sr-users April 2008

sr-users@lists.kamailio.org

141 participants
187 discussions

[OpenSER-Users] How to modify the "Referred-by" field inside a REFER

by Amelie CAMELIERE

Hi everybody, Is it possible to modify the "Referred-by" field inside a REFER SIP message? In fact I want to modify the name displayed by the phone when I use a REFER. Thanks for your help Amélie

16 years, 8 months

[OpenSER-Users] Openser with RTPProxy in a different scenario

by morton.olson

Hi all! I have a problem with Openser an RTPProxy. I am using nathelper module to establish a statefull connection with RTPProxy in between the both clients. I player around with it now for a while but I cant get it to work. All the examples in the internet a for clients behind NAT, etc... Maybe one of you might help me. I am using Openser 1.3.0 with RTPProxy 20040107 on Debian Linux "Etch". My scenario looks like this: UA(public ip) ----- Firewall (public IP; 192.168.0.1/24) ---- Openser+RTPProxy (192.168.0.5/24) ----- UA (192.168.0.23/24) The Firewall forward Port 5060 to the Openser which is listening on 5060 as well. Also Ports from 60000-61000 are forwarded to the Openser aswell. (RTPProxy was compile with these Ports) I have no access to the firewall and it does not manipulate SIP Messages. Is there a chance to get this to work? If so, I willl post my configuration file to see what I did wrong. Thanks a lot. Greetz Martin -- View this message in context: http://www.nabble.com/Openser-with-RTPProxy-in-a-different-scenario-tp16349… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

16 years, 8 months

[Serusers] Asterisk + OpenSER integration successful

by satish patel

Dear ser guys.. I have done asterisk with openser integration after spending 2 and 3 days...not time to get nice cold beer... I am user openser front of Asterisk. All sip client register on OpenSER server and call to local extension without Asterisk. caz Asterisk is a media server which are useing only for PSTN calling. When any sip client dialing outside number means mobile number or any outside number Openser forward call to Asterisk and asterisk forward it on PSTN use the E1 or FXO line. ( In my case it is E1 ). OpenSER only user for SIP users or SIP proxy server it can only handle sip request. Asterisk is Media server which can handle IVR , confrance , VoiceMail, and Meetmee or PSTN application. when call come down from PSTN Asterisk forward it on OpenSER as per dialplan configuration. It can handle 1000 of call and you can use it on large production. [sip_client]------------->[Openser]--------->[Asterisk]----[PSTN] My Openser configuration file. #cat ser.cfg debug=6 fork=yes log_stderror=yes listen=<openser_ipaddress> # INSERT YOUR IP ADDRESS HERE port=5060 children=4 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) 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" loadmodule "/usr/local/lib/ser/modules/textops.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" # ----------------- 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"); break; }; if (msg:len >= 2048 ) { 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 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"); 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # requests for Media server if(method=="INVITE") { if (uri =~ "sip:0[1-9]@*"){ route(3); 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); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3]{ if (uri =~ "sip:0[1-9]@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("<asterisk_ip>:5060"); route(1); break; } } #---------END------------# #Asterisk extension.conf [globals] SERADDRESS=203.124.16.109 [outgoing] exten => 5500,1,Playback(hello-world) exten => 5000,1,Answer() exten => 5000,2,Dial(SIP/5000) exten => 5000,3,Hangup() ; XXXX use to forward call on openser extensions. exten => _XXXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Explore your hobbies and interests. 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16 years, 8 months

[Serusers] How to set a global variable in ser.cfg and use it in following route() functions?

by lin9k＠yahoo.com

Hi, Is ser.cfg allow such feature? I'm thinking to use it to hold some variable that can be used for the life of ser.cfg. Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

16 years, 8 months

[Serusers] calling rewriteuser and rewriteuri from a module?

by lin9k＠yahoo.com

Hi, I want to provide a module of our own. How do I call rewriteuser and rewriteuri functions from a module? Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

16 years, 8 months

[OpenSER-Users] 1.3.x package?? forLinux??

by Rey Mulero

Hello, I am new to OpenSER. I wanted to know if there was a package already compiled to support a P4 Intel machine. I was looking in the website but it really does not spell it out. I see file names openser-1.3.0-tls_linux_i386.tar.gz and openser-1.3.0-notls_linux_i386.tar.gz I am wondering what the difference is? What does tls or notls stand for? Your help is appreciated. Thank you Rey ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

16 years, 8 months

[Serusers] Asterisk + SER call forwading

by satish patel

Dear all I want to achive forwading call from SER to Asterisk system i have read about it on voip-info real-time integration of SER & asterisk but its base on mysql i dont have mysql database server i just want to use asterisk as PSTN media means all SIP register on SER and "0XXXXXXXXXXX" number goes thorugh asterisk Ser and asterisk runing on defferent machine. I have put this code in my ser.cfg.sample file.. [globals] SERADDRESS=XXX.XXX.XXX.XXX:5060 [context] exten => <yourexten>,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) In ser.cfg: if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("XXX.XXX.XXX.XXX:5060"); t_relay(); break; } } But it is not working u got error "User not Found" i dont know way this happending... $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

16 years, 8 months

[OpenSER-Users] re : Cert advisory and OpenSER V1.3.1

by Jignesh Gandhi

Hello, I recently upgraded from SER.8.14 to OpenSER 1.3.1 and had a quick question regarding the SIP CERT advisory suite of tests. Has OpenSER been tested against the SIP CERT advisory suite of test cases, that can be downloaded from http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/. This is the SIP CERT website http://www.cert.org/advisories/CA-2003-06.html thanks, Jignesh Gandhi Movius Interactive Corp

16 years, 8 months

[Serusers] SER for NAT client

by satish patel

Dear all I have a bit confusen about server and nat its stupid question but i want to clear it wil your suggestions.. I am going to impliment SER for my customers with domain base authentication. Now if my client behind the NAT then how my RTP make connection between two IP phone [SER] / \ / \ / \ [A] [B] In this case RTP between [A] to [B] right But when both [A] and [B] behind the same NAT router then what about RTP what is the path ??? What is the pass of RTP between [A] and [B] in this case [SER] | | | [NAT] / \ / \ [A] [B] $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Now you can chat without downloading messenger. Click here to know how.

16 years, 8 months

[Serusers] SER and ptime ?

by Mike Trest - Personal

Hi, I have registering phones (Linksys) that use g711 or g729. I have a media question in a situation when I attempt to do pure-proxy without media gateway or proxy. Because these user are always messing with their adapter settings, I have variable call setup when I forward certain destinaitons because of ptime miss-match between Linksys and destination. Is it possible to accpet / reject / revise the inbound call when the "ptime" is not specifically 20ms or 30ms (or any specific value) before t_relay on to destination? I see the ptime value in SDP but before I embark on journey trying to "manage" call setup based on that, I want to know if others are working with this issue. ..mike..

16 years, 8 months

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sr-users April 2008