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37512 discussions

[Serusers] how to add mysql.so
by vijay tiwari 29 Sep '06

29 Sep '06

hi all i am not able to add the mysql in that locations /usr/local/lib/ser/modules thanks vijay --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.

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[Users] video rtp
by Ronald Wiplinger 29 Sep '06

29 Sep '06

Can anybody explain me this, please. How is the rtp stream with video? Which ports are usually used for that? Following thoughts I have how it could be: 1. video and audio are two indipendent different rtp streams 2. audio starts with the lower free udp numbers, video with the higher ones. Only that could give me an explanation, why audio is ok in our test, but no video, and only if we want to call to a NAT, but worked from it. Where can I get more info? bye Ronald

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[Serusers] Can I configure SER to send Radius Accounting packets?
by a.kolesnikov 29 Sep '06

29 Sep '06

Hello, everyone. Can I configure SER to send Radius Accounting packets? С уважением, Колесников Андрей Netris телефон: +7 495 950 55 25 ext. 1303 <http://www.netris.ru> http://www.netris.ru

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[Serusers] Busy Call Fwd..
by Natambu Obleton 29 Sep '06

29 Sep '06

I have my SER server setup to forward calls when busy and I am using polycomm 601 registered to SER. When the Polycomm is on the phone and another phone call its, they don't get a busy signal, but show up on line2 for answering. I think this is a stateful vs. stateless problem, but how can I set it up so that when the user is on the phone it rings busy? Really just limit the number of simultaneous calls. Thanks. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Serusers] Re: SER and SPA 942
by Alex Fler 29 Sep '06

29 Sep '06

Hello All, Sorry for stupid question, I have ser server configured, and it works fine for my needs, I never had register my UA's directly to SER. When I registered I SPA 942 phone directly to ser and put call on hold, my phone display the message "Trying to hold" for a few seconds, and after that call getting dropped. Does anybody know why? Thank you for help. Regards Alex Fler

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[Users] cdrtool/mediaproxy/openser
by Richard Bennett 29 Sep '06

29 Sep '06

Hi, Does anyone have an example openser.cfg file for using openser with Mediaproxy and CDRTool? I have Mediaproxy/openser working with the onsip getting started example, but this only passes calls that are behind nat to Mediaproxy. What is the best way to setup openser so that all calls pass through mediaproxy, and the corrected CDRs* can be written with accounting? * as mediaproxy disconnects zombie-calls, and corrects the accounting to the time the last rtp packets were detected. Thanks, Richard

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[Users] RTPProxy manual
by Jeremie Le Hen 29 Sep '06

29 Sep '06

Hi, sorry if this is a bit off-topic, but I am looking for a document as near as a manual page as possible. I remember to have found such a document a few weeks ago, but I can't find it any more. It basically explained what did the command-line options, what is the bridge mode, etc. Pointers are welcome. Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] mediaproxy working, but not if asterisk is involved
by Arne Van Theemsche 29 Sep '06

29 Sep '06

hi my users subscribe with openser, en asterisk is used as connectivity to pstn i am now installing a mediaproxy, for all users, so every call goes via a mediaproxy. I'm doing this as follows (relevant statements only) in route #I installed the t_on_reply here to be sure that every reply gets parsed, but normally in the INVITE section should be enough? t_on_reply("1"); if (method==INVITE) { use_media_proxy(); } onreply_route[1] { log(-3,"reply received"); if (!search("^Content-Length:[ ]*0")) { log(-3,"using mediaproxy"); use_media_proxy(); }; } the weird is, for all local users, this works fine, but as soon as asterisk is involved, the reply doesn't get triggered (not seeing the "reply received" either, only when disconnecting the call). The call get's established fine, asterisk is sending media to the mediaproxy, but the SDP towards the calling phone is not modified (since the onreply isn't triggered) am I missing something here? thanks Arne

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[Users] Cancel question/problem
by Andreas M. 29 Sep '06

29 Sep '06

Hello, i have following setup: Cisco AS5300 -- Openser --- Client Now if there is an incoming call from the CiscoGW to the client and is canceled, but the "cancel" message does not arrive at the client (because any reason), the phone doesn´t stop ringing. Cisco --- OSER --- Client Cancel -> <- Canceling Cancel -> thats it. It seems that no one is missing any OK or ACKS! It happens only sometimes, but who is responsible to handle this behavior ? The Cisco Gateway or the Proxy ? Using OSER 1.0.1 regards, Andreas M.

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[Users] tls. soket queues not flushed
by serega 29 Sep '06

29 Sep '06

Hi all. I have next problem while using openser with tls. System consists of sip statefull server (in next time, simply, server) and stateless sip gateway (gateway). Server used for connect to jabber server. Server contains rewrote jabber module logic and also use presence module. Gateway used to connect to other sip gateway (extgateway) via tls protocol. Gateway use 1.1 openser version and doesn’t contain changed code. When server together with gateway restarts and server in its database contains above 260 subscriptions (in watcherinfo table) i have error. Socket used by gateway to connect with extgateway contains in receive and sent queues a lot of data (above 50kByte on each sides). This data never flushed out. This happens because after restart sip server through gateway send notifications to extgateway. When I attached using gdb to process that send data, I saw that it was in infinity loop because tls library returns SSL_ERROR_WANT_WRITE and I think it’s correct because we have overfilled send queue. In this case I don’t interest why receive queue not empty (I think it happens because sending process have got block on socket). But I don’t understand why sent queue not flushed. I test this behavior using tcp – all was correct. Socket not closed by other side because this status can be unchanged above one day. Anybody can help me? Why this happens? I use openssl-0.9.8c and redhat os (Linux xdevel1 2.4.21-4.ELsmp #1 SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux). Computer have too Pentium 4 processors). Gateway.cfg (sip.qa.hbex.com – gateway address): # # $Id: router_qa.cfg,v 1.1 2006/08/02 18:14:34 ilya Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) log_facility=LOG_LOCAL0 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=20 fifo="/tmp/openser_fifo" server_signature=no fifo_db_url="mysql://root:@localhost/openser" listen = udp:sip.qa.hbex.com:5060 #incoming sip server address listen = tcp:sip.qa.hbex.com:5060 #tls address disable_tls = 0 listen = tls:sip.qa.hbex.com:5061 listen = udp:sip.qa.hbex.com:5061 tls_certificate = "/ home/inop/ex.com-cert.pem" tls_private_key = "/home/inop/ex.com-privkey.pem" tls_ca_list = "/home/inop/ex.com-calist.pem" tls_require_client_certificate=0 tls_verify_client=0 tls_verify_server=0 # ------------------ module loading ---------------------------------- loadmodule "/home/interop/openser/lib/openser/modules/rr.so" loadmodule "/home/interop/openser/lib/openser/modules/xlog.so" loadmodule "/home/interop/openser/lib/openser/modules/textops.so" loadmodule "/home/interop/openser/lib/openser/modules/maxfwd.so" loadmodule "/home/interop/openser/lib/openser/modules/sl.so" loadmodule "/home/interop/openser/lib/openser/modules/mysql.so" loadmodule "/home/interop/openser/lib/openser/modules/tm_unchanged.so" loadmodule "/home/interop/openser/lib/openser/modules/usrloc.so" loadmodule "/home/interop/openser/lib/openser/modules/registrar.so" # ----------------- setting module-specific parameters --------------- modparam("registrar", "default_expires", 120) modparam("registrar", "use_domain", 1) modparam("usrloc", "use_domain", 1) 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) # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("maxfwd", "max_limit", 10) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","To Many Hops"); drop(); }; if (dst_port==5061) { if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { # rewrite destination and forward to jabber (sip server) route(1); return; }; sl_reply_error(); return; } if (method=="REGISTER") { xlog("XXX: saving location msg=$mb\n"); if(!save_noreply("location")) { log("XXX: Error saving location!\n"); sl_reply_error(); } sl_send_reply("200","OK"); return; } if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { #forward to self route(3); return; } ##forward to ext gateway route(2); } route[1]{ if (method!="MESSAGE" && method!="SUBSCRIBE" && method!="NOTIFY") { log("XXX: Request not forwarded to sip server!\n"); sl_send_reply("202","Accepted"); return; } t_on_reply("1"); record_route(); if(!lookup("location")) { if (method!="SUBSCRIBE") { log("XXX: only subscribes are processed for user that are not registered\n"); sl_reply_error(); return; } subst_uri('/(.*)@(.*)/\1#\2*jabber.qa.im.hb.com@jabber01.qa.im.hb.com/ig'); if(!t_relay()) { log("XXX: error forwarding jabber01...\n"); sl_reply_error(); return; } else { sl_send_reply("200","OK"); return; } } else { log("XXX: forwarding to the address of record...\n"); if(!t_relay()) { log("XXX: error forwarding to address of record \n"); sl_reply_error(); return; } else { sl_send_reply("200", "OK"); return; } } } # forwarding to external gateway route[2]{ log("XXX: rewriting headers\n"); subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); if(!forward("tls:43.123.141.166:3000")) { log("XXX: Error forwarding to external gateway!\n"); sl_reply_error(); } else { sl_send_reply("200", "Accepted"); }; } # forwarding to SELF route[3]{ log("XXX: rewriting headers\n"); subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); log("XXX: about to forward to self\n"); if(!forward("tcp:sip.qa.hbex.com:5061")) { log("XXX: Error forwarding to self!\n"); sl_reply_error(); } else { sl_send_reply("200", "Accepted"); }; } # this is executed for replies onreply_route[1]{ subst('/^(To:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); route(2); }

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