sr-users August 2007

sr-users@lists.kamailio.org

169 participants
286 discussions

[OpenSER-Users] OpenSER Admin Training Session at VoN Europe 2007 Conference, Rome, Italy, September 26

by Alicia Codrington

Good morning I wish to register for the above and need to pay via wire transfer. Please forward the relevant details in order that all relevant registrations for staff can be expedited. Thank you so much for your speedy response. Cordially ________________________________ your internet phone company www.ctpcom.net <http://www.ctpcom.net/> www.calltheplanet.com <http://www.calltheplanet.com/> Alicia Codrington Office Administrator [ p ] +1-246-228-2159 [ f ] +1-246-430-1418 [ @ ] alicia(a)calltheplanet.com ________________________________

16 years, 8 months

[OpenSER-Users] Re: LCR not running

by jh＠tutpro.com

martin, were you able to solve your "lcr not running" problem? i tried to understand how your gw and lcr tables looked like, but due to line breaks was not able to figure it out. if you still have the "no gateways" found problem, please email to the list your table contents in readable form (also telling null and empty string from apart). -- juha

16 years, 8 months

[OpenSER-Users] To URI

by Marc LEURENT

Hello everybody, I have booked fights, hotel and tickets for the Rome VON meeting!! I hope most of you will be there! I have a little problem... My OpenSER server has the IP 88.191.45.91 My SIP/PSTN gateway has the IP 212.129.6.65 I do not require authentication to send INVITE from my openser IP. How is it possible to change the To: <sip:0677832974@sd-7501.dedibox.fr:5060;user=phone>. to To: <sip:0677832974@212.129.6.65:5060;user=phone>. ?? Thanks!! See you soon in Rome # U 88.191.45.91:5060 -> 212.129.6.65:5060 INVITE sip:0677832974@212.129.6.65:5060 SIP/2.0. Record-Route: <sip:88.191.45.91;lr=on;ftag=c0a80101-1ca265a>. Via: SIP/2.0/UDP 88.191.45.91;branch=z9hG4bKd06b.6af66bc6.0. Via: SIP/2.0/UDP 192.168.95.81:5060;rport=62704;received=81.57.0.22;branch=z9hG4bK4296575315214209209. From: "Marc LEURENT"<sip:mleurent@sd-7501.dedibox.fr:5060;user=phone>;tag=c0a80101-1ca265a. To: <sip:0677832974@sd-7501.dedibox.fr:5060;user=phone>. Call-ID: 7b7f245-c0a80101-0-20(a)192.168.95.81. CSeq: 1 INVITE. Max-Forwards: 50. Supported: timer, replaces. Session-Expires: 1800. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO. Contact: <sip:mleurent@81.57.0.22:62704;user=phone;nat=yes>. User-Agent: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4. Allow-Events: refer,dialog,message-summary,check-sync,talk,hold. Content-Type: application/sdp. Content-Length: 295. . v=0. o=mleurent 30025307 30025307 IN IP4 192.168.95.81. s=-. c=IN IP4 88.191.45.91. t=0 0. m=audio 35154 RTP/AVP 8 0 18 4 97. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:97 telephone-event/8000. a=fmtp:97 0-15. a=sendrecv. a=nortpproxy:yes.

16 years, 8 months

[Serusers] why combine ser with asterisk

by Nhadie

Hi All, What's the advantage of combining ser with asterisk? I always see comments like using ser with asterisk is a very good solution etc. etc. the thing i liked with ser is that it does not do codec translation, which saves me cpu usage and also bandwidth. if i combine it with asterisk, would it not use codec translation? i also read that there is also a problem when ser and asterisk is run on the same machine, why is it so? if use prepaid billing solution for asterisk like astcc, would i then be able to provide prepaid service? soryy for asking too much, i'd just like to really understand it. Thank You in advanced. Regards, Nhadie

16 years, 8 months

[Serusers] Calls Per Second

by Jesús Terán

Hello, I want to know How many simultaneous calls can i have with the SER? I believe this depends with the Fisical Memory and Processor, i did a test with a traffic generator with a SER installed in a PC with 150 MB of RAM and Pentium 3 with 650 MHZ, and with 15 calls per second the CPU goes 100% and goes down. What kind of server do you recomend us to get a lot of calls per second? Att: Jesus Teran

16 years, 8 months

[OpenSER-Users] I've questions about OpenSER

by mario gu

Dear All, I'm newbody on OpenSER, and i've some questiones about OpenSER as follow, 1. Can OpenSER run over Linux Redhat 9.0? If yes, how do I do? 2. From overview of OpenSER, I know that OpenSER can act as SIP dispatcher server, can you share more detail about it? Appreciate you very much in advance. Mario Gu mariogu2004(a)hotmail.com _________________________________________________________________ 与联机的朋友进行交流，请使用 MSN Messenger: http://messenger.msn.com/cn

16 years, 8 months

[OpenSER-Users] avp_check("$avp(some_avp)", "re/$avp(some_regex_avp)/g"), use AVP as Regex?

by Brandon Armstead

Hello, Is it possible to use an AVP as a regex string? Everytime I get 'regexp operation requires string value'. I'm looking for a way to compare a value against a list of regex strings... i.e. avp_check("$avp(some_avp)", "re/$avp(some_regex_avp)/g"); when $avp(some_regex_avp[0]) is equal to a regex string, $avp(some_regex_avp[1]) is equal to some regex string, and so on and so forth, any help would be appreciated, thank you!

16 years, 8 months

[OpenSER-Users] Interop with PSTN gateway sometimes fails - openser1.2.1

by Naija Man

Hi all, I am a beginner in openser and I really appreciate the help I have been getting from this group. I am setting up an interop with a PSTN gateway with intermittent success. I will try to explain my setup below. redirect server / / 2 / / 3 1 / / 4 5 asterisk -----> openser (v1.2.1) -----> pstnGateway ------> PSTN (cell phone) Calls are generated from asterisk (step 1) with a 3-digit prefix (say, 999) on the ANI to identify a particular group of calls, and sent to openser ( v1.2.1). Openser checks the from_uri; detects the 999 prefix, strips the leading 3 digits and forwards the INVITE to the redirect server (step 2). The redirect server responds with a 301, containing the location of the pstnGateway (step 3). Openser sends an INVITE to address in the 301 to the pstnGateway (step 4) and the call gets completed (step 5) This setup works intermittently. I sometimes have to restart openser before it works again. When it works, I get a 100 trying message back from the PSTN gateway, but when it fails, I see a number of invites sent from openser to the gateway without getting a 100 trying message back. And the invite times out with a 408 message. After a few seconds, the PSTN number actually rings, but I get dead air. My configuration is as below: ********************************************************** loadmodule "uac_redirect.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/etc/openser_fifo") modparam("rr", "enable_full_lr", 1) modparam("tm", "wt_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "fr_timer", 2) modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("uac","from_restore_mode","auto") # main routing logic route { if (!mf_process_maxfwd_header("16")) { xlog("L_INFO", " Reply: 483 - Too Many Hops\n"); sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { xlog("L_INFO", " Reply: 513 - Message Too Big\n"); sl_send_reply("513", "Message too big"); exit; }; if (loose_route()) { record_route(); t_relay(); exit; }; record_route(); # Outgoing route route(1); route(1); } # ROUTE 1 - OUTGOING CALLS route[1] { # Check prefix - if the ANI of the tenant starts with 999 # and the total number of digits is more than 12 if ((from_uri=~"^sip:999.*")&&$(fu{s.len}) > 12) { # replace both display and uri # Strip out the leading 999 digits before sending it to redirect server uac_replace_from("$(fU{s.substr,3,0})","sip:$(fU{s.substr ,3,0})@$fd"); # Check for re-directs t_on_failure("4"); xlog("L_INFO", "RequestUri:[$ru] FromUri:[$fu] forwarding to redirect server...\n"); ds_select_dst("6", "0"); # forward to redirect server t_relay(); exit; } } } ############################################################################### # Process Replies ############################################################################### onreply_route[1] { if( status =~ "18[0-9]" ) { # Reset the flag t_on_failure("0"); } } #============================================= # Default redirect handler - PSTN gateway inter-op #============================================= failure_route[4] { get_redirects("*"); t_relay(); } ********************************************************************************************* Below is a snippet of my logs when the call fails. -----Call 1 Aug 28 19:53:27 [12103]: [INVITE] [from: asterisk] [calling: 15552121212] [caller: 99915553131313] Aug 28 19:53:27 [12103]: RequestUri:[sip:15552121212@openser] FromUri:[ sip:99915553131313@asterisk] forwarding to redirect server... Aug 28 19:53:35 [12103]: [ACK] [from: asterisk] [calling: 15552121212] [caller: 99915553131313] Aug 28 19:53:35 [12103]: ERROR:uac:replace_from: decline FROM replacing in sequential request in auto mode (has TO tag) Aug 28 19:53:35 [12103]: RequestUri:[sip:15552121212@openser] FromUri:[ sip:99915553131313@asterisk] forwarding to redirect server... Aug 28 19:53:41 [12107]: [BYE] [from: redirectServer] [calling: 99915553131313] [caller: 15552121212] -----Call 2 Aug 28 19:53:53 [12105]: [INVITE] [from: asterisk] [calling: 15552121212] [caller: 99915553131313] Aug 28 19:53:53 [12105]: RequestUri:[sip:15552121212@openser] FromUri:[ sip:99915553131313@asterisk] forwarding to redirect server... Aug 28 19:53:56 [12105]: ERROR:tm:w_t_relay: t_forward_nonack failed Thanks and many regards, Tolu

16 years, 8 months

[OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?

by David Loh

Hi All, Greeting. I've been struggle with OpenSER TLS implementation for more than a week, since I've ported from UDP to TLS, everything work fine except the "BYE" request from Asterisk (loose route), my implementation was something like below: [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] My OpenSER.cfg already configured to listen on two port which is :- "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or even voicemail) having no problem, but when the callee disconnect the call, caller will never get hang up :( I've attached my ethereal trace/ngrep to pastebin, http://pastebin.ca/673392 Wondering if anyone can help me with the broken "BYE" that returned from Asterisk ? Line #131, supposedly this line should have contain 2 Via header, one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", but somehow the TLS via header was gone !! (compare to previous ACK (Line #117) /INVITE (Line #51). Due to the missing TLS via header, OpenSER log file was complaining "protocol/port mis-match". The last BYE request (Line #256) is actually firing from Client, which contain the "TLS" via. I've even tried "force_send_socket" to port 5061 (instead of 5060) from loose route, but it complaining TLS certificate error, since Asterisk doesn't support TLS natively, I've no clue why is the ACK/INVITE/CANCEL work but not BYE. if (loose_route) { .... if(is_method("BYE")) { force_send_socket(IP:5061); } } Has any one gone through of this kinda OpenSER over TLS + Asterisk setup, I'm really appreciate if you can share your experience with me, or pin point what's the mistakes I made here. Thanks in advance. Regards, David Loh

16 years, 8 months

[OpenSER-Users] Need help in configuring Open SER - XMPP module with Jabberd

by Aditya Chaturvedi

Need some help in configuring Open SER xmpp module (I tried configuring using FPSDEM 2007 presentation pdf) I have one single server on which Open SER and jabberd both are running. I tried configuring xmpp module on Open SER installation by changing the openser.cfg file. I added following things in openser.cfg file: loadmodule "xmpp.so" modparam("xmpp","domain _separator","*") modparam("xmpp","gateway_domain"," sipxmpp.catch22") modparam("xmpp","xmpp_domain","xmppsip.catch22") modparam("xmpp","xmpp_host","xmpp.catch22") modparam("xmpp","backend","component") modparam("xmpp","xmpp_password","0915fc59edc8bd9425ddff892d89c22b2f3f0030") if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; 'sipxmpp.catch22', 'xmppsip.catch22', 'xmpp.catch22' , all are alias names of same machine. So using this I am able to send Message from SIP client to XMPP Client but not in opposite direction. Also message is getting delivered some times with some difficulty and some time more than one time. SIP client registered user - user1@openser_domain XMPP client registered user - user2@xmpp_host I sent message from SIP client to user2*xmpp_host@openser_domain(xmpp_host = xmpp.catch22 in my case) and it got delivered but from opposite side XMPP client is trying to send message to user1*openser_domain@xmpp_domain(xmpp_domain = xmppsip.catch22 in my case) which is not getting delivered. So at one place xmpp_host is working and at other xmpp_domain is used which is not working. I am getting confused with the settings of domains i.e. gateway_domain and xmpp_domain in openser.cfg file, Please give clarity on that. probably i am doing mistake there.One more question , how gateway_domain is different from openser_domain?? is there some more documentation from where i can take reference? Thanks, Aditya

16 years, 8 months

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sr-users August 2007