sr-users August 2007

sr-users@lists.kamailio.org

169 participants
286 discussions

by Daniel-Constantin Mierla

Hello, version 1.2.2 of OpenSER has been released. This is a minor release in the branch of 1.2 release series, containing only fixes to bugs found since the previous release 1.2.1. See all changes at: http://www.openser.org/pub/openser/1.2.2/ChangeLog The configuration script used with any 1.2.x version is compatible. There was a change in the structure of table "active_watchers" for handling local and remote CSeq numbers -- you have to recreate this table if you are using presence module (instructions are provided here: http://openser.org/dokuwiki/doku.php/install:1.2.1-to-1.2.2). Source tarball can be downloaded from: http://www.openser.org/pub/openser/latest/src/ Or from sourceforge download page: http://sourceforge.net/project/showfiles.php?group_id=139143 Download page providing several options is available at: http://www.openser.org/mos/view/Download/ The documentation for modules is posted at: http://openser.org/docs/modules/1.2.x/ Main documentation page is available at: http://www.openser.org/mos/view/OpenSER-Documentation-Repository/ If someone wants to contribute with packages for different distros, please send the link from where can be downloaded to team(a)openser.org. We would like to thank to all developers and contributors for their work with coding and testing for this new release. Cheers, Daniel

16 years, 11 months

[Serusers] Presence annoyance.

by SIP

There is now a Grandstream GXV3000 in my network (neat little videophones, I must say). However, when someone sends a presence event through my server, the grandstream responds with a SIP packet and additional XML info: U 99.101.102.103:5060 -> 172.17.240.240:54330 NOTIFY sip:1101XXXXXXX@172.17.240.240:54330 SIP/2.0. Record-Route: <sip:99.101.102.103;ftag=f0a1719429e37564;lr=on>. Via: SIP/2.0/UDP 99.101.102.103;branch=z9hG4bK4204.13bfa134.0. Via: SIP/2.0/UDP 66.69.71.90:34202;branch=z9hG4bK04c43fa29c41b774. From: <sip:1101ZZZZZZZ@proxy.ourdomain.com>;tag=f0a1719429e37564. To: "Jessa"<sip:1101XXXXXXX@proxy.ourdomain.com>;tag=1b55e40b. Contact: <sip:1101ZZZZZZZ@66.69.71.90:34202>. Call-ID: ZjczZGMzOWEzMTk0NjhkMzRjMjI3ODZkNzIwMGIxZDA.. CSeq: 1843 NOTIFY. User-Agent: Grandstream GXV3000 (HW 0101, Ch:0) 1.0.1.7. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Event: presence. Subscription-State: active. Content-Type: application/xpidf+xml. Content-Length: 342. P-hint: LR. . <?xml version="1.0"?>. <!DOCTYPE presence. PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">. <presence>. <presentity uri=""Jessa"<sip:1101XXXXXXX@proxy.ourdomain.com>;tag=1b55e40b;method=SUBSCRIBE" />. <atom atomid="1000">. <address uri="<sip:1101ZZZZZZZ@66.69.71.90:34202>">. <status status="open" />. </address>. </atom>. </presence> The XML info is causing errors in the logs (which I've been ignoring for now, but that's not really an optimal solution). Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_F ORWARD header failed! Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_uri: uri too short: <> (0) Aug 15 15:16:12 proxy ser[19620]: is_user_in(): Error while parsing URI Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: find_first_route: Error while parsing headers Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_uri: uri too short: <> (0) Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_sip_msg_uri: bad uri <> Aug 15 15:16:12 proxy ser[19620]: WARNING: do_action:error in expression Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: find_credentials(): Error while parsing header s Aug 15 15:16:12 proxy ser[19620]: pre_auth(): Error while looking for credential s Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR: build_res_buf_from_sip_req: alas, parse _headers failed Aug 15 15:16:12 proxy ser[19620]: pre_auth(): Error while sending 400 reply Other than forwarding presence information to a presence-capable server that understands the XML, is there anything I can do on the SER side to ignore the XML stuff so I don't get a screen full of error messages every time a presence event comes through? N.

16 years, 11 months

[Serusers] presence server 2 presence server

by cherami＠gmx.de

Hi, I am student and new to the Presence Model und Protocols. I have in my student project the following scenario: company A wants to develop a presence service PA. The presence service PA should get the data from the presence server PB of an other Company B over the Internet. The Presence information of PA should be made available in the Company A for different applications. my question: can PA poll or subscribe all changed/updated Presence Information from PB regularly. Or it should request the Presence Information of each Presentity individually? which Presence server can implement the presence service PA. What should I do if PB uses another Protokoll? Thank you and Best Regards Koder -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kanns mit allen: http://www.gmx.net/de/go/multimessenger

16 years, 11 months

[Serusers] ser asterisk pstn

by ronald＠stanford.edu

hello all, I am stuck and need your help for +xlite +ser +asterisk +pstn calls. I dial 914082743304 and get "caller Not Found". I am able to authenticate and complete sip -> sip calls, but cannot complete PSTN calls I did monitor asterisk and no reroute was offered. please help! permissions.allow: ^sip:91[0-9]{10}@bigu\.edu$ ^sip:9011[0-9]*@bigu\.edu$ ^sip:9[0-9]{10}@bigu\.edu$ ngrep -vvvv proto udp port 5060 produces the following response: 171.66.176.123:47729 -> 192.168.11.243:5060 INVITE sip:914082743304@bigu.edu SIP/2.0..Via: SIP/2.0/UDP 192.168.1.100:47729;br anch=z9hG4bK-d87543-4f5eb56fc40f1f43-1--d87543-;rport..Max-Forwards: 70..Contact: <si p:cell@192.168.1.100:47729>..To: "914082743304"<sip:914082743304@bigu.edu>..From: "SER EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5 NTdlOGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER , NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agent: eyeBea m release 1011s stamp 41121..Content-Length: 421....v=0..o=- 1 2 IN IP4 192.168.1.100 ..s=CounterPath eyeBeam 1.5..c=IN IP4 192.168.1.100..t=0 0..m=audio 32942 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101..a=fmtp:18 annexb=yes..a=fmtp:101 0-15..a=rtpmap:107 BV 32/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap:105 SPEEX- FEC/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv..a=x-r tp-session-id:AEB31A44A04C4B32BD3D72349E4FBB80.. # U 192.168.11.243:5060 -> 171.66.176.123:47729 SIP/2.0 404 caller Not Found..Via: SIP/2.0/UDP 192.168.1.100:47729;branch=z9hG4bK-d875 43-4f5eb56fc40f1f43-1--d87543-;rport=47729;received=171.66.176.123..To: "914082743304 "<sip:914082743304@bigu.edu>;tag=21f9bfca23409407a263e4a7e40abe9d.f2a8..From: "SE R EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5NTdl OGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Server: Sip EXpress router (0.9.4 (i386/linux))..Co ntent-Length: 0..Warning: 392 192.168.11.243:5060 "Noisy feedback tells: pid=15307 re q_src_ip=171.66.176.123 req_src_port=47729 in_uri=sip:914082743304@bigu.edu out_u ri=sip:914082743304@bigu.edu via_cnt==1".... debug=3 fork=yes log_stderror=no listen=1xx.x4.11.xx3 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 alias=sip2.bigu.edu dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser[1]" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser[2]") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") #added these timers for voicemail forwarding modparam("tm", "fr_inv_timer", 25) modparam("tm", "noisy_ctimer", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("1xx.x4.11.xx3:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("sip2.bigu.edu","subscriber")) { www_challenge("sip2.bigu.edu","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:1[0-9]{10}@bigu.edu") { strip(1); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (uri=~"^sip:011[0-9]*@bigu.edu") { # International PSTN route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@bigu.edu") { # Domestic PSTN route(4); route(5); break; }; sl_send_reply("404", "caller Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("1xx.x4.8.xx8"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Links: ------ [1] mysql://ser:heslo@localhost/ser [2] mysql://ser:heslo@localhost/ser

16 years, 11 months

[OpenSER-Users] CallManager Express + Exchange 2007

by Brian D

Would anyone be willing to share their configuration of where they have integrated these two using OpenSER? Thanks - b

16 years, 11 months

[Serusers] Out of Office

by chayn＠imhocorp.com

Hello, I am out of the office and will be unable to respond to messages until Thursday, August 16. Claude

16 years, 11 months

[Serusers] ser to sip provider

by julia heller

Hi, i attempt to connect ser to a sip provider with user + password authentication. if i get it right, i have to load the uac modul and insert following line: modparam("uac","credential","user:sip-provider-ip:password"), but i can't find the uac modul .. i hope anybody can help me .. another way i tried is this one: route[x] { rewriteuri("username@sip-provider-ip"); rewriteuserpass("username:password"); rewritehost("sip-provider-ip"); forward("username:password@sip-provider-ip"); } but it worked either. it comes always up with: get_connection(): Connection not found in the pool .. kind regards, julia heller -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer

16 years, 11 months

[OpenSER-Users] question about DIALOG module

by Victor Gamov

Hi All As I understand dialog may be terminated by DIALOG module, e.g. due session timeout. In this situation module sends BYE timeout. When RTPproxy used does it receive signal to close RTP session whel local BYE generated? One more question -- is it possible to terminate session when RTPproxy does not get RTP packets for some predefined time? may be RTPproxy must be extended? Thanks! -- CU, Victor Gamov

16 years, 11 months

[Serusers] Regarding SNMP and Statistics in SER

by Parminder.Nayyar＠relianceada.com

Hi friends I have been on the look out for someting related to performance counters or statistics in SER. If i browse through the code, I find, that there is #ifdef STAT macro defined, which is used if STATs are enabled while compilation. Now, when i try to do that, it looks for a SNMP module related files... which it doesnt find... On searching on internet i find a page saying SNMP is dead... Can any one help me in finding out how to enable statistics for SER? It is extremely urgent for me... please help .. :(( Thanks Parminder Nayyar The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

16 years, 11 months

[Serusers] routing between ser and asterisk

by Steffen Waldmann

hi there, i have a question about routing between SER and asterisk: I run SER and * on the same machine, so i configured * to listen on port 5070 (sip.conf: bindport 5070). Now i want to route all numbers that are not found in SER to *, therefore i wrote to ser.cfg in the main route section: if (!lookup("location")) { #sl_send_reply("404", "nicht gefunden"); rewritehostport("192.168.2.183:5070"); route(1); break; }; route(1) is the following: route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } i have two SER-users: 1000 and 1001 i have one * user: 1002 when i dial 1002 i get immediately 'User not found' does anyone have a hint?

16 years, 11 months

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sr-users August 2007