sr-users May 2008

sr-users@lists.kamailio.org

121 participants
202 discussions

[OpenSER-Users] [OT?] voip-to-voip rating using cdrtool

by Adrian Georgescu

You can rate sip calls to specific SIP URIs, which are not telephone numbers in CDRTool. The SIP URIs must be fully qualified, that is you can set a price for user(a)example.com but you cannot do a wildcard matching for *(a)example.com Just add the SIP URI in the destination table as any other PSTN prefix. Regards, Adrian >>>>>>>>> Apologies if this is off-topic here, but this seemed like a good friendly place to ask. Apologies for being so naïve also... Question is, how can I get cdrtool to rate voip-to-voip calls when destination is non-numeric? As in, say, sip:holmes at example.com calls sip:watson at example.com. How do I get cdrtool to match *(a)example.com or *(a)example2.com? (FWIW, I'm just trying to see how this works... no plan of rolling out this Evil Scheme into a real world deployment.) While cdrtool has decent documentation, I wish this was a little more easy on me. Has anyone read/reviewed this book yet? http://www.packtpub.com/building-telephony-systems-with-openser/book I'm specifically interested in the "Billing with Freeradius and CDRTool" part. Thanks, Sajith.

16 years, 6 months

[OpenSER-Users] presence not working with X-lite

by Amit Vijayvargiya

Hi, we are done all the configuration of presence module in openser.cfg file set all the parameters..........i.e. modparam("presence|presence_xml", "db_url", "mysql://openser:openser@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:domain:5060") modparam("presence", "fallback2db", 1) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'pres') modparam("presence", "expires_offset", 10) modparam("xcap_client", "db_url", "mysql://openser:openser@localhost/openser") modparam("xcap_client", "xcap_table","xcap") modparam("presence_xml", "db_url", "mysql://openser:openser@localhost/openser") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "integrated_xcap_server", 1) and route the PUBLISH and SUBSCRIBE massages if( is_method("PUBLISH|SUBSCRIBE")) route(2); route[2] is::::: route[2] { sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } when we are run X-lite is asking you want to allow or deny the user-B........... but not showing online even we are allow the user-B.......... when we are allow the user x-lite not send any type of message...... we see in Notify openser sending a notify with watcher-list but with the status pending inside XML format. please tell me where we are wrong???????????? there is a problem with X-lite or our configuration????? Thanks in advance Amit Vijayvargiya

16 years, 6 months

[Serusers] Presence Server

by Allan Chao ( 趙廷睿 )

Dear All: I want to use SER's Presence module to do presence server , but someone say it's not stable , so , 1. Does anyone could tell me why it doesn't stable ? because it looks work after i do some testing with eyebeam. 2. how does openser's presence ? is it better than SER's presence ? thanks, allan DISCLAIMER: Sample Disclaimer added in a VBScript.

16 years, 6 months

[Serusers] TLS problem ser 2_0_0 - ERROR: tls_init.c:366: Unable to set the memory allocation functions

by .:: Francesco la Torre ::.

Have you found any solution for this issue ? I'm facing with a similar problem (not the same because the source of same error is line 520). Thanks in advance Francesco la Torre IIT/CNR-Pisa

16 years, 6 months

[OpenSER-Users] Presence, multiple status info handling

by Dan-Cristian Bogos

Guys, while dealing with presence/presence_xml modules, I came into one situation when one user crashes simultaneously and each time displaying different presence info (online, away, offline). Have noticed that the presence server is smart enough to inform all the other watchers about three different status information for the same user. My question is, how can the watcher know the most recent status information. Have looked on both order of info in the tuple as well as on ids but did not find any logic. Can u please guide me about possibilities I have? Thank you in advance, DanB PS: FYI, I am pasting such a tuple example. <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t9062"> <status xmlns="urn:ietf:params:xml:ns:pidf"> <basic>closed</basic> </status> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p8838"><rpid:activities/></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t5358"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Online</note> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p4226"><rpid:activities><rpid:away/></rpid:activities></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t7139"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Away</note> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p4606"><rpid:activities/></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t5632"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Online</note> </tuple></presence>

16 years, 6 months

[OpenSER-Users] Routing Logic

by VoIP Forums www.Go4Calls.com

Hi Friends, I have the following route plan in my openser.cfg all blocks are working well instead of 800800. if (uri=~"sip:800800[1-9][0-9]+@.*") { rewritehostport("officePBX-IP:5060"); route(1); exit; } else if ($(rU{s.len})>=8) { rewritehostport("MyPSTNprovider-IP:5061"); route(1); exit; }else{ lookup("location"); route(1); exit; }; When i am trying to call 800800 the error message appearing is "513 Message too big" 800800 is routed to the asterisk for our internal office's extensions. Pls guide us how i can solve this issue. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

16 years, 6 months

[OpenSER-Users] Authentcation using DISC is successful

by Dilip

Hello Sir, I have been able to make the DISC code compatible with openser-1.1.1-notls. I have eliminated some of the bugs which were there in the source code of DISC.But the thing is that the disc code is not able to work with x86_64 architecture. But it is successfully running with i386 arch. Could u please suggest me the neccessary task,i have to perform in order to make DISC compiled on x86_64 arch. I tried taking the successfully running code from i386 arch and copying it on x86_64 arch and then to compile it, but it is giving some architecture compilation problem which is given below. In file included from ../mem/shm_mem.h:53, from avp.c:33: ../mem/../lock_ops.h:182:2: error: #error "no locking method selected" ../mem/../lock_ops.h:265:2: error: #error "no lock set method selected" In file included from avp.c:33: ../mem/shm_mem.h:79: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ../mem/shm_mem.h: In function '_shm_malloc': ../mem/shm_mem.h:111: warning: implicit declaration of function 'lock_get' ../mem/shm_mem.h:111: error: 'mem_lock' undeclared (first use in this function) ../mem/shm_mem.h:111: error: (Each undeclared identifier is reported only once ../mem/shm_mem.h:111: error: for each function it appears in.) ../mem/shm_mem.h:113: warning: implicit declaration of function 'lock_release' avp.c: In function 'AAACreateAVP': avp.c:124: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:140: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c: In function 'AAAFreeAVP': avp.c:321: error: 'mem_lock' undeclared (first use in this function) avp.c:321: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type avp.c:323: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type avp.c: In function 'AAACloneAVP': avp.c:438: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:448: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:451: error: 'mem_lock' undeclared (first use in this function) avp.c:451: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type make[2]: *** [avp.lo] Error 1 Please Do help me ,to sort out the problem. regards, Dilip

16 years, 6 months

[Serusers] Asterisk->SER rtp problems

by Thorsten

Hi, I have an asterisk server running with an private IP. This asterisk forwards all calls to a SER server with a public IP. The SER server then forwards its calls to a public SIP provider. The problem now is that SER tries to stay in the loop which it shouldn't because there is no media proxy running. I don't get any audio because of this issue. But if I register the asterisk box directly to the SIP provider it works. Does anybody know how to fix this. My ser.cfg debug=4 # debug level (cmd line: -dddddddddd) #debug=0 fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #listen=0.0.0.0 #listen=82.98.89.140 port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/opt/ser/lib/ser/modules/mysql.so" loadmodule "/usr/local/ser/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/lib/ser/modules/avpops.so" #loadmodule "/usr/local/ser/lib/ser/modules/group.so" loadmodule "/usr/local/ser/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/lib/ser/modules/auth.so" loadmodule "/usr/local/ser/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/opt/ser/lib/ser/modules/auth.so" #loadmodule "/opt/ser/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) #modparam("registrar", "nat_flag", 6) #modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- 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=="ACK") { route(1); break; } if (method=="REGISTER") { #record_route(); save("location"); break; }; if (method=="INVITE") { #if (uri =~ "sip:[0-9]@*") { # if (nat_uac_test("19")) { # fix_nated_contact(); # fix_nated_sdp("3"); # } # route(3); # break; #} if (uri =~ "sip:[0-9]@*") { # record_route(); 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:[0-9]@*") { log(1, "Forwarding to mg3.net-m.de \n"); #rewritehostport("192.168.13.102:5060"); rewritehostport("62.214.145.199:5060"); #forward(62.214.145.199, 5060); route(1); break; } } My extensions.conf [toser] exten => _X.,1,Dial(sip/${EXTEN}(a)82.98.89.139) Thanks for any help Ciao Thorsten

16 years, 6 months

[OpenSER-Users] how to set FROM to a different uri

by sye

HI, I use aliases table set sip forwarding. During the forwarding I want to do is set FROM uri to a different sip address. ( I want to hide the original FROM uri). How do I do it? I check the cook book and don't see any function that I can use to do that work. Any hint? Thanks, Stanley

16 years, 6 months

[OpenSER-Users] regarding presence module facing problem

by Amit Vijayvargiya

hi, i m add presence module in openser 1.3. but it not working properly. when x-lite send a subscribe to client-B, it receive notify but with different body. i see in the table presentity it shows right body. where is the problem????? my configuration like this for subscribe and publish...... route[2] { sl_send_reply("100","trying"); append_to_reply("Contact: <sip:sip.pyrogroup.com:5060>\r\n"); if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } Thanks & Regards, Amit

16 years, 6 months

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sr-users May 2008