sr-users April 2004

sr-users@lists.kamailio.org

138 participants
266 discussions

[Serusers] Cisco ATA 186 adapter Firmware (SIP)

by Ashok Patel

Hi SER folks. I want to upgrade firmware on my cisco ATA 186 adapter from current version SIP v3.0 to v3.1 (or any latest SIP version available as of today). Does anybody know which web site provides ATA cisco 186 adapter firmware updrade version 3.1 (SIP). Currently I have SIP version 3.0 and I want to upgdare it to 3.1 or whatever is most recent version available. Please provide me some information about firmware version 3.1. at my e-mail address: ashokpatel(a)yahoo.com . Thanks in advance. -Ashok Patel ashokpatel(a)yahoo.com __________________________________ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html

20 years, 7 months

[Serusers] how to found SER server works like two local exchanges

by wangji

Hi all, The SIP phones on the Ser server have two different define: Exchange 1: number is like 61691xxx as caller: when it calls to PSTN, dialing 0+PSTN number when it calls to 6169xxxx(expect 61691xxx), dialing 0+6169xxxx when it calls to 61691xxx, dialing only 1xxx as callee: wher PSTN and 6169xxxx(expect 61691xxx) calls, dialing 61691xxx when 61691xxx calls, dialing 1xxx Exchange 2: number is like 6169xxxx(expect 61691xxx) as caller: call callee's number as callee: wher PSTN and 6169xxxx(expect 61691xxx) calls, dialing 6169xxxx when 61691xxx calls, dialing 0+6169xxxx Using prefix() and rewriteuser() to make call mabye cause call-leg can not find. How to do that? Br, Wangji

20 years, 7 months

[Serusers] LoadBalancer Script/Module

by Rose, John

Hi, Does anyone have some loadbalancer script/module code that I could take a look at? I know that you can load balance at the DNS level but I need to do it at the SER proxy level - as in: 1) INVITE arrives 2) route call to minimum call-count location 3) increment call-count data entry for that location 4) route call to that location 5) call progresses 6) BYE arrives 7) decrement call-count for location Sounds simple but all the intricacies and peculiarities of SIP/VoIP could cause the call-counter to get fubarred. John

20 years, 7 months

[Serusers] New bugfix version of radiusclient library (0.4.2)

by Maxim Sobolev

Hi, I've just released the 0.4.2 version of radiusclient library. Main changes since 0.4.1 include: - Some changes from Debian have been merged in (suggested by Sergei Golod <rover(a)tob.ru>): + IPv6 attributes added into generic dictionary and radiusclient.h; + fixed a bug preventing from using more than one Radius server in fall-back scenario. - Restored historoc behaviour when in the absence of explicit bindaddr option in the config file operating system is trusted to assign source IP address to outgoing UDP packets according to its routing table. Previously in 0.4.x series address rerutned by gethostbyhane(hostname()) was forced as source IP address for all outgoing UDP packets. The same behaviour can be ontained by specifying `bindaddr: *' (Problem reported by Juha Heinanen <jh(a)tutpro.com>, patch by sobomax). - Added new API function: rc_avpair_log, which allows to create pretty-print representation of full radius request or response (sobomax). - Extended dictionary parser so that now it is possible to use comments at the end of the configuration lines (Jan Janak <jan(a)iptel.org>). The new version is 100% API/ABI compatible with the old one. It can be downloaded at http://developer.berlios.de/project/showfiles.php?group_id=1208. As usually bug reports and suggestions are highly appreciated. Enjoy! -Maxim

20 years, 7 months

[Serusers] number reader

by R

Hi, I have ser 0.8.12 with sems 2004/01/04 on Fedora core 1. The number reader only reads the calling number, not the caller's number. Anyone had this problem before? Thanks, Richard __________________________________ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html

20 years, 7 months

[Serusers] Check From-User against To-User

by Andreas Granig

Hi, I was wondering if there is a simple possibility inside ser.cfg to check if someone is trying to call herself. Or is it only possible with exec_msg where I could match SIP_HF_FROM against SIP_HF_TO or something like this? I'd need it for the vm-module to determine if someone is trying to reach his own voice box. Regards, Andy

20 years, 7 months

[Serusers] rtp proxy problem

by santhosh

Hi , I am testing two cisco ATA behind NAT(using rtpproxy). Sometimes the call goes perfect but suddenly it stops.Sometimes it rings but the voice does't pass through. The logs show error connecting to the rtpproxy. I'm using RTPPROXY (1.19.2.10) and ser ( 0.8.12-tcp_nonb ). My configuration file is loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) route{ # compulsory processing of Route header fields and adding RR loose_route(); # ATA's are symmetric but don't advertise it -- force use of rport force_rport(); fix_nated_contact(); /* registration (uses rewritten contacts) */ if (method=="REGISTER") { save("location"); break; }; if (method=="INVITE") { record_route(); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); if (method == "INVITE" || method == "CANCEL") { if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; /* set up reply processing and forward statefuly */ t_relay(); } # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { if (status=~"2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); }; } Pls help me out ! thanks in advance santhosh

20 years, 7 months

[Serusers] serweb doesn't send mail

by Teppo Lunnas

Hello, I have installed latest ser and serweb. Now when I try to register a new user from serweb's subscribe page by filling the form I get the following message: " Sorry, there was an error when sending mail. Please try again later." So what went wrong? Does the serweb have a built in e-mail client and how does it know the address of the smtp-server?

20 years, 7 months

[Serusers] sl_send_reply warning

by Harold Workman

Hi, Im connecting to a cisco gateway, and the call looks fine, until the SER has to send an ACK. to bring the status of the call to active. When i look on my messages in the log, i am getting a message Warning: sl_sed_reply: I wont send a reply for ACK!!! what does this mean? ----------------------- Harold Workman CCNA, CCNP Cytel Communications hworkman(a)cytelcom.com Ph. 281-449-4000 x3098

20 years, 7 months

[Serusers] username numeric routing

by Harold Workman

Just a quick question....I am using numerics as the username (ex. 111000001) of my hosts. I am able to make pstn calls through my pstn gateway. My question is how do i configure the routing request lock to allow ip to ip calls between two hosts. if i dial 111000002 it routes the call to the pstn instead of to the other host. here is my config so far. # ------------------------- request 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 > max_len ) { 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # is it a PSTN destination? (is username nummerical and does not begin with 8?) if (uri=~"^sip:[0-79][0-9]*@") { # ... forward to gateways then; # check first to which PSTN destination the requests goes; # if it is US (prefix "1"), use the gateway 64.72.107.2... if (uri=~"^sip:1") { forward(64.72.107.2, 5060); } if (uri=~"^sip:011") { forward(64.72.107.2, 5060); } else { if (!lookup("location")) { sl_send_reply("404", "User Not Found"); log("SER: Dest User Not in location table.\n"); break; }; if (!t_relay()) { sl_reply_error(); }; 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("sipua.cytelcom.com", "subscriber")) { www_challenge("sipua.cytelcom.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } -- Harold Workman CCNA, CCNP Cytel Communications hworkman(a)cytelcom.com Ph. 281-449-4000 x3098 ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

20 years, 7 months

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sr-users April 2004