sr-users January 2008

sr-users@lists.kamailio.org

131 participants
218 discussions

[OpenSER-Users] Configuring openser to forward incoming to a conference agent
by Pascal Maugeri 09 Jan '08

09 Jan '08

Hi I'm developing a "conference agent" that will receive INVITEs and invites a list of invitees defined in the SIP INVITE message payload. Let's say that this agent is available behind the URI sip:conf@yourdomain.com and openser should forward all incoming "INVITE sip:conf@yourdomain.com SIP/2.0" to my agent. How should I configure openser to do that ? The easiest way would be to have my conference agent to REGISTER against the openser but I don't want to maintain this registration and I think there is a better way to do that. I've seen there is also a function called "ul_insert_urecord" in USRLOC module. Might it be a possible solution to add a new entry for my conference agent ? Any other solution or idea ? Regards, Pascal

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[OpenSER-Users] presence caching and retrieval
by Sigrid Thijs 09 Jan '08

09 Jan '08

Hi, I was wondering, when a SIP user publishes his presence to the openser presence module, is this presence data cached in memory by the server or is this immediately stored in the database? Is this somehow configurable? If the presence data is cached in memory, is there an interface available for an external server to query the presence of one or more resources? In particular, we are looking for a performant way to query the presence of multiple users (for example 20). We would like to avoid the overhead of creating a SIP subscription for each resource: - to avoid traffic overhead - we only need to know the current presence of the resource(s) and are not interested in future presence changes. Do you have any suggestions on if/how this can be achieved with openser? kind regards, Sigrid Thijs

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[OpenSER-Users] Working in TCP but NOT in UDP?
by David Villasmil Govea 09 Jan '08

09 Jan '08

Hello all, I've got a working conf on which RTPs work perfectly with TCP, but fails to work with UDP... can anyont tell me why?? Here's my conf: # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=10 #fifo="/tmp/openser_fifo" # --- module loading mpath="/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" #loadmodule "dispatcher.so" loadmodule "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "xlog.so" #loadmodule "/usr/lib/openser/modules/acc.so" #modparam("acc", "early_media", 0) #modparam("acc", "report_cancels", 1) #modparam("acc", "db_flag", 2) #modparam("acc", "db_url", "mysql://user:pwd@localhost/openser") #modparam("acc", "report_ack", 0) #modparam("acc", "db_missed_flag", 3) #modparam("acc", "failed_transaction_flag", 4) #modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") # --- setting module parameters modparam("usrloc|auth_db","db_url","mysql://user:pwd@localhost/openser") # --- setting module parameters #modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 3) # -- registrar params -- #modparam("registrar", "received_avp", "$avp(s:rcv)") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("registrar|nathelper", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@whatever.com") #modparam("nathelper", "received_avp", "$avp(s:rcv)") # -- xlog modparam("xlog", "buf_size", 8192) modparam("registrar", "max_contacts", 5) # --- main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("OPENSER_IP", "subscriber")) { www_challenge("OPENSER_IP", "0"); exit; }; if (isflagset(5)) { setbflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; # # - TERMINATIONS # if (method=="INVITE") { if (!proxy_authorize("OPENSER_IP","subscriber")) { if($si=="GW_IP"){ xlog("L_NOTICE", "\n************** ALLOWING INCOMING CALL AS IT COMES FROM $si ************\n"); }else{ proxy_challenge("OPENSER_IP","0"); exit; } } if (uri=~"^sip:00"){ xlog("L_NOTICE", "\n************** GOT OUTSIDE CALL ************\n"); strip(2); prefix("M3ll4m0d4v1d"); rewritehostport("GW_IP:5060"); sl_send_reply("100","Trying"); forward(); exit(); } } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; } MANY thanks to all. David

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[OpenSER-Users] Carrirroute
by TCB 09 Jan '08

09 Jan '08

Would appreciate some help with CarrierRoute module I get the following error ERROR:carrierroute:carrier_rewrite_msg: error during rewrite_uri_recursor This is my Route route[22] { # route calls based on hash over callid # choose route domain 0 of the default carrier if(!cr_rewrite_uri("0", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In case of failure, re-route the request t_on_failure("3"); # Relay the request to the gateway t_relay(); } } -- TC

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[OpenSER-Users] INVITE, exec_dset(), and BYE
by Juan Carlos Castro y Castro 09 Jan '08

09 Jan '08

I have a scenario in which an OpenSER machine distributes load among several Asterisk machines for outgoing PSTN. I use an external program which I call through exec_dset() to select which server a call goes to. I use record_route(). All calls are completed OK, but when I dial from some SIP user-agents and hang up from the caller side, OpenSER gets crazy and doesn't know where to route the BYE to. Here's the reason: Good scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:providercode#5551234@chosenserver SIP/2.0" - I get an OK back, all is well. (Sometimes the # in the uri above is sent as %23, but it works either way) Bad scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:5551234@mydomain SIP/2.0" - My openser.cfg doesn't know how to handle that and I get a "Loop Detected" back. The destination never gets a hangup signal. Is the second UA from the "bad" scenario disrespecting the RFC? Do I have any recourse to route a BYE that comes like that to the right Asterisk server? Thanks, Juan

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[OpenSER-Users] OpenXCAP: no percent-encoded URI allowed? can't fetch element that beneath multiple-folded elements?
by KevinKinnan 09 Jan '08

09 Jan '08

The OpenXCAP 0.9.7(version 0.9.7) allows the HTTP URI that is not percent-encoded while the percent-encoded URI is rejected by the server. There is no problem of installation. Two types of http URIs are shown below: Accept: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour…"friends"] Reject: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… Another problem is about fetching element that within multiple-folded elements, for example: <list name="friend"> <entry uri="sip:alice@example.com"> <display-name>Alice</display-name> </entry> <entry uri="sip:bob@example.com"> <display-name>Bob</display-name> </entry> </list> <list name="closefriends"> <entry uri="sip:tom@example.com"> <display-name>Tom</display-name> </entry> <entry uri="sip:mike@example.com"> <display-name>Mike</display-name> </entry> </list> If fetching the element of "mike", the responsed to the following URI is "404 Not found". http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… However, an alternative way of uing element sequence number works. http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… Hope anyone can give me some clues, thanks a billion Regards, Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

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[OpenSER-Users] Working in TCP but NOT in UDP?
by David Villasmil Govea 09 Jan '08

09 Jan '08

> > > Hello all, > > > > I've got a working conf on which RTPs work perfectly with TCP, but > > fails to work with UDP... can anyont tell me why?? > > Could you detail more what the issue is? Do you mean that the SIP works > fine > over TCP and fails over UDP? RTP has nothing to do here. > > > -- > I?aki Baz Castillo > Hello, That's exactly what I mean, signalling being TCP or UPD got nothing to do here.... but it DOES work on TCP and NOT over UDP! Weird... David

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Re: [OpenSER-Users] how to restrict invite when regiest unsuccessful
by liu york 09 Jan '08

09 Jan '08

yes, when i used same authenticating on both regiest and invite.the problem was resolved.many many thx. ======= 2008-01-08 15:16:35 您在来信中写道：======= > > >Alex Balashov schrieb: >> On Tue, 8 Jan 2008, liu york wrote: >> >>> I want to restrict invite when regiest unsuccessful even the invite >>> request have right authentication. actually,i just want to only one >>> person can call to pstn,others cant call even have sipid and >>> password.now i has set by modparam("registrar", "max_contacts", 10).but >>> cant restrict the invite by the sipphone no need regiestion. >> >> Are you authenticating INVITEs differently than you are authenticating >> registrations? In other words, are the authentication realm and required >> credentials different for your proxy_challenge() than they are for >> your www_challenge() for handling REGISTER requests? >> >> If they are, it seems that all you should have to do is do a lookup() >> and see if it fails. Presumably, you are doing a save() into your >> registrar/usrloc only on valid REGISTER challenges? >> >> >> i.e. >> >> if(is_method("INVITE")) { >> if(!lookup("location")) { > > if(!registered("location")) { >is sufficient > >btw: this does not give you any security > >regards >klaus > > >> sl_send_reply("403", "Forbidden -- REGISTER invalid."); >> exit; >> } >> } >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : +1-678-954-0670 >> Direct : +1-678-954-0671 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2008-01-09

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Re: [OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

Sorry, not all of the text I originally sent has appeared on the list archives, so here is the full email another time! :P Hi, can anyone provide any advice on getting CDRTool to work? I have installed CDRTool 6.0 on a Debian Linux system 2.6 Kernel with Apache/2.2.6 (Unix) PHP/5.2.5 I am attempting to get it to work with OpenSER only (I have commented out the Asterisk and sip_trace sections in global.inc) and I am recieving the error: Error initializing CDRTool datasource >From in my web browser. It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

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Re: [OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

mmm, again! :S well here is the bottom half of my email.... It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

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sr-users January 2008