sr-users August 2005

sr-users@lists.kamailio.org

220 participants
558 discussions

[Serusers] is SER Dependable for prepaid System ? Anyone using it for prepaid system?

by Sanjim Sanjim

is SER Dependable for prepaid System ? Anyone using it for prepaid system? I very much like the SER performance in the test environment. Only drawback(maybe advantage if we provide unlimited plan) ) is stateless operation. But I want to know anyone using it in realtime prepaid business? Thanks, JiM

18 years, 9 months

[Users] via header not added when transport=TCP?

by Erich Nahum

Howdy, I'm playing with OPENSER 0.9.5 with sipp as the workload generator on a bunch of linux boxes. When the transport is UDP, openser adds a via header like it is supposed to as a proxy. But when I use TCP no via header is added. This is a UAC->proxy->UAS scenario. Here's the header as seen at the UAS: | UDP message received [633] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Record-Route: <sip:10.0.1.42;ftag=1;lr=on> | Via: SIP/2.0/UDP 10.0.1.42;branch=z9hG4bKe9cc.cc24a79.0 | Via: SIP/2.0/UDP bronxville1:5060;received=10.0.1.40;branch=z9hG4bK-1-0 | From: sipp <sip:sipp@bronxville1:5060>;tag=1 | To: sut <sip:service@10.0.1.41:5060> | Call-ID: 1-31891@bronxville1 | CSeq: 1 INVITE | Contact: sip:sipp@bronxville1:5060 | Max-Forwards: 16 | Subject: Performance Test | Content-Type: application/sdp | Content-Length: 132 | P-hint: outbound And the corresponding message received at the UAS when using TCP: | TCP message received [495] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Via: SIP/2.0/TCP bronxville1:5060;branch=z9hG4bK-1-0 | From: sipp <sip:sipp@bronxville1:5060>;tag=1 | To: sut <sip:service@10.0.1.41:5060> | Call-ID: 1-31865@bronxville1 | CSeq: 1 INVITE | Contact: sip:sipp@bronxville1:5060 | Max-Forwards: 70 | Subject: Performance Test | Content-Type: application/sdp | Content-Length: 132 Note the absence of the 2nd Via header. Openser complains about this on the way back to the UAC. Any suggestions appreciated. Thanks, -Erich -- Erich M. Nahum IBM T.J. Watson Research Center Research Staff Member P.O. Box 704 nahum(a)watson.ibm.com Yorktown Heights NY 10598

18 years, 9 months

[Users] lcr from_gw in reply_route

by Klaus Darilion

Hi! Currently it is not possible to use the from_gw function in a reply_route. I'm no code expert, but to me it looks like it would be safe also to use it in a reply_route. Can please someone review this? regards, klaus

18 years, 9 months

[Serusers] Please help me with NAT...

by Yiannis Martis

18 years, 9 months

[Users] current calls

by Ozan Blotter

dear list, i'm still working on the billing system... there'r a few things left like active calls and active registers... from openser database, in "location" table i can read current registered users, and it's easy to make a PHP from there to show "active registers" from a web page. but in addition to it, i wan to see active calls, is there a specific way to do it from ser.cfg or database ? debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX port=5060 children=4 alias=212.XXX.104.XXX dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) 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 and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { 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 # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # this is our PSTN gateway route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; 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(); }; }

18 years, 9 months

[Users] Problems with digest authentication

by aimable

I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost and to the the IP address of the server, with no luck. And below is the result of ngrep I tried to grep the messages from the phones and here below is one message from a Cisco 186 ATA which has failed to register ########### Beginning of the capture ################## U PHONEIP:5060 -> SERVERIP:5060 REGISTER sip:SERVERIP SIP/2.0. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Contact: <sip:06090003@PHONEIP:5060;user=phone;transport=udp>;expires=3600. User-Agent: Cisco ATA 186 v2.16.2 ata18x (030829a). Content-Length: 0. # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8af0 . Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="talk.artel.rw", nonce="42edb29e1dbcc6fa814dd3396634ed7be68eea56". Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". Any idea? Aimable

18 years, 9 months

[Serusers] Xlite not working with SER+SEMS

by yu wang

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

18 years, 9 months

Re: [Serusers] URI rewrite utils

by Klaus Darilion

Pavol Segeč wrote: > Hi, > > But as I see in rfc3261 such header as Remote-Party-ID is not there, or I'm > wrong? What this headre means? In which rfc it is described? I suppose that It never reached the RFC state, it was only a draft, but is supported by Cisco, Asterisk, ser, Inalp, ... http://www.ietf.org/proceedings/02mar/I-D/draft-ietf-sip-privacy-04.txt > for wished functionality should we need rewrite From and Contact headers Do not rewrite From: as it is forbidden (used for dialog matching in RFC2543) Do not rewrite Contact:, user rpid. regards, klaus PS: cc to the mailing list > > pavol > > >>-----Original Message----- >>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >>Sent: Tuesday, August 02, 2005 1:31 PM >>To: Pavol Segeč >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] URI rewrite utils >> >>This can be achieved by using the Remote-Party-Id: header (rpid). >> >>Take a look at append_rpid_hf from auth module: >>http://openser.org/docs/modules/0.10.x/auth.html#APPEND-RPID-H >>F-NO-PARAMS >> >>regards, >>klaus >> >>Pavol Segeč wrote: >> >>>Hi, >>> >>>Ser has a few cmd which allows rewrite request uri or some >> >>part of it. >> >>>Does SER have some cmd which allow rewrite caller uri or >> >>some part of it? >> >>>For example, my uri is 1767(a)mydomain.com but when I'm >> >>calling outside >> >>>of my VoIP world to PSTN (for example through Cisco call manager) I >>>would like to identify myself by E.164 address format, i.e. >> >>+4214151341767(a)mydomain.com. >> >>>It should be helpful for me change my URI. >>> >>>Thanks, >>> >>>palo >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> > >

18 years, 9 months

[Serusers] Radius Authentication

by Ryan Pagquil

Guys, I'm testing SER to authenticate with radius. But when I start authenticating I see this on the radius log: rad_recv: Access-Request packet from host 127.0.0.1:1131, id=222, length=262 User-Name = "rpagquil@server4all" Digest-Attributes = "\n\nrpagquil" Digest-Attributes = "\001\014server4all" Digest-Attributes = "\002*42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-Attributes = "\004\020sip:server4all" Digest-Attributes = "\003\nREGISTER" Digest-Response = "1df283adcf333605c0007d8a86a2e332" Service-Type = Sip-Session Sip-URI-User = "rpagquil" Cisco-AVPair = "call-id=CE373D63037311DABFB500E04CAB4AB4@server4all" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 modcall: entering group authorize for request 150 modcall[authorize]: module "preprocess" returns ok for request 150 modcall[authorize]: module "chap" returns noop for request 150 modcall[authorize]: module "eap" returns noop for request 150 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "rpagquil" Digest-Realm = "server4all" Digest-Nonce = "42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-URI = "sip:server4all" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 150 rlm_realm: Looking up realm "server4all" for User-Name = "rpagquil@server4all" rlm_realm: No such realm "server4all" modcall[authorize]: module "suffix" returns noop for request 150 users: Matched rpagquil@server4all at 138 modcall[authorize]: module "files" returns ok for request 150 modcall[authorize]: module "mschap" returns noop for request 150 modcall: group authorize returns ok for request 150 rad_check_password: Found Auth-Type Digest auth: type "Digest" modcall: entering group Auth-Type for request 150 A1 = rpagquil:server4all:test A2 = REGISTER:sip:server4all KD = 94b43b69398cb3ca2eef355e9875d36f:42ef48a123c4e75c2d998852eaa5d4fb14bc9917:33f62c1688cd77a13c84b07b3877bb1c *rlm_digest: FAILED authentication modcall[authenticate]: module "digest" returns reject for request 150 modcall: group Auth-Type returns reject for request 150 auth: Failed to validate the user. Login incorrect: [rpagquil@server4all/<no User-Password attribute>] (from client me2 port 5060)* It says that there is no User-Password attribute contained in my authentication request. What could be the problem? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

18 years, 9 months

[Users] How to rewrite source IP -- SIP packet forwarding.

by Ray Van Dolson

I'm using nathelper with OpenSER 0.10.x. Here's a quickie diagram of my setup: [SIP ATA] <-------------> [OpenSER] <--------------> [Asterisk] 10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1> eth0 - <public_ip0> I'm using the basic nathelper example configuration, except for one modification. I removed the call to t_relay and replaced it with: t_relay_to_udp("<public_ip1>", "5060"); Basically I want a SIP packet to come in to eth1 on OpenSER server, get rewritten as needed by nathelper, then get sent out to public_ip1 on my Asterisk PBX. This appears to be working except a tcpdump on eth0 on OpenSER shows: 14:39:08.654908 IP 10.0.201.7.5060 > <public_ip1>.5060: UDP, length 629 Obviously the packets will never get back with that source IP address. What command should I be looking at to rewrite this IP so it shows up as <public_ip0> automatically? I'm trying to get SIP ATA's to REGISTER on my Asterisk box but proxy through OpenSER ... (and eventually make use of the rtp proxy). Can post more info if requested, I am new to SER. :-) -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

18 years, 9 months

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sr-users August 2005