sr-users March 2007

sr-users@lists.kamailio.org

192 participants
381 discussions

[Users] Can not access script variables in openser 1.2.0
by Asterisk Expert 26 Mar '07

26 Mar '07

Hello, Sorry last mail sent accidentally. I've initialized some SCRIPT VARIABLES in request route. I modify the values of some of these SCRIPT VARIABLES in reply route. But I'm not able to get value of some of those variables in either request or reply route. Any idea? -- Regards Ruchir Brahmbhatt

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[Serusers] Changing expires in 200 OK for Register
by Baljeet Gill 26 Mar '07

26 Mar '07

Hi All, Is there a way to change the "expires" value in the 200 response (sent by SER) for REGISTER? Thanks, Baljeet To access the latest news from CSR copy this link into a web browser: http://www.csr.com/email_sig.php To get further information regarding CSR, please visit our Investor Relations page at http://ir.csr.com/csr/about/overview

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[Users] rewritehost
by Kro Kamil 26 Mar '07

26 Mar '07

hello, suppose that i am calling xxxxx(a)10.0.0.9 and i want to send this call to 10.0.0.10. is there any difference between rewritehost(10.0.0.10); and $ruri=sip:xxxxx@10.0.0.10; thanks _________________________________________________________________ Live.com'u deneyin: çevrimiçi dünyanızı bir araya getirin; haberler, spor, hava durumu ve çok daha fazlası. http://www.live.com/getstarted

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[Serusers] FIFO file
by Meidiana Ten 26 Mar '07

26 Mar '07

Hi, I'd like to know if we had 2 SER config file (home proxy and voicemail server) in same machine, do I have to setup different fifo file? For example, SER home proxy has fifo file /tmp/ser_fifo so what fifo I have to use for SER voicemail server? And how do I create the other fifo file? Thanx Regards, Meidiana --------------------------------- Don't be flakey. Get Yahoo! Mail for Mobile and always stay connected to friends.

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Re: [Users] Not able to access avp variables in reply route
by Asterisk Expert 26 Mar '07

26 Mar '07

Yes I know radius accounting will work but I have to do it without radius. So I'm doing it manually using avp_db_query. On 3/26/07, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > Hi > > U can use the Radius for account > > with Acc of radius some avp functions are there > > u means > like this > onroute_reply[] > { > if( status==200) > { > i f ( method == "invite" ) > accouting start of avp module > if methosd == bye) > acctoung stop of avp module > } > } > > > On 3/26/07, Asterisk Expert < asterisk.expert(a)gmail.com> wrote: > > > Hello, > > I'm trying to implement custom CDR using OpenSER. What I do is, insert a > > record in cdr table on invite request. I get the id of the inserted row > > using LAST_INSERT_ID() function of mysql in avp_db_query and store in avp > > variable rowid. I set connect time in reply route when I receive 200 OK for > > Invite request. When I get 200 OK for BYE request, I update the cdr record > > and store connect time, disconnect time, duration, etc. But at this > > place(reply route) I get the null rowid so as connect time. According to tm > > module's onreply_avp_mode parameter, I should be able to see the values set > > from request route in reply route if I set that parameter to 1 but it has no > > effect. I can not use either avp variables or script variables as > > transaction variables to record custom cdr. Can anyone guide me implementing > > this scenario or tell me what I've done wrong? > > -- > > Regards > > Ruchir Brahmbhatt > > Ecosmob Technologies > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > 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 -- Regards Asterisk Expert

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[Users] No audio when openSER is forworded to Asterisk server in the Same LAN network.
by raviprakash sunkara 26 Mar '07

26 Mar '07

Hello Users . For last 2 days onwards i'm fighting with this promlem, In integrating with Asterisk and openSER, No audio when openSER is forworded to Asterisk server in the Same LAN network. OpenSER and Asterisk are in the Same LAN network. With NAT , i forward the the port to 192.168.2.11 of openSER sip port and rtp port. With out NAT is Working Fine , openSER ip is 192.168.2.11 Asterisk is in 192.168.2.75 alias=sip.hyperion-tech.com if ( method =="INVITE") { if(uri=~"sip:1234567@sip.hyperion-tech.com || uri=~" sip:12345678@sip.hyperion-tech.com") { route(12); }; }; route[12] { rewrtitehostport("192.168.2.75:5060"); # ASTERISK SERVER if(!t_relay()) { sl_send_error(); exit; }; } -- 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] No audio when openSER is forwarded to Asterisk server in the Same LAN network.
by raviprakash sunkara 26 Mar '07

26 Mar '07

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[Users] VOIP Patent Bomb Goes Off In U.S.
by michael papet 26 Mar '07

26 Mar '07

For those of you that haven't heard, here's a nice rundown on Verizon's patent enforcement activities. http://ipurbia.com/2007/03/verizon-patent-analysis.html This news makes me sad.

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[Users] Openser not starting
by Knight Tiger 26 Mar '07

26 Mar '07

Hi I ran [root@openser ~]# openser start Listening on udp: A.B.C.D [A.B.C.D]:5060 udp: 127.0.0.1 [127.0.0.1]:5060 tcp: A.B.C.D [A.B.C.D]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 Aliases: tcp: localhost.my.domain:5060 tcp: localhost:5060 tcp: dhcpwl009.testdomain.com:5060 udp: localhost.my.domain:5060 udp: localhost:5060 udp: dhcpwl009.testdomain.com:5060 but I dont see any process running with the ps command So I checked the config files using -c [root@openser ~]# openser -c Listening on udp: A.B.C.D [A.B.C.D]:5060 udp: 127.0.0.1 [127.0.0.1]:5060 tcp: A.B.C.D [A.B.C.D]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 Aliases: tcp: localhost.my.domain:5060 tcp: localhost:5060 tcp: dhcpwl009.testdomain.com:5060 udp: localhost.my.domain:5060 udp: localhost:5060 udp: dhcpwl009.testdomain.com:5060 config file ok, exiting... 0(16175) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy [root@openser ~]# So the config files are okay but then openser does not start. So I tried root@openser % openserctl moni [cycle #: 1; if constant make sure server lives] and the process does not die.. I had to manually kill the process. I checked the subscriber table on the database and it was corect. I had registered a few users and it reflected them. The Postgres database is running. What am I doing wrong ? Any help would be appreciated. Thanks Knight

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[Users] OpenSER as a Load Balancer for Asterisk and RE-INVITEs
by Edoardo Serra 25 Mar '07

25 Mar '07

Hi all, I have an OpenSER acting as a registrar server and as a load balancer (with dispatcher module) for a cluster of PSTN Gateways based on Asterisk. When an inbound call is coming from the PSTN, Asterisks dial SIP/user@openser and the call is properly forwarded to the correct user in OpenSER location table But, if the user wants, for example, to put the call On-Hold, its user agent sends a RE-INVITE to OpenSER. My problem is that OpenSER (correctly) dispatch the call to a random Asterisk GW, and, if it's not the GW which is handling the call, On-Hold does not work and another call is originated. Tnx in advance for help Edoardo My openser.cfg follows # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # ----------- global configuration parameters ------------------------ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" #uid=nobody #gid=nobody # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/exec.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("usrloc", "timer_interval", 120) modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "db_url", "mysql://user:pass@192.168.252.5/db") modparam("uri_db", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "max_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("nathelper", "natping_interval", 20) # Ping interval 20 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("dispatcher", "force_dst", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) { sl_send_reply("405", "Method Not Allowed"); exit; } if (!method=="REGISTER") { record_route(); }; if ((src_ip==ip.of.asterisk.1) || (src_ip==ip.of.asterisk.1)) { if (!lookup("location")) { sl_send_reply("486", "Busy here"); exit; }; if (!t_relay()) { sl_reply_error(); }; exit; }; if (nat_uac_test("3")) { if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) { fix_nated_contact(); force_rport(); setflag(6); # Mark as NATed } } if (method=="REGISTER") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_from()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } } # loose-route processing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); exit; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); if ((src_ip!=ip.of.asterisk.1) && (src_ip!=ip.of.asterisk.2)) { ds_select_dst("1", "0"); } if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

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sr-users March 2007