxy not authenticating
by Stephen Helfen 10 May '06

10 May '06

I have several sip proxies running ser in the same network. The most recent proxy I have brought up is not authenticating requests, but rather forwarding them off to another proxy in the nextwork. The new proxy is essentially doing nothing more than passing messages back and forth between the UA and the other proxy when it should be able to handle the call itself. Any idea what I could be missing here?

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RE: [Serusers] ser.cfg children=
by Gould, Aaron 09 May '06

09 May '06

thanks evan. could i do this ...? 'listen=tls:eth0' thanks aaron ________________________________ From: Evan Borgström [mailto:evan.borgstrom@ca.mci.com] Sent: Tue 5/9/2006 12:51 PM To: Gould, Aaron Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] ser.cfg children= There are plenty of legitimate uses for having SER listen on a loopback address. I have one server that runs two instances of SER, one for SIP messaging and the other as a SIMPLE<->XMPP (jabber) gateway. They communicate over the loopback interface. To turn off different interfaces just modify your listen statements. For instance if you only want things to listen on eth0 add 'listen=eth0' to your global statements. If you want just UDP on eth0 use 'listen=udp:eth0'. FWIW, a good rule of thumb is to run services blacklist centric (ie. deny everything by default and then allow only the things you want) so the more specific your listen statements are the easier it is to anticipate the traffic you will receive. -Evan Gould, Aaron wrote: > why is there 4 of each? is there a legitimate need for the loopbacks? also, may i turn off the 4 udp and leave only the 4 tcp? if so how please. and furthermore, i beleive tls is security over tcp , if so, how may i turn on tls ? > > thanks > aaron > > > ________________________________ > > From: serusers-bounces(a)lists.iptel.org on behalf of Andrey Kouprianov > Sent: Tue 5/9/2006 3:36 AM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers] ser.cfg children= > > > > There was a privious post on this exactly like yours. Anyway... > > "children=4" mean that there will be 4 UDP, 4 TCP and 4 loopback > listenters on your SER, when you start it (you can check with "serctl > ps" to verify). > > On 5/9/06, Giuseppe Parlato <gparlato(a)tnet.it> wrote: >> Hi >> in ser.cfg what does #children=4 means.. I've been looking in many >> configurations file but there is no explenation >> >> Giuseppe >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Users] ACK is sent before the re-INVITE to OpenSER, but
by Gene Willingham 09 May '06

09 May '06

I too am having this problem. Are the reinvites being generated by Asterisk? I have a test environment where making adding two lines of code to asterisk will help the issue. My only concern is whether the code change creates a performance issue. I tried asking the question the asterisk-dev list, but did not get a response. My solution was to insert a usleep(20000) between the ack and the reinvite. This delays the reinvite by 20 ms. The concern I had was is Asterisk fully threaded or does the 20 ms delay delay all processing of sip requests. Gene

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[Users] openser on host without public IP
by j.obi＠troja.net 09 May '06

09 May '06

Hi Is it possible to use openser on a host without a public IP address? I have a router by my internet provider for the SDSL connection and behind that a wrt54g with openwrt on it. That works as Firewall and I want to use it as sip proxy now too. My Question is now, are there any possibilities to use openser with stun or something similiar to work around the problem with the IP address? Bye Jens

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