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37527 discussions

[Serusers] Can SER do this?
by David Good 16 Feb '07

16 Feb '07

I have a problem that I *think* SER can help with, but looking through the docs and samples, I don't see anything quite like what I need to do. Here's my situation: I have an Asterisk box setup internally with a few PSTN trunks. We have another box in a colo that we're using to communicate to a SIP trunk from bandwidth.com. Currently we have Asterisk running there, too. What we want to do is: * Have all outbound calls from Asterisk go to the SIP trunk at the colo. We have this working now using Asterisk at the colo. * Have all incoming calls from the SIP trunk (it has several DID numbers) be routed to the internal Asterisk box just like the existing PSTN trunks. I haven't been able to get this part working using Asterisk (which is why I'm looking at SER now) There is NAT involved between the colo server and bandwidth.com. Is this possible?

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[Users] Open SER for SIP NNI
by konstantinos nikou 16 Feb '07

16 Feb '07

Dear all, I am trying to use the Open SER for SIP NNI. I would like all calls with the prefix 077 to be routed to another SIP server and add a prefix (+49). I am using the following code: if (uri=~"^sip:077[0-9]*@*") { strip(1); prefix("+49"); rewritehost("172.17.2.35"); # forward( 172.17.2.35, 5060 ); route(1); } Normally, i believe that when i run this code and i am sending an INVITE from my user to a number e.g. 0771234567, the SER server should send this invite to the IP 172.17.2.35. But i don't see this. The SER responsds directly with a 404 NOT FOUND message. Do you have any idea of how i could solve the problem? I am using Suse Linux and attached you can find my configuration file. Thank you all, K.N. ____________________________________________________________________________________ Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/

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[Users] Openser- parser error for openser.cfg
by K Biju 16 Feb '07

16 Feb '07

Hi All, I have installed openser version 1.1.1-notls(i386/linux). I am trying to configure the openser to route all calls to a gateway. For this I have added the following rules in openser.cfg route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; } record_route(); if (src_ip==1.1.1.1) { forward_tcp(4.4.4.4, 5060); } else { forward_udp(1.1.1.1, 5060); } } Now when I bring up the server I get the following error [root@localhost openser-1.1.1-notls]# /usr/local/sbin/openser -n 1 -ddddd -E -l tcp:47.100.105.195:5060 -l udp:47.100.105.195:5060 0(18381) read 3133282016 from /dev/urandom 0(18381) seeding PRNG with 9917133 0(18381) test random number 521145638 0(18381) parse error (27,46-50): parse error 0(18381) parse error (27,50-51): bad arguments 0(18381) parse error (30,28-29): parse error 0(18381) parse error (30,41-42): bad arguments ERROR: bad config file (4 errors) What is wrong with the syntax or am I missing any modules ? Thanks in advance -Biju ____________________________________________________________________________________ Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

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[Users] NAT Types
by raviprakash sunkara 16 Feb '07

16 Feb '07

Hello Users, Good Morning, What are Type of the NAT is used for OpenSER for production , OpenSER is inside the NAT (router/firewall ) 1) I tested with 2 SJ phones and 2 cisco ATA phones, UAC's are in Behind the NAT, UACs are in the Same N/w with one public_address, SJ phone show the NAT type is Symmetric NAT this case is Working Fine, For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT1 ] SJphone -------------------------------> OpenSER---------------------> ATA phone 2) When SJ phones or HardPhones are in Different network behind the NAT of the OpenSER server, here SJ phone shows the Port Restricted core NAT Here media is not is signaling at all , in this Case.... For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT3 ] SJphone -------------------------------> OpenSER---------------------> ATA phone SIP NAT Traversal support only Symmetric NAT , ? Then How to solve this issue ? Please Help ? -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

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[Users] NAT Types Isssues
by raviprakash sunkara 16 Feb '07

16 Feb '07

Hello Users, Good Morning, What are Type of the NAT is used for OpenSER for production , OpenSER is inside the NAT (router/firewall ) 1) I tested with 2 SJ phones and 2 cisco ATA phones, UAC's are in Behind the NAT, UACs are in the Same N/w with one public_address, SJ phone show the NAT type is Symmetric NAT in Display... this case is Working Fine, 2) When SJ phones or HardPhones are in Different network behind the NAT of the OpenSER server, here SJ phone shows the Port Restricted core NAT Here media is not is signaling at all , in this Case.... SIP NAT Traversal support only Symmetric NAT , ? Then How to solve this issue ? Please Help ? -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 Client and Parent company :- www.august-networks.com

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Re: [Users] help-me in 800
by Daniel-Constantin Mierla 16 Feb '07

16 Feb '07

Hello, the PIN is sent usually via IVR codes. OpenSER is just a signaling proxy, it doen't interpret media stream. To get auth via PIN, you need a media server in the midle, like Asterisk. Cheers, Daniel On 02/14/07 01:30, Jobson Andrade wrote: > > Hi All, > > > > I,m use openser 1.0.1 > > > > We imagined the following setting: > > > > 1- It Receive the connections of a pstn that sends all of the > requisition for my one openser > > > > 2- it authenticate the coming calls of that IP > > > > 3- It Receive the PIN, authenticate and direct the call for the routes > > > > The question is like do that verification of the PIN´s and send the > calls after that verification > > > > It plan is: > > > > It receives itself 0800 123 123 of an ONLY IP, through the > src_ip==xxx.xxx.xxx.xxx sends for a route > > > > I obtained to remove with the strip I number him 0800 123 123 and with > the prefix (""); send with the rewritehostport for a route > > > > I need that be attended, authenticated the PIN and receive of the side > "A" I number of fate and sent for the route of smaller cost > > > > How to made this???? > > > > Please help-me!! > > > > Thanks in advanced!! > > > > > > > > Jobson Andrade > > > > Projetos & Desenvolvimento > Obelisk - The Asterisk & VoIP Experts > > > > phone/fax: (11) 2164-4808 ext. 115 > cell Phone: (11) 8175-9916 / 8271-0480 > email: jandrade(a)obelisknet.com.br <mailto:jandrade@obelisknet.com.br> > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] Re: Message dispatching to local scripts - problem in modifying the RURI
by Dan-Cristian Bogos 15 Feb '07

15 Feb '07

Bogdan, thxs a loot. It is more clear now when I know how the message is handled. Cheers, Dan

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[Users] error editing openser.cfg
by Javier Allende Astigarraga 15 Feb '07

15 Feb '07

i follow this instrucctions and when i execute open ser appear one failure, this is the openser.cfg # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion 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 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/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/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", 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") # -- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); #exit; # break; # }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } thanks for all -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

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Re: [Users] Send Call using authentication
by Bogdan-Andrei Iancu 15 Feb '07

15 Feb '07

Hi Jobson, does your ITSP require digest authentication? if so, you better take a look at uac module. See: http://www.openser.org/docs/modules/1.2.x/uac.html regards, bogdan Jobson Andrade wrote: > > Hi All, > > I need of send the calls for my ITSP using verification. > > Ex.: > > User = user_itsp > > Passwd = pass_itsp > > SIP Server = itsp.com.br > > I my configuration I made of this way, more not work. > > ……… > > ##Obelisk > > if (uri=~"^sip:00[1-9][1-9][0-9]*@") { > > route(3); > > exit; > > } > > ……… > > route[3]{ > > #sending route to DDI > > strip(2); > > rewritehostport("itsp.com.br:5060"); > > rewriteuser("user_itsp"); > > rewriteuserpass("pass_itsp"); > > route(1); > > exit; > > } > > ……… > > How to made this in openser? > > I use the openser version 1.0.1 > > Thanks in advanced, > > Jobson Andrade > > Projetos & Desenvolvimento > Obelisk - The Asterisk & VoIP Experts > > phone/fax: (11) 2164-4808 ext. 115 > cell Phone: (11) 8175-9916 / 8271-0480 > email: jandrade(a)obelisknet.com.br <mailto:jandrade@obelisknet.com.br> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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Re: [Users] Message dispatching to local scripts - problem in modifying the RURI
by Bogdan-Andrei Iancu 15 Feb '07

15 Feb '07

Dan, the env. vars. are also available in 1.1.x : http://www.openser.org/docs/modules/1.1.x/exec.html#AEN33 the buffer that is sent to the script by exec_msg() is the original received buffer, so it does not contain any of your script changes. regards, bogdan Dan-Cristian Bogos wrote: > Hi Bogdan, > > Till 1.2.x will be stable, I have to use 1.1.0 in my production, > therefore I cannot use the new features from 1.2.x (even if I am keen > of doing this). > > About the external script, I am passing the full message to the > external script, and I have considered that this is after I have done > the ruri modification through avps. I have tried also by using strip() > and prefix() functions from the core, but got the same result: my > changes were not applied when echoing the message to the external > script. > > As I have wrote also to users mailing list, here is my routing flow: > > > if (avp_check("$avp(s:message_dst)","re/sip:00.*/g")) { > avp_subst("$avp(s:message_dst)/g", "/sip:00(.*)/sip:+\1/"); > xlog("I have changed the destination from sip:00 into sip:+ and avp > is $avp(s:message_dst)"); > }; > > if (!avp_pushto("$ru","$avp(s:message_dst)")) { > sl_send_reply("403", "Cannot set destination from AVP"); > exit; > }; > > exec_msg("/scripts/echo_message"); > t_newtran(); > t_reply("200", "OK"); > t_release(); > exit; > > Thxs in advance, > Dan > > On 2/15/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >> Hi Dan, >> >> how are you accessing the ruri from the external script? maybe you >> should use the env vars set by openser - see: >> http://www.openser.org/docs/modules/1.2.x/exec.html#AEN33 >> >> guess SIP_RURI is what you need. >> >> regards, >> bogdan >> >> Dan-Cristian Bogos wrote: >> > Guys, >> > >> > I've got a problem with MESSAGE processing. >> > I am trying to push an avp to replace $ru before dispatching the >> > message to a local script. I can log the modification of the ru >> > successful with xlog, but, as soon as I will send the message to the >> > local script, it looks like the changes are dropped and to the script >> > will be presented the original message. Any ideea why? Are the changes >> > not applied for messages not forwarded out, and dispached locally? >> > >> > In my routing I have: >> > >> > if (avp_check("$avp(s:message_dst)","re/sip:00.*/g")) { >> > avp_subst("$avp(s:message_dst)/g", "/sip:00(.*)/sip:+\1/"); >> > xlog("I have changed the destination from sip:00 into sip:+ and >> avp >> > is $avp(s:message_dst)"); >> > }; >> > >> > if (!avp_pushto("$ru","$avp(s:message_dst)")) { >> > sl_send_reply("403", "Cannot set destination from AVP"); >> > exit; >> > }; >> > >> > exec_msg("echo_message"); >> > t_newtran(); >> > t_reply("200", "OK"); >> > t_release(); >> > exit; >> > >> > I appreciate any kind of advice. >> > >> > Thxs, >> > Dan >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >> >

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