- sr-users - kamailio.org

[Serusers] forward breaks URI rewrite?
by Chris Roberts 01 Sep '05

01 Sep '05

Heya, This might be an obvious answer, but when I forward a packet to another box using the forward() method, this seems to break the URI? rewriteuri("sip:test@test.co.uk"); forward(x.x.x.x, 5060); Now on my asterisk box and looking at ngrep, the test(a)test.co.uk has disappeared, and asterisk takes the SIPDOMAIN as 'x.x.x.x', rather than 'test.com'. Is this the desired behaviour? Is there a way to do what I want to do? I need to be able to pick out the original SIPDOMAIN in asterisk even after the packet has been forwarded by SER. People seem to have talked about doing this, but I'm not sure how I can forward it without obliterating the URI. Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.16/83 - Release Date: 26/08/2005

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[Serusers] suppress CLI in SIP-SIP calls
by Gerd Feiner 01 Sep '05

01 Sep '05

Hi, how do I suppress the CLI (CLIR) in SIP-SIP calls? It seems to me, that SIP-clients of any kind send the CLI information (or originating user-ID) in any case. Can I somehow suppress CLIs in SER on a per- user basis? Brgds, Gerd

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[Serusers] xmlrpc calls
by zkeatts 01 Sep '05

01 Sep '05

I am currently trying to do an xml-rpc call to usrloc.show_contacts. The xml packet I am sending is the following POST /RPC2 HTTP/1.0 User-Agent: Radio UserLand/7.1b7 (WinNT) Host: localhost:5060 Content-Type: text/xml Content-length: 131 <?xml version=\"1.0\"?> <methodCall> <methodName>usrloc.show_contacts</methodName> <params> </params> </methodCall> As a return packet I am getting HTTP/1.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:32769 Server: Sip EXpress router (0.10.99-janakj_experimental (i386/linux)) Content-Length: 303 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=5820 req_src_ip=127.0.0.1 req_src_port=32769 in_uri=/RPC2 out_uri=/RPC2 via_cnt==1" <?xml version="1.0" encoding="UTF-8"?> <methodResponse> <fault> <value><struct> <member><name>faultCode</name> <value><i4>400</i4></value></member> <member><name>faultString</name> <value><string>More Parameters Expected</string></value></member> </struct></value> </fault> </methodResponse> This tells me that I am not entering the correct information, but I am at a loss to exactly what else I should be inserting. I did a search for rpc_show_contacts and found in ./sip_router/modules/usrloc/ul_rpc.c static void rpc_show_contacts(rpc_t* rpc) After searching some more I could not find the rpc_t structure. Does anyone know what values I should be passing via XML? Thanks, Zach Keatts Software Engineer Nuvio.com

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[Serusers] URGENT:Authenitcation Mode
by Ashutosh Kumar 01 Sep '05

01 Sep '05

Hi, IF I set auth_db:calculate_ha1=no, will the radius authentication also be in non-digest (plaintext/basic) mode?? Reply please. Very urgent. Regards, Ashutosh Kumar

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[Serusers] maxfwd module with Max-Forwards > 255 broken
by Hendrik Scholz 01 Sep '05

01 Sep '05

Hi! I checked how SER would behave with large Max-Forwards values. Paragraph 20.22 of RFC 3261 notes Max-Forwards to be anywhere between 0 and 255 but SER lacks a check for the upper boundry in the maxfwd module. I'm not sure how the correct way to remove the broken header field and introducing a correct one would be so I leave this for somebody who knows :) Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

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[Serusers] overflowing Expire values
by Hendrik Scholz 01 Sep '05

01 Sep '05

Hi! I gave 3.1.2.4 of draft-ietf-sipping-torture-tests-07 a try and sent - a REGISTER with Expires > 2^31 and lower than 2^32 (2147483748 in my case). - a REGISTER with Expires > 2^32 (4294967596 = 2^32 + 300) I used print_expires() to print the resulting values SER (0.9.3 + changes) parsed. It's a negative value in first case and 300 in the latter one. parser/parse_expires.h defines exp_body_t to have 'int val' which is not enough to store the parsed values. Changing it to 'unsigned int val' prevents the negative values from coming up in SER but does not solve the main problem. The draft stats that a default value of 3600 seconds should be assumed in case of errors, so an additional check to make sure _e->val will always be below 2^32 is necessary. I do have a small patch for our non-standard branch and could wrap one up for HEAD if needed. Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

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Fwd: Re: [Serusers] Blocking calls from unregistered users
by Pavol Segec 01 Sep '05

01 Sep '05

Hi, The code means, that if an user is establishing a call (UA generate INVITE message) and the user is not already registered on your.domain register server, the user will be challenged to authenticate itself before the call is progressed, and only authentication of users from your.domain will be authorized (accepted). Personally I think that good practise is to put some code into ser.cfg where REGISTER message is handled, which allow to register (and make a call) only users, which have account on your.domain register (orproxy with register features) server, for example: sl_send_reply("100","Trying"); if (!www_authorize("your.domain", "subscriber")) { www_challenge("your.domain", "0"); break; }; # only registered users are allowed # do not register someone else if(!check_to()){ log(1,"LOG: unregistered user registration attempt\n"); sl_send_reply("401","Unauthorized"); sl_send_reply("403","Only registered users are allowed"); break; }; # save the registration process results if (!save("location")){ sl_reply_error(); }; BR pavol Citát Ryan Pagquil <rpagquil(a)philonline.com>: > Hi, > I'm currently subscribed to iptel.org. Well iptel.org's proxy doesn't > allow unregistered users to call to anybody, I already tried it. Does > having my domain specified in the proxy_authorize section means that > only users from my domain will be asked to register first before they > can place a call? I think I already tried this.. but I'll still test. =) > > Thanks, > Ryan > > Pavol Segec wrote: > > >Hi, > > > >If I understand well, you just need to put authorization challenge into your > >ser.cfg file where INVITE messages are handled, as follows: > > > >if(!proxy_authorize("your.domain","subscriber")){ > > proxy_challenge("your.domain","0"); > > sl_send_reply("403","Forbidden"); > > break; > > }; > > > >pavol > > > >CitĂĄt Dave <ddx66(a)yahoo.com>: > > > > > > > >>You can't as far as I know. You must use a USer Agent > >>that does not allow a user to make a calls unless the > >>UA is registered. > >> > >>--- rpagquil(a)philonline.com wrote: > >> > >> > >>--------------------------------- > >>Hi, I'm setting up ser so that unregistered users > >>can't make any calls to anybody. I have configured to > >>allow all other domains to make a call to my local > >>users. But when my local user that is unregistered it > >>can still make calls to other local users. How would I > >>do to block him totally? here is my ser.cfg: debug=3 > >>fork=yes log_stderror=yes listen=202.84.24.107 > >>port=5060 children=4 dns=no rev_dns=no > >>fifo="/tmp/ser_fifo" > >>fifo_db_url="mysql://ser:heslo@localhost/ser" > >>alias=sip.philonline.com #load module part > >>loadmodule "/usr/local/lib/ser/modules/mysql.so" > >>loadmodule "/usr/local/lib/ser/modules/domain.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/mediaproxy.so" > >>loadmodule "/usr/local/lib/ser/modules/nathelper.so" > >>loadmodule "/usr/local/lib/ser/modules/textops.so" > >>loadmodule "/usr/local/lib/ser/modules/acc.so" > >>loadmodule "/usr/local/lib/ser/modules/permissions.so" > >> #module parameter setup modparam("rr", > >>"enable_full_lr", 1) > >>modparam("auth_db|uri_db|usrloc|domain|permissions", > >>"db_url", "mysql://ser:heslo@localhost/ser") > >>modparam("auth_db", "calculate_ha1", 1) > >>modparam("auth_db", "password_column", "password") > >>modparam("usrloc", "db_mode", 2) > >>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("registrar", "nat_flag", 6) modparam("acc", > >>"log_level", 2) modparam("acc", "log_fmt", > >>"cdfimorstup") modparam("acc", "report_ack", 1) > >>#modparam("acc", "failed_transactions", 1) > >>modparam("acc", "log_flag", 1) #modparam("acc", > >>"report_cancels", 1) modparam("acc", "db_flag", 1) > >>modparam("acc", "db_missed_flag", 3) modparam("acc", > >>"db_url", "mysql://ser:heslo@localhost/ser") > >>modparam("uri_db", "uri_table", "uri") > >>modparam("uri_db", "uri_user_column", "username") > >>modparam("uri_db", "uri_domain_column", "domain") > >>modparam("domain", "db_mode", 1) modparam("domain", > >>"domain_table", "domain") modparam("domain", > >>"domain_col", "domain") modparam("permissions", > >>"default_allow_file", > >>"/usr/local/etc/ser/allow.permissions") > >>modparam("permissions", "default_deny_file", > >>"/usr/local/etc/ser/deny.permissions") #our routing > >>logic route { 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#### if (method=="INVITE" && > >>client_nat_test("3")) { > >>record_route_preset("202.84.24.107:5060;nat=yes"); > >> } else if (method!="REGISTER") { > >>record_route(); }; ###call tear down > >>section### if (method=="BYE" || > >>method=="CANCEL") { > >>end_media_session(); }; ###accounting### > >> if ((!has_totag() && (method=="INVITE" || > >>method=="ACK")) || (method=="BYE")) { > >>setflag(1); }; ###loose route### 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### > >>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); } ##Default message handler## route[1] { > >> t_on_reply("1"); if (!t_relay()) { > >> if (method=="INVITE" || method=="ACK") { > >> end_media_session(); > >> }; sl_reply_error(); }; } > >> ##Register message handler## route[2] { > >>sl_send_reply("100", "Trying"); if > >>(!search("^Contact:\ +\*") && client_nat_test("7")) { > >> setflag(6); > >>fix_nated_register(); force_rport(); > >> }; if > >>(!www_authorize("sip.philonline.com","subscriber")) { > >> > >>www_challenge("sip.philonline.com","0"); > >> break; }; if (!check_to()) { > >> sl_send_reply("401", "You are Unauthorized"); > >> break; }; > >>consume_credentials(); if (!save("location")) > >>{ sl_reply_error(); }; } > >>##INVITE message Handler## route[3] { > >>if (client_nat_test("3")) { > >>setflag(7); force_rport(); > >> fix_nated_contact(); }; if > >>(!search("To: .*(a)sip.philonline.com")) { if > >>(!proxy_authorize("","subscriber")) { > >>proxy_challenge("", "0"); break; > >> }; if (!check_from() && method=="INVITE") > >>{ sl_send_reply("403", "User > >>From=ID"); break; }; > >>}; lookup("aliases"); if > >>(uri!=myself) { route(1); > >> break; }; if (!lookup("location")) > >>{ sl_send_reply("404", "User not > >>found"); break; }; if > >>(method=="CANCEL") { route(1); > >> 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(); }; } Thanks, > >>--ryanRyan PagquilInfodyne Inc. > >>(www.philonline.com)Tel. (632)-6870715> > >>_______________________________________________ > >> > >> > >>>Serusers mailing list > >>>serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > >>> > >>> > >>> > >> > >> > >>____________________________________________________ > >>Start your day with Yahoo! - make it your home page > >>http://www.yahoo.com/r/hs > >> > >> > >>_______________________________________________ > >>Serusers mailing list > >>serusers(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > -- > Ryan Pagquil > Infodyne Inc. - PhilOnline.com > 3603 Antel Global Corporate Center > DoĂąa Julia Vargas Ave. > Ortigas Center Pasig City > Tel: 687-0715 > Web: www.philonline.com > > ----- Koniec preposlanej správy -----

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[Serusers] Dchp or Static ip
by chinaj123 01 Sep '05

01 Sep '05

To All, Has anyone tried to setup SER using a DHCP assigned address behind a NAT? I just want to know what the pro's and the con's were. Thanks, Alfred ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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[Serusers] Blocking calls from unregistered users
by rpagquil＠philonline.com 31 Aug '05

31 Aug '05

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[Users] DROP statement
by Bogdan-Andrei Iancu 31 Aug '05

31 Aug '05

Hi there, The core "drop" statement has a new special meaning in some route types (by default it's equivalent to "exit"): - BRANCH ROUTE - the request branch is discarded (not fwd); - ONREPLY ROUTE - if provisional reply (<200), it will be discarded (not fwd, not stored, but only timers updated); Wiki will be updated soon with this text ;) regards, bogdan

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[Serusers] RURI or URI
by Nando 31 Aug '05

31 Aug '05

hello, somebody knows the true difference between this: sip:1111@sip.domain.com y sip:1111 at sip.domain.com Because in logs (with ngrep) only I see the URI default????? (sip:1111@.....) Why i do not see the RURI??? Thanks

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[Serusers] Support for multiple PSTN Gateways (hunting, crank back)
by Frank Fischer 31 Aug '05

31 Aug '05

Hi all i have multiple gateway devices to PSTN. All are equal in terms of what destinations may be reached behind it or the termination costs. I use multiple devices for two reasons: capacity and failover. So what i need is functionality to hunt lines or at least doing a crank back (or even better both). The best way to this (IMHO) would be to track the line usage on each gateway and check it before routing to a gateway to make sure, there is a line available to place the call to (this would be a kind of line hunting). But i could also "live" with a solution that works more like a failover mechanism. In that case i would route the call to the first gateway in my list and if it is not available or has no more free lines, i would need to detect this, take the call back and place it on another gateway. As far as i could find out, there is no hunting, failover or crank back support directly built into SER. Is it possible to implement such functionality using the "common" routing scripts in SER? Did anyone of you ever had similar requirements and did you find a way to fullfill it? I'm very thankfull for any idea, hint, snipplet, .... Thanks a lot for your help. Best Regards Frank

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[Serusers] FWD: URGENT:Authenitcation Mode
by Ashutosh Kumar 31 Aug '05

31 Aug '05

Hi, Can I do basic mode authentication with radius. If yes, how? Regards, Ashutosh Kumar __ ______________________________________ From: Ashutosh Kumar [mailto:ashutoshk@chetu.com] Sent: Wednesday, August 31, 2005 9:51 PM To: 'serusers(a)lists.iptel.org' Subject: URGENT:Authenitcation Mode Hi, IF I set auth_db:calculate_ha1=no, will the radius authentication also be in non-digest (plaintext/basic) mode?? Reply please. Very urgent. Regards, Ashutosh Kumar

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[Serusers] Adding +1 to From:/To: for E.164 compliance
by Ryan Tucker 31 Aug '05

31 Aug '05

Greetings... We are starting to work with a carrier which requires From: and To: to be in E.164 format (e.g. +17005551212). prefix() only seems to change the INVITE, which doesn't appear to cut the mustard with these guys. Is there any way to make this happen within SER? Thanks! -rt -- Ryan S. Tucker Network Engineer ISP Direct, Inc. 1159 Pittsford-Victor Road Building 5, Suite 140 Pittsford, New York 14534 +1 585.419.8200 http://www.isp-direct.com/ http://www.vocal-net.net/ The information contained in this message may be privileged, confidential, and protected from disclosure. If the reader of this message is not the intended recipient, or any employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution, or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by replying to the message and deleting it from your computer. Thank you.

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[Users] set timer for Request Timeout
by Jens Carl 31 Aug '05

31 Aug '05

Hello all, I have following records in the table 'usr_preferences': username attribute value 1000 fwdbusy sip:1001@example.de 1000 fwdnoanswer sip:1001@example.de I use these records to get the functionallity of "Call Forwarding - Busy" and "Call Forwarding - No Answer" into the OpenSer - server. I use a failure_route to act on the SIP - messages "Request Timeout (408)" and "Busy(486)" and forward the INVITE - message to the URI saved in the table 'usr_preferences'. See my snippet of the openser.cfg: ----snip --- failure_route[1] { if (isflagset(27) && t_check_error(408)) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); route(6); break; }; }; if (isflagset(26) && t_check_error(486)) { if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); resetflag(26); route(6); break; }; }; }; ----snap --- These configuration works fine. Is there a possibillity to set a timer for the phone after the server receives the message "Request Timeout" from the phone? Or can I let the server ring for example five times and generate the message? Greetings Jens

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[Serusers] Error using TCP
by Federico Giannici 31 Aug '05

31 Aug '05

I'm using SER 0.9.3 with OpenBSD 3.7 amd64. Everything works correctly if I use normal UDP connections. Now I have to use a TCP connection (UDP is blocked in that network). Using ngrep I see SER receives the requests ma no reply is sent. The following errors are logged: Aug 23 12:48:06 eowyn /usr/local/sbin/ser[16406]: ERROR: send_fd: sendmsg failed on 19: Invalid argument Aug 23 12:48:06 eowyn /usr/local/sbin/ser[16406]: ERROR: send2child: send_fd failed Aug 23 12:48:06 eowyn /usr/local/sbin/ser[16406]: ERROR: tcp_main_loop: no children available So it seems that no TCP childrens are available but I can see them using "serctl ps": 0 10963 attendant 1 24595 fifo server 2 19807 receiver child=0 sock= 195.120.250.50:5060 3 24795 receiver child=1 sock= 195.120.250.50:5060 4 6812 receiver child=2 sock= 195.120.250.50:5060 5 27930 receiver child=3 sock= 195.120.250.50:5060 6 22186 timer 7 1034 tcp receiver 8 3296 tcp receiver 9 605 tcp receiver 10 15085 tcp receiver 11 18905 tcp main process What could be the problem? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Users] Combining AVPs
by Martin Kjeldsen 31 Aug '05

31 Aug '05

Hi Can anyone offer some helps on how to combine multiple AVPs. What I would like to do is to create an uri (eg. sip:bob@biloxi.com), where the "bob" part have been taken from e.g. the username column in the alias table and the rest either prefixed/appended from some static data or grabbed from another tuple! Can anyone offer some insight as to how this may best be accomplished. -- Med venlig hilsen / Kind Regards Martin Kjeldsen Cuatro-Group P/S (4G) Hindegade 3, DK-1303 København K Tel: (+45) 82 35 65 00 Fax: (+45) 82 35 65 65 Mob: (+45) 60 64 65 87 Url: http://www.cuatro.dk

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[Serusers] NATHelper IPv4-IPv6
by Ahmed Kellal 31 Aug '05

31 Aug '05

Hello everyone, I am working on setting up a heterogeneous platform with both IPv4 and IPv6 UAs. Problem I am facing right now is that I need to add a little to the code, in order to: 1- Extract the hostname part of the Contact header field, when needed 2- Change the said hostname contact 3- Extract the IP (v4 or v6) address in the owner ("o=") field of the SDP body 4- Change it with the value the RTP Proxy assigns dynamically to represent an IPv4 UA in the IPv6 world and vice versa In order to do these things, I tried to understand as much as I can of the code of the NAtHelper module. I wrote four functions to do the above, but (naturally enough in my case :-)) they're not doing what I want them to. So if anyone could help me out here, I would sure be grateful, esp. that I have a serious deadline! Thanks Here are the sections I added (I drew inspiration from the functions already existing in nathelper.c) /************* Added Section 1: Define the newly inserted functions ****************/ static int extract_sip_contact_host(struct sip_msg *, char *); static int extract_sdp_ownerip(str *, str *, int *); static int alter_sdp_ownerip(struct sip_msg *, str *, str *, int, str *, int, int); static int modify_contact (struct sip_msg *, char *, char *, char *); /************* End of Added Section 1 ******************************/ /********************* Added Section 2: The modify_contact function ******************/ /********* Replaces ip:port pair in the Contact: field with the hostname desired **********/ static int modify_contact (struct sip_msg* msg, char* str1, char* str2, char* hostname) { int offset, len, len1; char *cp, *buf, temp[2]; contact_t* c; struct lump* anchor; struct sip_uri uri; if (get_contact_uri(msg, &uri, &c) == -1) return -1; if (uri.proto != PROTO_UDP && uri.proto != PROTO_NONE) return -1; if (uri.port.len == 0) uri.port.s = uri.host.s + uri.host.len; offset = c->uri.s - msg->buf; anchor = del_lump(msg, offset, c->uri.len, HDR_CONTACT); if (anchor == 0) return -1; cp = hostname; len = c->uri.len + strlen(cp) + 6 /* :port */ - (uri.port.s + uri.port.len - uri.host.s) + 1; buf = pkg_malloc(len); if (buf == NULL) { LOG(L_ERR, "ERROR: fix_contact: out of memory\n"); return -1; } temp[0] = uri.host.s[0]; temp[1] = c->uri.s[c->uri.len]; c->uri.s[c->uri.len] = uri.host.s[0] = '\0'; len1 = snprintf(buf, len, "%s%s:%d%s", c->uri.s, cp, msg->rcv.src_port, uri.port.s + uri.port.len); if (len1 < len) len = len1; uri.host.s[0] = temp[0]; c->uri.s[c->uri.len] = temp[1]; if (insert_new_lump_after(anchor, buf, len, HDR_CONTACT) == 0) { pkg_free(buf); return -1; } c->uri.s = buf; c->uri.len = len; return 1; } /****************************** End of Added Section 2 ************************/ /**** Added Section 3: Function to extract the host part only of the Contact SIP URI *******/ static int extract_sip_contact_host(struct sip_msg* msg, char* hostname) { struct sip_uri uri; contact_t* c; if (get_contact_uri(msg,&uri,&c)==-1) return -1; *hostname = uri.host.s; return 1; } /*********************** End of Added Section 3 ******************************/ /************ Added Section 4: Function to extract the IP @ in 'o' SDP parameter **********/ static int extract_sdp_ownerip(str *body, str *ownerip, int *pf) { char *cp, *cp1; int len, nextisip; cp1 = NULL; for (cp = body->s; (len = body->s + body->len - cp) > 0;) { cp1 = ser_memmem(cp, "o=", len, 2); if (cp1 == NULL || cp1[-1] == '\n' || cp1[-1] == '\r') break; cp = cp1 + 2; } if (cp1 == NULL) { LOG(L_DBG, "ERROR: extract_mediaownerip: no `o=' in SDP\n"); return -1; } ownerip->s = cp1 + 2; ownerip->len = eat_line(ownerip->s, body->s + body->len - ownerip->s) - ownerip->s; trim_len(ownerip->len, ownerip->s, *ownerip); nextisip = 0; for (cp = ownerip->s; cp < ownerip->s + ownerip->len;) { len = eat_token_end(cp, ownerip->s + ownerip->len) - cp; if (nextisip == 1) { ownerip->s = cp; ownerip->len = len; nextisip++; break; } if (len == 3 && memcmp(cp, "IP", 2) == 0) { switch (cp[2]) { case '4': nextisip = 1; *pf = AF_INET; break; case '6': nextisip = 1; *pf = AF_INET6; break; default: break; } } cp = eat_space_end(cp + len, ownerip->s + ownerip->len); } if (nextisip != 2 || ownerip->len == 0) { LOG(L_ERR, "ERROR: extract_mediaownerip: " "no `IP[4|6]' in `c=' field\n"); return -1; } return 1; } /************************** End of Added Section 4 ****************************/ /**** Added Section 5: Function to alter the IP information in the 'o' SDP parameter ********/ static int alter_sdp_ownerip(struct sip_msg *msg, str *body, str *oldip, int oldpf, str *newip, int newpf, int preserve) { char *buf; int offset; struct lump* anchor; str omip, nip, oip; /* check that updating mediaip is really necessary */ if (oldpf == newpf && isnulladdr(oldip, oldpf)) return 0; if (newip->len == oldip->len && memcmp(newip->s, oldip->s, newip->len) == 0) return 0; if (preserve != 0) { anchor = anchor_lump(msg, body->s + body->len - msg->buf, 0, 0); if (anchor == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: anchor_lump failed\n"); return -1; } if (oldpf == AF_INET6) { omip.s = AOLDMEDIP6; omip.len = AOLDMEDIP6_LEN; } else { omip.s = AOLDMEDIP; omip.len = AOLDMEDIP_LEN; } buf = pkg_malloc(omip.len + oldip->len + CRLF_LEN); if (buf == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: out of memory\n"); return -1; } memcpy(buf, omip.s, omip.len); memcpy(buf + omip.len, oldip->s, oldip->len); memcpy(buf + omip.len + oldip->len, CRLF, CRLF_LEN); if (insert_new_lump_after(anchor, buf, omip.len + oldip->len + CRLF_LEN, 0) == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: insert_new_lump_after failed\n"); pkg_free(buf); return -1; } } if (oldpf == newpf) { nip.len = newip->len; nip.s = pkg_malloc(nip.len); if (nip.s == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: out of memory\n"); return -1; } memcpy(nip.s, newip->s, newip->len); } else { nip.len = newip->len + 2; nip.s = pkg_malloc(nip.len); if (nip.s == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: out of memory\n"); return -1; } memcpy(nip.s + 2, newip->s, newip->len); nip.s[0] = (newpf == AF_INET6) ? '6' : '4'; nip.s[1] = ' '; } oip = *oldip; if (oldpf != newpf) { do { oip.s--; oip.len++; } while (*oip.s != '6' && *oip.s != '4'); } offset = oip.s - msg->buf; anchor = del_lump(msg, offset, oip.len, 0); if (anchor == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: del_lump failed\n"); pkg_free(nip.s); return -1; } if (insert_new_lump_after(anchor, nip.s, nip.len, 0) == 0) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: insert_new_lump_after failed\n"); pkg_free(nip.s); return -1; } return 0; } /***************** End of Added Section 5 ******************************/ /***********Added Section 6: added section in force_rtpproxy_2****************/ if (extract_sdp_ownerip(&body, &oldownerip, &ownerpf) == -1) { LOG(L_ERR, "ERROR: force_rtp_proxy2_2: can't extract media owner IP " "from the message\n"); return -1; } if (asymmetric != 0 || real != 0) { newownerip = oldownerip; } else { newownerip.s = ip_addr2a(&msg->rcv.src_ip); newownerip.len = strlen(newownerip.s); } body2.s = oldownerip.s + oldownerip.len; body2.len = body.s + body.len - body2.s; if (extract_sdp_ownerip(&body2, &oldownerip1, &ownerpf1) == -1) { oldownerip1.len = 0; } if (oldownerip1.len > 0 && ownerpf != ownerpf1) { LOG(L_ERR, "ERROR: force_rtp_proxy2_2: mismatching address " "families in SDP\n"); return -1; } /********************* End of Added Section 6********************/ /*********** Added Section 7: added section in force_rtpproxy_2****************/ if (alter_sdp_ownerip(msg, &body, &oldownerip, ownerpf, &newip, pf1, 0) == -1) return -1; if (oldownerip1.len > 0 && alter_sdp_ownerip(msg, &body2, &oldownerip1, ownerpf, &newip, pf1, 0) == -1) return -1; /*********** End of Added Section 7************/ I apologize for this email's length, but I'm trying to be as detailed as I can, and I really do need the help :-) Thank you again Ahmed _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] load testing mediaproxy for more than 50 calls
by Simone Cittadini 31 Aug '05

31 Aug '05

I need to test the new ser/radius/mediaproxy server we have bought. As a fast upper bound to the maximum number of concurrent calls I tought to use rtpgenerator, but it can't increase the number of simultaneous calls more than 50 ... Is there any way to increase that limit ? (why it is set anyway ?) Launching a second instance raises cpu usage from 0,5 to 98, so it doesn't seems a good solution ...

1 0

[Serusers] FIFO not running or bad path to it
by Erdem HAKİ 31 Aug '05

31 Aug '05

Hi, I have also this problem when I logged in, I'm getting an error on the first page as "FIFO not running or bad path to it" and can't still resolve. Any idea? Thanks Erdem HAKI ---------------------------------------------------------------- Read the INSTALL file!! Requirements: ------------- - PEAR package LOG (optional - not included in PHP realase) see http://pear.php.net/manual/en/installation.getting.php for details Karel Marc Khayat napsal(a): > Anybody got the Log.php file from serweb (ser-0.9.3), coz it seems like > it's missing from my package. > > Thx, > > Marc > > -----Original Message----- > From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] On > Behalf Of Marc Khayat > Sent: Friday, August 19, 2005 11:25 AM > To: Karel Kozlik > Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > Subject: RE: [Serusers] Serweb > > /tmp/ser_fifo/ is wrong since it'll be looking for a folder... > > Marc > > -----Original Message----- > From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] On > Behalf Of Marc Khayat > Sent: Friday, August 19, 2005 11:03 AM > To: Karel Kozlik > Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > Subject: RE: [Serusers] Serweb > > Hi Karel, > Thx for bearing with me :) > In config_paths.php, I changed it from "/tmp/ser_fifo" to > "tmp/ser_fifo/" > The errors at the top disappeared but I'm still getting the "FIFO not > running or bad path to it" message. > > Is it wrong to put "tmp/ser_fifo/"? knowing that in ser.cfg, it's > /tmp/ser_fifo > > Thanks again, > > Marc > > -----Original Message----- > From: Karel Kozlik [mailto:karel at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] > Sent: Friday, August 19, 2005 9:28 AM > To: Marc Khayat > Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > Subject: Re: [Serusers] Serweb > > sorry Marc, haven't idea what may be wrong. Please check again setting > of fifo in serweb. Check variable $config->fifo_server in > config_paths.php. It's value must be '/tmp/ser_fifo'. > > Try enable logging in serweb ($config->enable_loging = true in > config.php), set log level to debug ($config->log_level = > "PEAR_LOG_DEBUG") and examine the log file if there are some messages > refered fifo. > > Also try search for string 'fifo_server' throught log files (or whole > serweb) and check if it's value isn't changed somewhere. You should find > > this string only in file config_paths.php (one times) and in file > functions.php (four times). > > Karel > > Marc Khayat napsal(a): > >>Hi Karel, >>I already tried this one by manually changing the permissions on >>ser_fifo to 666; I just tried adding fifo_mode=0666 to ser.cfg, but >>still the same problem. >> >>Marc >>-----Original Message----- >>From: Karel Kozlik [mailto:karel at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>Sent: Friday, August 19, 2005 8:21 AM >>To: Marc Khayat >>Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>Subject: Re: [Serusers] Serweb >> >>Hi, >>try add: >>fifo_mode=0666 >> >>to your ser.cfg >> >>Karel >> >>Marc Khayat napsal(a): >> >> >>>/tmp/ser_fifo exists, and is pipe type >>>User running Apache is "apache". Owner of ser_fifo is User: "root", >>>Group: "root". >>>I joined user "apache" to group "root", same. >>> >>>I created group "ser" and joined "root" "apache" to that group, and >>>added fifo_group=ser to the ser.cfg, but still same problem: >>> >>>Warning: filetype(): Lstat failed for (null) (errno=13 - Permission >>>denied) in /var/www/html/serweb/html/functions.php on line 306 >>> >>>Warning: Cannot modify header information - headers already sent by >>>(output started at /var/www/html/serweb/html/functions.php:306) in >>>/var/www/html/serweb/html/page.php on line 35 >>> >>> >>>Thanks, >>> >>>Marc >>> >>>-----Original Message----- >>>From: Karel Kozlik [mailto:karel at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>>Sent: Thursday, August 18, 2005 1:38 PM >>>To: Marc Khayat >>>Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>Subject: Re: [Serusers] Serweb >>> >>>Hello, >>>please check if file /tmp/ser_fifo exists and if it's type is pipe. >> >>Also >> >> >>>check rights - if user under which apache is running have access to >> >>read >> >> >>>and write to this file. >>> >>>Karel >>> >>>Marc Khayat napsal(a): >>> >>> >>> >>>>The domain thing is working, but problem 2 is still bugging me. >>>>Help anyone? >>>> >>>>Marc >>>> >>>>-----Original Message----- >>>>From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>> >>>On >>> >>> >>> >>>>Behalf Of Marc Khayat >>>>Sent: Thursday, August 18, 2005 11:15 AM >>>>To: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>Subject: RE: [Serusers] Serweb >>>> >>>>Never mind guys... it's working! >>>> >>>>Marc >>>>-----Original Message----- >>>>From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>> >>>On >>> >>> >>> >>>>Behalf Of Marc Khayat >>>>Sent: Thursday, August 18, 2005 10:53 AM >>>>To: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>Subject: [Serusers] Serweb >>>> >>>>Hi all, >>>>I'm using SER-0.9.3 and its serweb version. >>>>I have 2 problems with it: >>>>1- On the login page, when I enter the username, it automatically >>>>completes it with the FQDN, while I need only the domain name; i.e. I >>>>login to http://ser.xyz.com, I enter the username aloha, it's >>> >>>completed >>> >>> >>> >>>>as aloha at ser.xyz.com <http://lists.iptel.org/mailman/listinfo/serusers> while I need it as aloha at xyz.com. <http://lists.iptel.org/mailman/listinfo/serusers> Currently, I'm >>>>writing in manually >>>> >>>>2- When I login, I'm getting an error on the first page only as "FIFO >>>>not running or bad path to it". The path to it (in config_paths.php >>> >>>file >>> >>> >>> >>>>is /tmp/ser_fifo, which is correct), and when I enter the command >>> >>>serctl >>> >>> >>> >>>>monitor, the counter isn't constant, and same path. >>>> >>>>Can anyone help me plz, >>>> >>>>Thanks, >>>> >>>>Marc >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >> > > > _______________________________________________ > Serusers mailing list > Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > Serusers mailing list > Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Serusers] offline v unknown
by Kevin Brennan 31 Aug '05

31 Aug '05

Does anybody have any tips on solving the following I want to differenciate between offline users and unknown users. When you use lookup("location") there is no differenciation between the two. Does this require uri_db ? Using 0.8.14 Br /Kevin

5 26

[Serusers] Hairpin solutions?
by Federico Giannici 31 Aug '05

31 Aug '05

Considering that we DON'T want to use an RTP Proxy (bad scalability, waste of bandwidth and longer delays), is there any solution to the "hairpin" problems? Are there any "boxes" that look at the SIP messages and short-circuit calls from and to the local network? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

7 10

[Users] nat flag and branch routes
by Klaus Darilion 31 Aug '05

31 Aug '05

Hi! The nat flag which will be loaded during lookup() - will it be set for each branch so it can be queried in branch_route for all of the registered contacts, or should I use isdsturiset() instead of testing for the NAT flag? regards klaus

5 10

[Serusers] Cancel'ing filure_route
by Dmitry Isakbayev 31 Aug '05

31 Aug '05

Hello, I need help understanding how to handle "CANCEL" messages. Here is the case - 1 Proxy rebuilds URI, sets 'failure_route', and executes 't_relay' 2 't_realy' times out and the proxy executes 'failure_route' after the fr_timer hits If the OUI cancels the call between steps 1 and 2, the proxy still executes the 'failure_route' after the fr_timer hits. Can I cancel any "t_on_failure" logic after receiving a CANCEL message? Or, while in in the 'failre_rout' block, can I figure out that there was a CANCEL message? Thank You, Dmitry

2 3

[Users] branchroute problem
by Klaus Darilion 31 Aug '05

31 Aug '05

Hi! In the following scenario, a call is forked to 3 phones. 2 of them are behind the same NAT (same publich IP, different port, one uses STUN, one not). I xlog all branches,dest-sets... in route[] and then in branch_route. But the branch route is executed on 2 times, instead of three times (the destination set shows that there are 3 destinations) Although the destination set is changed in the second branch processing. Is this the intended way? regards klaus route[]: request's first branch: $br=sip:10.10.0.50:15360 request's all branches: $bR=sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 request's destination set: $ds=Contact: sip:klaus@83.136.33.19:5060, sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 request's destination uri: $du= request's uri: $ru=sip:klaus@83.136.33.19:5060 activating t_on_branch(1) branch_route[1]: first branch processing branch_route[1]: request's first branch: $br=sip:10.10.0.50:15360 branch_route[1]: request's all branches: $bR=sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination set: $ds=Contact: sip:klaus@83.136.33.19:5060, sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination uri: $du= branch_route[1]: request's uri: $ru=sip:klaus@83.136.33.19:5060 branch_route[1]: second branch processing branch_route[1]: request's first branch: $br=sip:10.10.0.50:15360 branch_route[1]: request's all branches: $bR=sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination set: $ds=Contact: sip:10.10.0.50:15360, sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination uri: $du=sip:83.136.33.3:3003 branch_route[1]: request's uri: $ru=sip:10.10.0.50:15360 ...no third branch procesing :-(

2 5

[Serusers] REGISTER with negative Content-Length
by Hendrik Scholz 31 Aug '05

31 Aug '05

Hi! I've handcrafted a REGISTER that contains a negative Content-Length: from ngrep: # U x.x.x.x:5060 -> y.y.y.y:5060 REGISTER sip:123@mydomain.net SIP/2.0. Via: SIP/2.0/UDP x.x.x.x:5060. From: 123 <sip:123@mydomain.net:5060>;tag=33. To: 123 <sip:123@mydomain.net:5060>. Call-ID: 33-11277(a)x.x.x.x. CSeq: 1 REGISTER. Contact: sip:123@x.x.x.x:5060. Content-Length: -1000. Expires: 300. syslog content: Aug 31 11:04:48 xxx ser[3866]: ERROR:parse_content_length: parse error near char [45][-] Aug 31 11:04:48 xxx ser[3866]: ERROR:get_hdr_field: bad content_length header Aug 31 11:04:48 xxx ser[3866]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed Aug 31 11:04:48 xxx ser[3866]: pre_auth(): Error while sending 400 reply So the parser works fine but I cannot send an error message to prevent retransmissions from the UAC. The same is true for alphanumeric characters within the Content-Length: value. To test the Content-Length stuff even more I've set the Content-Length within the REGISTER to 1000 which yields a 'ERROR: t_newtran: EoH not parsed' warning but SER still sends the 407 that I was hoping for. Now the question is if there is any way to add something like a failure_route block to allow logging and sending a generic or customized 4xx/5xx error message to prevent retransmissions. Btw: My config is far from a stock one but it all comes down to a standard REGISTER with radius_www_authorize() in the REGISTER block. Any help appreciated, Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

1 0

RE: [Serusers] Bye messages not logged
by Marc Khayat 31 Aug '05

31 Aug '05

Anyone can help me with the BYE messages that are not being logged? Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marc Khayat Sent: Tuesday, August 30, 2005 12:26 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Bye messages not logged Hi all, I upgraded my system from ser-0.8.14 to ser-0.9.3, and updated my ser.cfg. However, with the new thing, I'm getting INVITE and ACK messages, but no the BYE nor the REGISTER messages. I made no changes to the config file, only added the accounting requirements. But is the below stateful? Here's my ser.cfg: loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc", "log_flag", 1 ) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 5) modparam("acc", "db_missed_flag", 6) # 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 within 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; }; setflag(5); setflag(6); 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=="INVITE") { if (uri=~"^sip:9") { t_relay_to_udp("50.65.11.23", "5060"); break; }; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("mydomain.net", "subscriber")) { # www_challenge("mydomain.net", "0"); # break; # }; save("location"); 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(); }; } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] AVPops trouble
by Martin Kjeldsen 31 Aug '05

31 Aug '05

Hi I've been fiddeling with the AVPops module for some time now and havn't really gotten it to work as intended. I was hoping that someone out there might be able to explain the basics of the module, cause apparently I'm misunderstanding it... I think :-) I having trouble with pulling data from my openser DB with a scheme, and then pushing (pushto) it into a request header. The thing is I cant quite figure out, if I need to store the data I get from my scheme or if I can use it directly in a pushto! I have tried using a scheme: modparam("avpops", "db_scheme", "scheme0:uuid_col=contact;value_col=username;value_type=string;table=aliases") and then loading and pushing it with: avp_db_load("$from","s:pai/$scheme0"); avp_pushto("$P-Asserted-Identity","s:pai"); But this leaves me with no new field in the header, and nothing to show for in my MySQL log! I'm using these additional params for the AVPops: modparam("avpops","avp_url","mysql://openser:heslo@localhost/openser") modparam("avpops","avp_table","avptable") modparam("avpops","use_domain",1) The avptable is one I've created, since I wasn't sure if it needed to be there or not. Any and all help will be greatly appreciated, cause I'm really getting stuck here. Regards Martin

2 2

[Serusers] caller id with name
by Tulika Pradhan 31 Aug '05

31 Aug '05

hi ! has anyone implemented the caller id with name feature on ser ? regards, tulika

1 0

[Serusers] SER's nathelper
by Frank Kostin 31 Aug '05

31 Aug '05

Hi everybody, I have two devices behind a NAT and with different codecs that can not agree upon a common codec and can not make it working. Anyone have a sample config, using SER's nathelper in conjunction with Asterisk to rewrite the SDP so that the private IP addresses are not included in the re-invite. Thanks Frank --------------------------------- Yahoo! Mail Stay connected, organized, and protected. Take the tour

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[Serusers] serweb msilo ser problem
by Walter Willis 31 Aug '05

31 Aug '05

i am problem with msilo and serweb :) i have installed the ser (cvs) and serweb, send IM put any thing in sip address of recipient: and send the message and view: sadasd no exist. Message has been successfully sended to address sip:sadasd@domain.com and debug the SER: 3(9953) Warning: run_failure_handlers: no UAC support (0, 0) the scrit ser: route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } wath is the problem????

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[Serusers] OT: Monitoring tools
by Daniel Corbe 31 Aug '05

31 Aug '05

Hello, I'm going to attempt to build a box that listens on a SPAN switch port which has all my proxy traffic mirrored to it. I was curious if anyone has ever attempted something like this before, and what software/tools were used to listen for signaling (SIP) traffic and subsequently log the traffic out to disk. -Daniel

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Re: [Serusers] NAT example
by Kamran Ahmad 30 Aug '05

30 Aug '05

hello Why uri!=myself what could be the reason using "SER 0.9.3 source package for Getting Started" from Onsip.org when i try to use port 5060 then it only work for public ip UA and dont work for nated users. But when i change port to 5069 it try to listen for nated UA but the problem is still why uri!=myself i am using and for configuration ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg after lot of changes in ports exec_msg("cat >> /tmp/call.log"); if (uri!=myself) { xlog("L_ERR","uri!=myself"); route(1); break; }; i found this /var/log/messages -------------------------------------------------- Aug 30 21:58:09 SIP ser[10560]: uri!=myself and /tmp/call.log -------------------------------------------------- REGISTER sip:SERPUBIP SIP/2.0 Content-Length: 0 Contact: <sip:8000@192.168.0.21:5060>;events="message-summary" Call-ID: B692E8F0-DDC4-42A7-8A87-79B8133FA79B(a)192.168.0.21 Max-Forwards: 16 From: <sip:8000@SERPUBIP>;tag=587922581424 CSeq: 5 REGISTER To: <sip:8000@SERPUBIP> Via: SIP/2.0/UDP 203.99.58.250:52904;rport;branch=z9h8001009User-Agent: SJLabs-SJphone/1.40.258 --- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Don't worry, > These messages comes from the startup process, when > SER tries to get the PTR DNS records of the IP > addresses present in the config file. These pointers > map IP addresses to domain names, instead of the > "standard" domain->IP mapping provided by the A|AAA > DNS records. > > Probably your SER instance is running, have you > tried ps?? > > Enjoy! > > Samuel. > > > Unclassified. > >>> Kamran Ahmad <p_kami(a)yahoo.com> 08/30/05 10:35AM > >>> > i tried this > > ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg > > > but it is not working > > /var/log/messages > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > > > --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > > > Try to Getting Started document from > > http://onsip.org/ You will find both a > > working config, as well as detailed explanation of > > the theory and actual > > commands. > > g-) > > > > Kamran Ahmad wrote: > > > Hello > > > > > > i am following this link for ser-0.9.3 but Nated > > > Clients > > > are not working. Can any one please tell > > me > > > what is the problem in my configuration. > > > > > > > > > http://www.voip-info.org/tiki-index.php?page=SER+nat+support > > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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Re: [Users] #DEFINE equivalent?
by Kanakatti Mahesh Subramanya 30 Aug '05

30 Aug '05

because i didn't think of it? :-) cheers, and thanx Bogdan-Andrei Iancu wrote: > Kanakatti Mahesh Subramanya wrote: > >> I totally agree. >> >> Personally, I've been ripping out large chunks of my config file, and >> replacing it with varioud avpops based stuff - it may be *slightly* >> more convoluted, but it is completely database configurable, and >> hence easier to maintain >> >> re: rr, its actually neither. Its a static, which i'm using to >> identify NAT'ed clients to account for re-INVITEs > > > why don't you use the append_rr_param() function to added the NAT > identifier to the RR hdr? do a normal record_route() and if nat is > detected, add the param > > regards, > bogdan > >> >> cheers >> >> Bogdan-Andrei Iancu wrote: >> >>> Hi Kanakatti, >>> >>> I'm for m4'ing (for the moment) for two reasons: >>> 1) since we can do it with m4, I prefer to use the cycles for >>> something still missing; >>> 2) using dynamic parameters (like via AVPs) reduce the >>> performance - time to look for the AVP and most critical the >>> impossibility of calling fixup functions at startup (which will >>> force to fix params at runtime, for each processing). >>> >>> regarding the record_route_preset() - do you use dynamic parameters >>> or dynamic rr URI? >>> >>> regards, >>> bogdan >> >>

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[Serusers] Reviewers needed for Open Solaris SIP API design review
by Rao Shoaib 30 Aug '05

30 Aug '05

Dear Colleagues, We are conducting a community design review for the open solaris SIP API at http://www.opensolaris.org/jive/thread.jspa?threadID=1936&tstart=0 We are very interested in hearing your comments on the design. Please post your comments at the above location. Thanks. Rao.

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[Serusers] supporting multiple calls from one user
by Iqbal 30 Aug '05

30 Aug '05

Hi Does ser support dialing to multiple destinations using one username/password, especially for setups where hardware may have multiple voice channels, but only space for one username. Anyone have any test setups Iqbal

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[Serusers] SER outgoing through TCP?
by Pat Wang 30 Aug '05

30 Aug '05

Hi, Does anyone know how to configure SER proxy signalling through TCP? For example, UAC1 -------> SER -------> UAC2, I would like to have SER proxy the INVITE (and the following) message to UAC2 using TCP. Is that possible and how to setup on ser.cfg? Or if it possible to do TCP signalling between two SERs: SER ------TCP---------SER? Thanks a million, Pat

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[Serusers] Routing - Looking up database table for subscriber accounts not registered at the moment
by Frank Fischer 30 Aug '05

30 Aug '05

Hi all i just started working with SIP and i'm currently trying to implement the routing logic as needed in my setup. Doing this i came across a requirement i couldn't find a solution for. Maybe someone of you may help me? I have the following situation: I have users with (only) e164Numbers as usernames in my SIP domain and i have a PSTN Gateway behind which i may reach also e164Numbers. Now, i have a subscriber base in my domain where not all of the subscribers with a e164Number may be registered at a certain moment (but still have a valid account and phonenumber). When i get a call at such a moment, i will try to look up the callee username in the location database. If i find one, fine. If not, there are two possibilities left - first, it could be a user only reachable in the pstn, so route to the pstn gateway or second it might be a user in my domain with a valid account but not registered at the moment, so reject or redirect to a voicemail or what ever. If I would work with alphanumerical numbers only within my SIP cloud, i knew how this could be done - all numerical numbers automatically would be PSTN numbers. But how can this be done in my setup? Thanks a lot for your help and kind Regards Frank

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[Serusers] ACK
by Sebastian Kühner 30 Aug '05

30 Aug '05

Hi! I noticed a "phenomenon" that I don't understand: I make accounting with ser. If all works fine, ser puts into the database: sip_method: ACK i_uri: sip:0182011542614xxxxxx@xxx.xxx.xxx.xxx:5060 (xxx.xxx.xxx.xxx = IP of PSTN-Gateway) ... and after the call, the BYE message reaches my server. But sometimes, the ACK is logged in the database like this (in the most cases form users "far away"): sip_method: ACK i_uri: sip:0182011542614xxxxxx@my-ip:5060 In this case, the BYE message doesn't reach my server. Does anybody know what could be the reason and why ser logs the ACK from himself? Many thanks! Sebastian

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Re: [Serusers] Radius Server sending reply to another machine!!!
by Ashutosh Kumar 30 Aug '05

30 Aug '05

Many thanks Jan, Adding bindaddr directive to radiusclient.conf worked!!!!. Now, the problem remains as to why is the radiusclient/ser picking up the dictionary.ascend, when I have included the dictionary.ser at the end of my dictionary file. Also, I have auth_radius:service_type modparamed' to value of 15. Please guide. Regards, Ashutosh Kumar -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, August 30, 2005 2:40 PM To: Ashutosh Kumar Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Radius Server sending reply to another machine!!! Check the configuration file of the radius client library, perhaps you have set the bindaddr parameter to the wrong IP address ? Jan. On 30-08-2005 03:57, Ashutosh Kumar wrote: > Hi, > I am facing a strange bewildering problem out here. Earlier I was > running ser -0.9 on machine a.b.c.116. Now we shifted it to a.b.c.113 for > testing for radius testing purposes. > But in the radius packet dump, it is showing as the radius packet still > originating from a.b.c.116 machine, despite the fact, my new revised ser.cfg > doesn't contain any listen address or alias for .116 box. > Also, the /etc/hosts doesn't mention .116 address. Also, it is not the case > that .116 and .113 are IPs homes on same machine. My ser.cfg code for radius > authorization is as follows. > > ... > ..... > > if (!radius_www_authorize("")) { > # if (!www_authorize("","subscriber")) { > www_challenge("","0"); > break; > }; > ... > ..... > > In the DB also, there is no table . The Domains table has no records. But > still SER is usng .116 address to send radius packets from. Here is the > radius packet dump. > > [root@sip radiusclient]# tcpdump -i eth1 -s 1500 -w - udp | raddump > -vvvvvvvvvvvvvvvvvvvvvsn > tcpdump: listening on eth1 > Looks like a(n) libpcap/tcpdump packet trace > 17:45:13.051205 radius a.b.c.113:50912 > p.q.r.s:1812 Access-Request 180 > User-Name = test(a)a.b.c.112 > Ascend-PW-Warntime = 168195173 > Ascend-PW-Warntime = 17774137 > Ascend-PW-Warntime = 36320307 > Ascend-PW-Warntime = 68383593 > Ascend-PW-Warntime = 51008069 > Ascend-Menu-Item = 75dd216d3efa9bfb9c2d7e013d54f7bb > Service-Type = 15 (unknown) > Ascend-PW-Lifetime = 1952805748 > NAS-IP-Address = a.b.c.116 > NAS-Port = 5060 > > (Is there any table /file in ser other than ser.cfg or mysql://ser that > contains the domain information that might have been lying there unchanged > since machine migration?) > > Also, why is SER using incorrect SER dictionary NAS client (Ascend)( > shouldn't it have been Cisco or sometinhg like that). I have included the > dictionary.ser included in ser-0.9 source in the radiusclient conf file. > > Help me, its driving me crazy. > > > Regards, > Ashutosh Kumar > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] CANCEL problem
by Klaus Darilion 30 Aug '05

30 Aug '05

Hi! I just having problems cancelling forked calls. openser forwards the incoming CANCEL to both phones. As you see, openser added 2 via headers to the last CANCEL. The via header of branch 1 and branch 2. Maybe something got wrong when introducing branch routes? I'm using todays CVS. klaus 83.136.33.3:1200 -> 83.136.32.83:5060 CANCEL sip:klaus@enum.at SIP/2.0 Via: SIP/2.0/UDP 10.10.0.76:8418 Max-Forwards: 70 From: "klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq: 2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 # U 83.136.32.83:5060 -> 83.136.33.19:5060 CANCEL sip:klaus@83.136.33.19:5060 SIP/2.0 Record-Route: <sip:83.136.32.83;ftag=b3669e387d784f1dbd080f83fdd62306;nat=yes;lr=on> Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.0 Via: SIP/2.0/UDP 10.10.0.76:8418;rport=1200;received=83.136.33.3 Max-Forwards: 16 From:"klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq: 2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 P-Behind-NAT: Yes # U 83.136.32.83:5060 -> 83.136.33.3:8718 CANCEL sip:klaus@10.10.0.50:8718 SIP/2.0 Record-Route: <sip:83.136.32.83;ftag=b3669e387d784f1dbd080f83fdd62306;nat=yes;lr=on> Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.0 Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.1 Via: SIP/2.0/UDP 10.10.0.76:8418;rport=1200;received=83.136.33.3;rport=1200;received=83.136.33.3 Max-Forwards: 16 From: "klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq:2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 P-Behind-NAT: Yes

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[Serusers] 1 SER & 2 Providers
by Ozan Blotter 30 Aug '05

30 Aug '05

Dear List, Assume that there are two different service providers and one SER in middle of them. A client needs to use/connect SER but also he/she has accounts in both providers and both supplied him/her DDIs (landline phone). Now again assume that client registers both of phones into SER. But still he/she wants to receive calls from both providers, is it possible to receive calls from different providers to that client and/or how to i setup ? In current SER.CFG below, i only setup one prefix which is "0", but i can change it to any and/or add many prefixes, ex: 0049 (germany), 001 (usa), 0044 (uk), 0090 (turkey) to Provider 1 and 0033 (france), 0038 (italy) to Provider 2 and so on. debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX # This is SER's Public IP port=5060 children=4 alias=212.XXX.104.XXX # This is SER's Public IP 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"); # This is SER's Public IP } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { # I do not want to make accounting if clients' prefix is beginning with 833 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]*@*") { # If calling number is beginning with prefix 0, route it to PSTN rewritehost("195.XXX.122.XXX"); # This is PSTN Gateway's Public IP 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(); }; }

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[Serusers] clarification about authentication
by Iqbal 30 Aug '05

30 Aug '05

Hi www_authorize , will this always fail, ? because it seems to and then a www.challenge is sent. In www.challenge, is it okay to leave the realm blank, as I have done, or is it better practice to have the same realm as that in www_authorise("", "1"); check_to should only be run if auth is correct above , is that correct? Why run check_to at all, ? and then u have proxy_auth and proxy_challenge, and is_from_local and is_uri_local, surely all these dont need to be used as checks in INVITE or do they. Iqbal

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[Serusers] Bye messages not logged
by Marc Khayat 30 Aug '05

30 Aug '05

Hi all, I upgraded my system from ser-0.8.14 to ser-0.9.3, and updated my ser.cfg. However, with the new thing, I'm getting INVITE and ACK messages, but no the BYE nor the REGISTER messages. I made no changes to the config file, only added the accounting requirements. But is the below stateful? Here's my ser.cfg: loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc", "log_flag", 1 ) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 5) modparam("acc", "db_missed_flag", 6) # 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 within 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; }; setflag(5); setflag(6); 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=="INVITE") { if (uri=~"^sip:9") { t_relay_to_udp("50.65.11.23", "5060"); break; }; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("mydomain.net", "subscriber")) { # www_challenge("mydomain.net", "0"); # break; # }; save("location"); 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(); }; }

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Re: [Serusers] NAT example
by Kamran Ahmad 30 Aug '05

30 Aug '05

Thanks for reply [root@SIP ~]# ps -A | grep ser 131 ? 00:00:00 kseriod when i try to use asterisk on same port it is working. means port is free. it tried to change port from5060 to 5066. i have used killall ser before starting ser i tried to put in log into route2 but request is not reaching route2 route[2] { xlog("L_ERR","inside route2"); --- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Don't worry, > These messages comes from the startup process, when > SER tries to get the PTR DNS records of the IP > addresses present in the config file. These pointers > map IP addresses to domain names, instead of the > "standard" domain->IP mapping provided by the A|AAA > DNS records. > > Probably your SER instance is running, have you > tried ps?? > > Enjoy! > > Samuel. > > > Unclassified. > >>> Kamran Ahmad <p_kami(a)yahoo.com> 08/30/05 10:35AM > >>> > i tried this > > ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg > > > but it is not working > > /var/log/messages > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > > > --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > > > Try to Getting Started document from > > http://onsip.org/ You will find both a > > working config, as well as detailed explanation of > > the theory and actual > > commands. > > g-) > > > > Kamran Ahmad wrote: > > > Hello > > > > > > i am following this link for ser-0.9.3 but Nated > > > Clients > > > are not working. Can any one please tell > > me > > > what is the problem in my configuration. > > > > > > > > > http://www.voip-info.org/tiki-index.php?page=SER+nat+support > > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Different Path for Request and Reply
by Andreas Granig 30 Aug '05

30 Aug '05

Hi, Let's say there are two SER servers A and B, a UAC and a UAS. If a request from UAC to UAS is routed via A and right after that A fails and B takes over, which impacts does it have if 1xx/200 replies from UAS to UAC are routed via B? The replies could theoretically be routed correctly because the next hop is available in the Via header. But since there is no transaction stored in SER for this, does this work? And what about accounting? What other functionality would be affected? Andy

2 1

[Serusers] Radius Server sending reply to another machine!!!
by Ashutosh Kumar 30 Aug '05

30 Aug '05

Hi, I am facing a strange bewildering problem out here. Earlier I was running ser -0.9 on machine a.b.c.116. Now we shifted it to a.b.c.113 for testing for radius testing purposes. But in the radius packet dump, it is showing as the radius packet still originating from a.b.c.116 machine, despite the fact, my new revised ser.cfg doesn't contain any listen address or alias for .116 box. Also, the /etc/hosts doesn't mention .116 address. Also, it is not the case that .116 and .113 are IPs homes on same machine. My ser.cfg code for radius authorization is as follows. ... ..... if (!radius_www_authorize("")) { # if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; ... ..... In the DB also, there is no table . The Domains table has no records. But still SER is usng .116 address to send radius packets from. Here is the radius packet dump. [root@sip radiusclient]# tcpdump -i eth1 -s 1500 -w - udp | raddump -vvvvvvvvvvvvvvvvvvvvvsn tcpdump: listening on eth1 Looks like a(n) libpcap/tcpdump packet trace 17:45:13.051205 radius a.b.c.113:50912 > p.q.r.s:1812 Access-Request 180 User-Name = test(a)a.b.c.112 Ascend-PW-Warntime = 168195173 Ascend-PW-Warntime = 17774137 Ascend-PW-Warntime = 36320307 Ascend-PW-Warntime = 68383593 Ascend-PW-Warntime = 51008069 Ascend-Menu-Item = 75dd216d3efa9bfb9c2d7e013d54f7bb Service-Type = 15 (unknown) Ascend-PW-Lifetime = 1952805748 NAS-IP-Address = a.b.c.116 NAS-Port = 5060 (Is there any table /file in ser other than ser.cfg or mysql://ser that contains the domain information that might have been lying there unchanged since machine migration?) Also, why is SER using incorrect SER dictionary NAS client (Ascend)( shouldn't it have been Cisco or sometinhg like that). I have included the dictionary.ser included in ser-0.9 source in the radiusclient conf file. Help me, its driving me crazy. Regards, Ashutosh Kumar

2 1

[Serusers] "Makefile.rules:80: action.d: No such file or directory" Errors
by Jaroslaw Gawron 30 Aug '05

30 Aug '05

Hello, I try to install ser-0.9.3 from source file on Mandriva 2005 limited edition and I get the following errors : Makefile.rules:80: action.d: No such file or directory Makefile.rules:80: crc.d: No such file or directory Makefile.rules:80: daemonize.d: No such file or directory Makefile.rules:80: data_lump.d: No such file or directory Makefile.rules:80: data_lump_rpl.d: No such file or directory Makefile.rules:80: dprint.d: No such file or directory Makefile.rules:80: dset.d: No such file or directory Makefile.rules:80: error.d: No such file or directory Makefile.rules:80: fifo_server.d: No such file or directory Makefile.rules:80: flags.d: No such file or directory Makefile.rules:80: forward.d: No such file or directory Makefile.rules:80: hash_func.d: No such file or directory Makefile.rules:80: ip_addr.d: No such file or directory Makefile.rules:80: main.d: No such file or directory Makefile.rules:80: md5.d: No such file or directory Makefile.rules:80: md5utils.d: No such file or directory Makefile.rules:80: modparam.d: No such file or directory Makefile.rules:80: msg_translator.d: No such file or directory Makefile.rules:80: pass_fd.d: No such file or directory Makefile.rules:80: proxy.d: No such file or directory Makefile.rules:80: qvalue.d: No such file or directory Makefile.rules:80: re.d: No such file or directory Makefile.rules:80: receive.d: No such file or directory Makefile.rules:80: resolve.d: No such file or directory Makefile.rules:80: route.d: No such file or directory Makefile.rules:80: route_struct.d: No such file or directory Makefile.rules:80: script_cb.d: No such file or directory Makefile.rules:80: socket_info.d: No such file or directory Makefile.rules:80: sr_module.d: No such file or directory Makefile.rules:80: stats.d: No such file or directory Makefile.rules:80: tcp_main.d: No such file or directory Makefile.rules:80: tcp_read.d: No such file or directory Makefile.rules:80: timer.d: No such file or directory Makefile.rules:80: tsend.d: No such file or directory Makefile.rules:80: udp_server.d: No such file or directory Makefile.rules:80: unixsock_server.d: No such file or directory Makefile.rules:80: usr_avp.d: No such file or directory Makefile.rules:80: mem/f_malloc.d: No such file or directory Makefile.rules:80: mem/mem.d: No such file or directory Makefile.rules:80: mem/memtest.d: No such file or directory Makefile.rules:80: mem/q_malloc.d: No such file or directory Makefile.rules:80: mem/shm_mem.d: No such file or directory Makefile.rules:80: mem/vq_malloc.d: No such file or directory Makefile.rules:80: parser/hf.d: No such file or directory Makefile.rules:80: parser/msg_parser.d: No such file or directory Makefile.rules:80: parser/parse_allow.d: No such file or directory Makefile.rules:80: parser/parse_content.d: No such file or directory Makefile.rules:80: parser/parse_cseq.d: No such file or directory Makefile.rules:80: parser/parse_disposition.d: No such file or directory Makefile.rules:80: parser/parse_diversion.d: No such file or directory Makefile.rules:80: parser/parse_event.d: No such file or directory Makefile.rules:80: parser/parse_expires.d: No such file or directory Makefile.rules:80: parser/parse_fline.d: No such file or directory Makefile.rules:80: parser/parse_from.d: No such file or directory Makefile.rules:80: parser/parse_hname2.d: No such file or directory Makefile.rules:80: parser/parse_hostport.d: No such file or directory Makefile.rules:80: parser/parse_methods.d: No such file or directory Makefile.rules:80: parser/parse_nameaddr.d: No such file or directory Makefile.rules:80: parser/parse_param.d: No such file or directory Makefile.rules:80: parser/parser_f.d: No such file or directory Makefile.rules:80: parser/parse_rpid.d: No such file or directory Makefile.rules:80: parser/parse_rr.d: No such file or directory Makefile.rules:80: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:80: parser/parse_to.d: No such file or directory Makefile.rules:80: parser/parse_uri.d: No such file or directory Makefile.rules:80: parser/parse_via.d: No such file or directory Makefile.rules:80: parser/digest/digest.d: No such file or directory Makefile.rules:80: parser/digest/digest_parser.d: No such file or directory Makefile.rules:80: parser/digest/param_parser.d: No such file or directory Makefile.rules:80: parser/contact/contact.d: No such file or directory Makefile.rules:80: parser/contact/parse_contact.d: No such file or directory Makefile.rules:80: db/db.d: No such file or directory Makefile.rules:80: db/db_fifo.d: No such file or directory Makefile.rules:80: lex.yy.d: No such file or directory Does anyone know how to fix this problem ? Any help will be great ! Best regards, Jarosław Gawron

2 1

Re: [Serusers] NAT example
by Samuel Osorio Calvo 30 Aug '05

30 Aug '05

Don't worry, These messages comes from the startup process, when SER tries to get the PTR DNS records of the IP addresses present in the config file. These pointers map IP addresses to domain names, instead of the "standard" domain->IP mapping provided by the A|AAA DNS records. Probably your SER instance is running, have you tried ps?? Enjoy! Samuel. Unclassified. >>> Kamran Ahmad <p_kami(a)yahoo.com> 08/30/05 10:35AM >>> i tried this ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg but it is not working /var/log/messages Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: could not rev. resolve ip Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: could not rev. resolve ip --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > Try to Getting Started document from > http://onsip.org/ You will find both a > working config, as well as detailed explanation of > the theory and actual > commands. > g-) > > Kamran Ahmad wrote: > > Hello > > > > i am following this link for ser-0.9.3 but Nated > > Clients are not working. Can any one please tell > me > > what is the problem in my configuration. > > > > > http://www.voip-info.org/tiki-index.php?page=SER+nat+support > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] "Makefile.rules:80: action.d: No such file or directory"Errors
by Samuel Osorio Calvo 30 Aug '05

30 Aug '05

Don't worry, this files are created after the first compilation. Probably SER is already properly compiled in your host. Enjoy! Samuel. Unclassified. >>> "Jaroslaw Gawron" <j.gawron(a)wasko.pl> 08/30/05 10:20AM >>> Hello, I try to install ser-0.9.3 from source file on Mandriva 2005 limited edition and I get the following errors : Makefile.rules:80: action.d: No such file or directory Makefile.rules:80: crc.d: No such file or directory Makefile.rules:80: daemonize.d: No such file or directory Makefile.rules:80: data_lump.d: No such file or directory Makefile.rules:80: data_lump_rpl.d: No such file or directory Makefile.rules:80: dprint.d: No such file or directory Makefile.rules:80: dset.d: No such file or directory Makefile.rules:80: error.d: No such file or directory Makefile.rules:80: fifo_server.d: No such file or directory Makefile.rules:80: flags.d: No such file or directory Makefile.rules:80: forward.d: No such file or directory Makefile.rules:80: hash_func.d: No such file or directory Makefile.rules:80: ip_addr.d: No such file or directory Makefile.rules:80: main.d: No such file or directory Makefile.rules:80: md5.d: No such file or directory Makefile.rules:80: md5utils.d: No such file or directory Makefile.rules:80: modparam.d: No such file or directory Makefile.rules:80: msg_translator.d: No such file or directory Makefile.rules:80: pass_fd.d: No such file or directory Makefile.rules:80: proxy.d: No such file or directory Makefile.rules:80: qvalue.d: No such file or directory Makefile.rules:80: re.d: No such file or directory Makefile.rules:80: receive.d: No such file or directory Makefile.rules:80: resolve.d: No such file or directory Makefile.rules:80: route.d: No such file or directory Makefile.rules:80: route_struct.d: No such file or directory Makefile.rules:80: script_cb.d: No such file or directory Makefile.rules:80: socket_info.d: No such file or directory Makefile.rules:80: sr_module.d: No such file or directory Makefile.rules:80: stats.d: No such file or directory Makefile.rules:80: tcp_main.d: No such file or directory Makefile.rules:80: tcp_read.d: No such file or directory Makefile.rules:80: timer.d: No such file or directory Makefile.rules:80: tsend.d: No such file or directory Makefile.rules:80: udp_server.d: No such file or directory Makefile.rules:80: unixsock_server.d: No such file or directory Makefile.rules:80: usr_avp.d: No such file or directory Makefile.rules:80: mem/f_malloc.d: No such file or directory Makefile.rules:80: mem/mem.d: No such file or directory Makefile.rules:80: mem/memtest.d: No such file or directory Makefile.rules:80: mem/q_malloc.d: No such file or directory Makefile.rules:80: mem/shm_mem.d: No such file or directory Makefile.rules:80: mem/vq_malloc.d: No such file or directory Makefile.rules:80: parser/hf.d: No such file or directory Makefile.rules:80: parser/msg_parser.d: No such file or directory Makefile.rules:80: parser/parse_allow.d: No such file or directory Makefile.rules:80: parser/parse_content.d: No such file or directory Makefile.rules:80: parser/parse_cseq.d: No such file or directory Makefile.rules:80: parser/parse_disposition.d: No such file or directory Makefile.rules:80: parser/parse_diversion.d: No such file or directory Makefile.rules:80: parser/parse_event.d: No such file or directory Makefile.rules:80: parser/parse_expires.d: No such file or directory Makefile.rules:80: parser/parse_fline.d: No such file or directory Makefile.rules:80: parser/parse_from.d: No such file or directory Makefile.rules:80: parser/parse_hname2.d: No such file or directory Makefile.rules:80: parser/parse_hostport.d: No such file or directory Makefile.rules:80: parser/parse_methods.d: No such file or directory Makefile.rules:80: parser/parse_nameaddr.d: No such file or directory Makefile.rules:80: parser/parse_param.d: No such file or directory Makefile.rules:80: parser/parser_f.d: No such file or directory Makefile.rules:80: parser/parse_rpid.d: No such file or directory Makefile.rules:80: parser/parse_rr.d: No such file or directory Makefile.rules:80: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:80: parser/parse_to.d: No such file or directory Makefile.rules:80: parser/parse_uri.d: No such file or directory Makefile.rules:80: parser/parse_via.d: No such file or directory Makefile.rules:80: parser/digest/digest.d: No such file or directory Makefile.rules:80: parser/digest/digest_parser.d: No such file or directory Makefile.rules:80: parser/digest/param_parser.d: No such file or directory Makefile.rules:80: parser/contact/contact.d: No such file or directory Makefile.rules:80: parser/contact/parse_contact.d: No such file or directory Makefile.rules:80: db/db.d: No such file or directory Makefile.rules:80: db/db_fifo.d: No such file or directory Makefile.rules:80: lex.yy.d: No such file or directory Does anyone know how to fix this problem ? Any help will be great ! Best regards, Jaroslaw Gawron

1 0

[Serusers] NAT example
by Kamran Ahmad 30 Aug '05

30 Aug '05

Hello i am following this link for ser-0.9.3 but Nated Clients are not working. Can any one please tell me what is the problem in my configuration. http://www.voip-info.org/tiki-index.php?page=SER+nat+support ---------------------------------------------------------------- #ser.cfg # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 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=4 fifo="/tmp/ser_fifo" alias=mydomain.dyndns.org # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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 (len_gt( max_len )) { 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); 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=="REGISTER") { save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #inserted by klaus if (method=="INVITE") { record_route(); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #inserted by klaus # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { if (status=~"[12][0-9][0-9]") force_rtp_proxy(); } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

[Serusers] RTPProxy
by Anderson Alves de Albuquerque 29 Aug '05

29 Aug '05

Does someone have some doc (how-to, manual ...) about RTPProxy?

3 2

[Users] SER and SIP headers compact form
by Vincent Luba 29 Aug '05

29 Aug '05

Hi, Is there a way to force SER to use compact form when it sends SIP message? i.e. presence notification provided by "pa module" Regards, Luba Vincent

1 0

[Serusers] ser and database: using a view instead of a table for 'subscribers' table
by Arnaud Ligot 29 Aug '05

29 Aug '05

Hello, I already have a database with username, password (clear), firstname, lastname and I want to use ser with this database. I already have an administration interface for tis database so I want to keep it as is. I want to create a view for the 'subscribers' table based on my database. I'm looking for a good documentation about fields of 'subscribers' table. I also want to know if I can use ser without timestamp and encrypted fields of 'subscriber' table and if I have to create other tables in order to use ser only with auth_db. best regards, Arnaud Ligot

1 0

RE: [Serusers] hi,about mysql.so problem.
by Marc Khayat 29 Aug '05

29 Aug '05

- compile the mysql module, by doing: make modules=modules/mysql modules - copy mysql.so (found in modules/mysql) to /usr/lib/ser/modules/ Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of viking Sent: Monday, August 29, 2005 12:12 PM To: serusers(a)lists.iptel.org Subject: [Serusers] hi,about mysql.so problem. hi,everyone, I have installed ser-0.9.3-src on red-hat 9.0，i wanna to use mysql module,the problem is how can get the /usr/lib/ser/modules/mysql.so if i compile them from the src? thanks a lot! 　　　　　　　　yours viking 　　　　　　　　tieban(a)sina.com 　　　　　　　　　　2005-08-29 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] failure route problem
by Klaus Darilion 29 Aug '05

29 Aug '05

Hi! I have a problem with my failure route. In the failure route, I send the call to voicemail, except the call is CANCELED by the caller: failure_route[1] { if(t_check_status("487")) { xlog("L_INFO","caller cancelled call, no voicemail needed...\n"); exit; } route(4); # route to voicemail } The problem is, when a call is forked and one of the destinations is busy or has any other reasons to not accept the call. /-------- <-486 phone1 / caller ----------------- <-488 phone2 \ \-------- <-487 phone3 In the above case, phone1 and 2 immedately response with an error message. Then several seconds later, the caller hangs up and now phone3 sends 487. If I now check the status in the failure route it will be 486 (the lowest one). Also any other algorithm (the highest one, the first one, the last one) would be ambiguous. Thus, deciding if the call will be sent to voicemail or not is not possible using t_check_status. AFAIK there is no other method to check the status of the transaction in the failure route. IMO this is an issue which should be solved. I suggest setting the status explicitly to "487" if the transaction was cancelled by the caller, independent of the resonse status received from the phones. regards klaus

3 9

[Serusers] strange registration !?
by harry gaillac 29 Aug '05

29 Aug '05

Hello, Look at my ethereal file. When User agent send REGISTER message to ser this one try to register with itself so User agents receive authorization + "401 Unauthorized" !? What's wrong ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] hi,about mysql.so problem.
by viking 29 Aug '05

29 Aug '05

hi,everyone, I have installed ser-0.9.3-src on red-hat 9.0，i wanna to use mysql module,the problem is how can get the /usr/lib/ser/modules/mysql.so if i compile them from the src? thanks a lot! 　　　　　　　　yours viking 　　　　　　　　tieban(a)sina.com 　　　　　　　　　　2005-08-29

1 0

[Users] #DEFINE equivalent?
by Kanakatti Mahesh Subramanya 29 Aug '05

29 Aug '05

It would be useful if we could DEFINE parameters in the .cfg file e.g. define SOME_IP=1.2.3.4 and somewhere later in the file record_route_preset("SOME_IP;nat=yes"); Speaking as someone who has to manage openser.cfg instances across multiple servers (not to mention test environments!), it would certainly make my life a heck of a lot easier.... cheers

3 4

[Users] 1 OpenSER & 2 Providers
by Ozan Blotter 29 Aug '05

29 Aug '05

Dear List, Assume that there are two different service providers and one OpenSER in middle of them. A client needs to use/connect OpenSER but also he/she has accounts in both providers and both supplied him/her DDIs (landline phone). Now again assume that client registers both of phones into OpenSER. But still he/she wants to receive calls from both providers, is it possible to receive calls from different providers to that client and/or how to i setup ? In current SER.CFG below, i only setup one prefix which is "0", but i can change it to any and/or add many prefixes, ex: 0049 (germany), 001 (usa), 0044 (uk), 0090 (turkey) to Provider 1 and 0033 (france), 0038 (italy) to Provider 2 and so on. debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX # This is OpenSER's Public IP port=5060 children=4 alias=212.XXX.104.XXX # This is OpenSER's Public IP 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"); # This is OpenSER's Public IP } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { # I do not want to make accounting if clients' prefix is beginning with 833 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]*@*") { # If calling number is beginning with prefix 0, route it to PSTN rewritehost("195.XXX.122.XXX"); # This is PSTN Gateway's Public IP 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(); }; }

3 5

[Serusers] error starting ser
by chris 29 Aug '05

29 Aug '05

i tried the command ser -c -f ser.cfg and this is the result. bash-2.05# ser -c -f ser.cfg 0(2092) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: ld.so.1: ser: fatal: libmysqlclient.so.10: open failed: No such file or directory 0(2092) parse error (15,13-14): failed to load module ERROR: bad config file (1 errors) bash-2.05# but, /usr/local/lib/ser/modules/mysql.so is existing, pls advise what could have gone wrong.

3 9

[Serusers] Radius packets not reaching the server
by Ashutosh Kumar 29 Aug '05

29 Aug '05

Hi, My ser-0.9.0 is sending the radius packets and is getting the packets reflected in ngrep output. But they are not reaching the radius server.What may be the reason? The bgrep output output is below: U a.b.c.d:34128 -> x.y.z.s:1812 .~...F..vn......./t...abcd@a.b.c.d....abcd....a.b.c.d.,.*430f5a3a34325aca96d 48bea7555c8ed6d0ab56b....sip:a.b.c.d....REGISTER."e035a87503a63d6d2ae8680c27 475cfd........abcd..E..p....... Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

[Serusers] MARK RODDY/LAKE is out of the office.
by mark.roddy＠lakecommunications.com 29 Aug '05

29 Aug '05

I will be out of the office starting 08/26/2005 and will not return until 09/01/2005. I will not be checking my mail. For Daidalos related issues please contact Stephen Butler. *********************************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)lakecommunications.com This footnote also confirms that this email message has been scanned for the presence of computer viruses and other security threats. ***********************************************************************************

1 0

[Serusers] SER + ASTERISK voicemail
by harry gaillac 28 Aug '05

28 Aug '05

Hello, I try set Ua---SER----Asterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://lists.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] nathelper/rtpproxy.
by harry gaillac 27 Aug '05

27 Aug '05

Hello, Could you send me back more informations about nathelper/rtpproxy/MOH support ? Does nathelper/rtpproxy/MOH require a "music server" ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 2

[Users] $hdr[From] vs $hrd(From)
by Kanakatti Mahesh Subramanya 27 Aug '05

27 Aug '05

why $hdr[From] in avp_write(), and $hdr(From) in avp_printf? just haven't gotten around to it? or some valid syntactical reason? cheers

2 1

[Users] $rt setting avp to the "To" header instead of the "Refer-To" header
by Kanakatti Mahesh Subramanya 27 Aug '05

27 Aug '05

Given the following packet REFER sip:9546236300@69.25.47.137:5090 SIP/2.0 Via: SIP/2.0/UDP 192.168.28.118;branch=z9hG4bK5eb7146d4B8F8EA4 From: <sip:mahesh.aptela@66.23.129.213>;tag=AB447651-36547B1E To: "operator.aptela" <sip:9546236300@69.25.47.137:5090>;tag=as6c1b062a Route: <sip:66.23.129.213;ftag=as6c1b062a;lr=on> CSeq: 2 REFER Call-ID: 688b089d4b10573715a49690228f1586(a)69.25.47.137 Contact: <sip:mahesh.aptela@192.168.28.118> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Refer-To: sip:7100@flareon.aptela.com:5060 Referred-By: <sip:mahesh.aptela@66.23.129.213> Max-Forwards: 70 Content-Length: 0 avp_printf() sets $rt to "sip:9546236300@69.25.47.137:5090" instead of "sip:7100@flareon.aptela.com:5060" trace follows When you run avp_write("$hdr[Refer-To]", "$savedRefer"); avp_print(); you get 2(5034) DEBUG: id=<65> 2(5034) DEBUG: val_str=<sip:7100@flareon.aptela.com:5060> When you run avp_printf("$savedRefer", "$rt"); avp_print(); you get 2(5034) DEBUG: id=<65> 2(5034) DEBUG: val_str=<sip:9546236300@69.25.47.137:5090> t

2 1

[Serusers] NATHelper Issues
by Darren Nay 27 Aug '05

27 Aug '05

Hey All, Got a question for you all. I've had a nathelper implementation in place for months thinking that it was working, but recent events have made me question my configuration. It doesn't seem to be rewriting the contact header when the src_port and contact_port are different. Even though I am using fix_nated_contact() for REGISTER requests. Please see my details below. I am using SER v8.12 Here is an example ngrep capture for a REGISTER. Notice that the source port is 53492, and the contact port is 53616. -------------- U 66.110.232.221:53492 -> 208.253.183.40:5060 at 01:52:55 08/26/2005 REGISTER sip:proxy.ionosphere.net SIP/2.0 Via: SIP/2.0/UDP 66.110.232.221:53492;branch=z9hG4bK-7c0d1eaa From: spectrum wireless <sip:+13367908202@proxy.ionosphere.net>;tag=f8c505c5a8fc2e3o0 To: spectrum wireless <sip:+13367908202@proxy.ionosphere.net> Call-ID: e0bd4b00-2770688f(a)192.168.1.29 CSeq: 10 REGISTER Max-Forwards: 70 Contact: spectrum wireless <sip:+13367908202@66.110.232.221:53616>;expires=600 User-Agent: IONOSPHERE I-PHONE v1.0 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura ------- U 208.253.183.40:5060 -> 66.110.232.221:53492 at 01:52:55 08/26/2005 SIP/2.0 200 OK Via: SIP/2.0/UDP 66.110.232.221:53492;branch=z9hG4bK-7c0d1eaa From: spectrum wireless <sip:+13367908202@proxy.ionosphere.net>;tag=f8c505c5a8fc2e3o0 To: spectrum wireless <sip:+13367908202@proxy.ionosphere.net>;tag=309667ea73a5d01d816917cdbd4fb3ac .6723 Call-ID: e0bd4b00-2770688f(a)192.168.1.29 CSeq: 10 REGISTER Contact: <sip:+13367908202@66.110.232.221:53616>;q=0.00;expires=600 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 208.253.183.40:5060 "Noisy feedback tells: pid=31855 req_src_ip=66.110.232.221 req_src_port=53492 in_uri=sip:proxy.ionosphere.net out_uri=sip:proxy.ionosphere.net via_cnt==1" ------------- Here is a snipet for REGISTER requests from my ser.cfg -------- # Nathelper if (nat_uac_test("3")) { if (method == "REGISTER" || !search("^Record-Route:")) { log("LOG:Someone trying to use private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP force_rport(); setflag(6); }; }; # Turn on SIP accounting setflag(3); if (!method=="REGISTER") record_route(); # loose-route processing if (loose_route()) { route(1); break; }; lookup("aliases"); if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("proxy.ionosphere.net", "subscriber")) { www_challenge("proxy.ionosphere.net", "0"); break; }; save("location") break; }; ## etc.................. }; -------- Then the registration is always recorded as the contact port. Here is a serctl ul output: <sip:+13367908202@66.110.232.221:53616>;q=0.00;expires=529 Does anyone have idea why this is happening? Is my SER.CFG incorrect? Thanks so much! Darren Nay VoIP Network Development Ionosphere, Inc. dnay(a)ionosphere.net

3 3

[Serusers] recording REGISTER messages and errors
by Iqbal 27 Aug '05

27 Aug '05

Hi Is there a way, to do this, in acc table Iqbal

2 1

[Serusers] Radiusclient problem
by Ashutosh Kumar 27 Aug '05

27 Aug '05

Hi, I have installed radiusclient-ng-5.1 (from tarball) with ser-0.9.0. But still SER shuts down saying "ERROR: load_module: could not open module </usr/local/lib/ser/modules/auth_radius.so>: libradiusclient.so.2: cannot open shared object file: No such file or directory". Why is SER still trying to find radiusclient library instead of the radiusclient-ng library which is already installed. Regards, Ashutosh Kumar

2 1

[Serusers] Save call details to mySQL
by Andrew Madrigallos 27 Aug '05

27 Aug '05

Hi Everyone, I'm new in using SER, I was using different server before, my question is how can I save call details on mySQL for future accounting? I browse the archive and everybody answer same thing which is not for newbie please post sample configuration here. Thanks, Andrew _____________________________________________________ Powered by Crown Multimedia (http://www.crownmultimedia.net)

1 0

Re: [Serusers] Checking URI
by Ryan Pagquil 26 Aug '05

26 Aug '05

Yes, it really make sense. I'll find the way how to check those From and To header of a SIP request. Maybe I will use textops.so to do this or any other module that can handle these kind of application. In your experience what did you use to make work this kind of scenario? --Ryan Steve Blair wrote: > > Sorry I missed the route[3] stuff because my mail client wrapped > the message text. Even though I got the route block number wrong I > think my original comment is still valid. If these inter-domain calls hit > route[3] then there you need to either check if the "from" user is > permitted to send calls or you need to check if the "to" address is a > valid > local address. Which approach you take depends on your implementation. > > If your iptel.org users are on soft clients where they can easily > enter the > authentication credentials then I think that would be a good approach. > If you > are trying handle multiple doamins (such as other business partners) > then the > problem is a little more difficult. We simply trust certain remote IP > addresses > to be the address of a "friendly" peer. We cannot prompt them for > credentials > because their users don't have accounts on our system. > > Does this make sense? > > _Steve > > Ryan Pagquil wrote: > >> Hi Steve, >> Here is the route[1] configuration. The calls from outside >> domains are falling to route[3] which make the call to be blocked. >> >> ##Default message handler## >> route[1] { >> >> t_on_reply("1"); >> >> if (!t_relay()) { >> >> if (method=="INVITE" || method=="ACK") { >> end_media_session(); >> }; >> >> sl_reply_error(); >> }; >> } >> >> Thanks, >> Ryan >> >> Steve Blair wrote: >> >>> >>> It looks like the type of call you describe should fall through to >>> route[1] is that >>> the behavior you are seeing? Anyway in route[1] (or somewhere ) you >>> need to >>> accept that the from address might not be a local user and allow for >>> the >>> case where the from address is a "trusted alternate domain like >>> iptel.org". I'm >>> guessing, but I cannot confirm from this code, that route[1] is >>> checking the >>> from user credentials and providing you with the authentication >>> required >>> message. >>> >>> At that point you could elect, and I think a soft phone will, prompt >>> the user for these >>> credentials. >>> >>> -Steve >>> >>> Ryan Pagquil wrote: >>> >>>> Hi Steve, >>>> >>>> Here is what happens with my current configuration. When I >>>> call from my iptel.org account going to my local users it prompts >>>> me "407, Proxy Authenctication Required". So what I like to do is >>>> to allow those calls going to my local users from other domain. >>>> BTW, here is my ser.cfg's INVITE message handler. >>>> >>>> ##INVITE message Handler## >>>> route[3] { if (client_nat_test("3")) { >>>> setflag(7); >>>> force_rport(); >>>> fix_nated_contact(); >>>> }; # if (!proxy_authorize("", "subscriber")) { >>>> # proxy_challenge("", "0"); >>>> # break; >>>> # } else >>>> >>>> if (!check_from() && method=="INVITE") >>>> { sl_send_reply("403", "User From=ID"); >>>> break; }; >>>> >>>> lookup("aliases"); >>>> if (uri!=myself) { >>>> route(1); >>>> break; >>>> }; >>>> >>>> if (!lookup("location")) { >>>> sl_send_reply("404", "User not found"); >>>> break; >>>> }; >>>> >>>> if (method=="CANCEL") { route(1); >>>> break; >>>> }; >>>> >>>> >>>> consume_credentials(); >>>> >>>> if (isflagset(6) || isflagset(7)) { >>>> use_media_proxy(); >>>> }; >>>> >>>> route(1); >>>> } >>>> >>>> I misse call with your SIP phone, just testing it. =) >>>> >>>> Thanks, >>>> Ryan >>>> >>>> Steve Blair wrote: >>>> >>>>> >>>>> Set the alias=<your sip domain> parameter then use if >>>>> (uri==myself) construct. >>>>> >>>>> -Steve >>>>> >>>>> Ryan Pagquil wrote: >>>>> >>>>>> Hello, >>>>>> How can I check if the call is for my domain? What >>>>>> specific module should I use? >>>>>> >>>>>> 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

1 0

[Users] clienst behind same NAT
by jayesh nambiar 26 Aug '05

26 Aug '05

hi all, Thanx Klaus. Sorry to ask a question again related to the same problem. I had already tried writing the avp_write after lookup but even that causes a problem. When avp_write is done it writes the private IP address of the callee while the source_ip is the IP address of the NAT. I beleive this IP address is written because of the fix_nated_contact. Again, when avp_check is done, it returns false and again the RTP session gets started. IS there any way out to this problem. Thanx again.. Jayesh --------------------------------- Start your day with Yahoo! - make it your home page

3 4

[Serusers] exec_dset & rtp stream
by Mik Cheez 26 Aug '05

26 Aug '05

Does anybody have a sample of a script that uses exec_dset to route to a PSTN, as well as handle the RTP stream for NAT'ed and non NAT'ed clients? Everything I've done either ends with a 483 or a large amount of the calls get a 481 "transation does not exist". the exec_dset could return many different values, so I cannot just use 'rewritehost'. :)

1 0

[Serusers] DB Mode in Permissions module
by Ricardo Martinez 26 Aug '05

26 Aug '05

Hello List. I'm using the permission modules (in particular de allow_trusted command) to authorize the trusted hosts in my ser box. I'm also using the db_mode "1" (cache mode), to avoid that every INVITE became a request to my local database. Once in a while i need to add new "trsuted hosts" to my platform, but the changes in the database are not reflected in the allow_trusted query (since is cached). My question is : Is there a way to force the SER to read the changes in the DB without restarting SER.? or i have to use db_mode 0 to acomplish this?. I really appreciate any help here. Thanks in advance. Regards, Ricardo Martinez.-

2 1

[Users] help with uac_redirect
by Kevin M. Murphy 26 Aug '05

26 Aug '05

Greeting all, I am currently working at migrating a Vovida deployment to OpenSer, and need to keep some of the Vovida architecture. I have been messing with the uac_redirect module and have found it to be almost exactly what i am looking for with one exception. I would like to fork calls serially based on contacts in a 302, however i have only been able to do this in parallel. Is there a way to make this happen. Here is some example config stuff, Thanks in advance to all. ---------------------------------------------------------------------- #test redirect config (to service two vovida Proxyies) : # the idea here is that the two ip addresses will query the redirect server first and then route apropriately # the first host here is invalid, while the second should work, if (method=="INVITE") { if(src_ip=="xxx.xxx.xxx.15") { rewritehostport("xxx.xxx.xxx.20:5066"); #prefix("23801"); append_branch("sip:4439121133@xxx.xxx.xxx.20:5060"); sl_send_reply("302", "Moved Temporarily"); } if(src_ip=="xxx.xxx.xxx..20") { rewritehostport("xxx.xxx.xxx.15:5066"); prefix("23801"); append_branch("sip:238014105716405@xxx.xxx.xxx.20:5060"); sl_send_reply("302", "Moved Temporarily"); } #currently the Proxies have a failed route statment like this: failure_route[1] { get_redirects("3:2"); append_branch(); t_relay(); } ---------------------------------------------------------- The behavior I would like to see is that the first contact is tried, and, after failure, the second contact is tried. Thanks again, Kevin

2 1

[Serusers] how do I route to ranges?
by Dana Olson 26 Aug '05

26 Aug '05

I am not sure how to approach this, or even if it's possible. I want to have this kind of thing (pseudocode): if sipuri >0 & sipuri <96 { do stuff } if sipuri >95 & sipuri < 120 { do stuff } And so on. Assuming that the sipuri is a number, and anything outside of the ranges, or letters is ignored (and handled elsewhere). I could build individual routes for each number, but I am hoping there is an easier way to do it. If anyone could help me out, that'd be really awesome. -- Dana

2 2

[Serusers] FW: Radius packets not being sent to radius server
by Ashutosh Kumar 26 Aug '05

26 Aug '05

Hi, Finally I got the radius working. But now I am getting the error message rc_avpair_gen: received unknown VSA attribute 103, vendor 9 of length 18: 0x683332332D72657475726E2D636F64653D31 I am using cardsaver radius (quantum). Can someone explain the error message. Regards, Ashutosh Kumar ________________________________________ From: Ashutosh Kumar [mailto:ashutoshk@chetu.com] Sent: Tuesday, August 23, 2005 12:13 AM To: 'serusers(a)lists.iptel.org' Subject: Radius packets not being sent to radius server Hi, I m using radius with SER-0.9.0 and also using all the reqd functions viz. www_raidus_authorize etc.. , but when I ngrep for any connection with my radius server, I dont get any radius-packets in the result. Also the log file contains rc_send_server: no reply from RADIUS server a.b.c.d.:1812 (a.b.c.d is the radius servers IP). This is despite the fact that the radius server is running and hosting another running service, also radisuclient is configured fine. Kindly guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

1 0

[Users] authentication in multi-domain
by jayesh nambiar 26 Aug '05

26 Aug '05

hi all, Thanx Klaus and Bogdan, for rectifying my idea about using Openser in multi-domain environment. It did help me a lot. But sorry to ask one more question regarding the same issue. How do we authenticate the clients in multi-domain environment. I beleive openser allows to add users with same username in different domain!! If so then is it required that, when a call is made to such user the caller needs to specify the complete username with domain.. I did not actually check how it works but i thought just to clear the doubt. And how do we authenticate the users registered in different domains. pls help me understand this.. Thanx a lot for previous responses. Jayesh ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

2 2

[Serusers] Total NewBie to Linux
by Herman Janse van Vuuren 26 Aug '05

26 Aug '05

Hi There, I have looked at the manuals and stuff, got everything installed, but how to I get it to work? where to I go and enter my IP of my billing server? How do I configure calling cards on this server? How do I register a user with a permanent account. Best Regards, Herman Janse van Vuuren Q-Kon (PTY) LTD Tel:+27 12 665 0052 ext 216 Mobile: +27 82 920 6959 Email: hjvvuuren(a)qkon.com

1 0

Re: [Serusers] Checking URI
by Ryan Pagquil 26 Aug '05

26 Aug '05

Hi Steve, Here is the route[1] configuration. The calls from outside domains are falling to route[3] which make the call to be blocked. ##Default message handler## route[1] { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } Thanks, Ryan Steve Blair wrote: > > It looks like the type of call you describe should fall through to > route[1] is that > the behavior you are seeing? Anyway in route[1] (or somewhere ) you > need to > accept that the from address might not be a local user and allow for the > case where the from address is a "trusted alternate domain like > iptel.org". I'm > guessing, but I cannot confirm from this code, that route[1] is > checking the > from user credentials and providing you with the authentication required > message. > > At that point you could elect, and I think a soft phone will, prompt > the user for these > credentials. > > -Steve > > Ryan Pagquil wrote: > >> Hi Steve, >> >> Here is what happens with my current configuration. When I call >> from my iptel.org account going to my local users it prompts me "407, >> Proxy Authenctication Required". So what I like to do is to allow >> those calls going to my local users from other domain. BTW, here is >> my ser.cfg's INVITE message handler. >> >> ##INVITE message Handler## >> route[3] { if (client_nat_test("3")) { >> setflag(7); >> force_rport(); >> fix_nated_contact(); >> }; >> # if (!proxy_authorize("", "subscriber")) { >> # proxy_challenge("", "0"); >> # break; >> # } else >> >> if (!check_from() && method=="INVITE") { >> sl_send_reply("403", "User From=ID"); >> break; }; >> >> lookup("aliases"); >> if (uri!=myself) { >> route(1); >> break; >> }; >> >> if (!lookup("location")) { >> sl_send_reply("404", "User not found"); >> break; >> }; >> >> if (method=="CANCEL") { route(1); >> break; >> }; >> >> >> consume_credentials(); >> >> if (isflagset(6) || isflagset(7)) { >> use_media_proxy(); >> }; >> >> route(1); >> } >> >> I misse call with your SIP phone, just testing it. =) >> >> Thanks, >> Ryan >> >> Steve Blair wrote: >> >>> >>> Set the alias=<your sip domain> parameter then use if (uri==myself) >>> construct. >>> >>> -Steve >>> >>> Ryan Pagquil wrote: >>> >>>> Hello, >>>> How can I check if the call is for my domain? What >>>> specific module should I use? >>>> >>>> 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

1 0

[Serusers] Check if OUI is registered
by Dmitry Isakbayev 26 Aug '05

26 Aug '05

Hi, When I process an Invite, is there a way to check if the originating UI is registered with the proxy? Thanks, Dmitry

2 1

[Serusers] dbtext module
by Carlos 26 Aug '05

26 Aug '05

Hi, I'm trying to use dbtetx module, i've written the ser.cfg with the necesaries changes, also i do a directory /tmp/serdb with those files(location, aliases, version & subscriber), but when i run my ser i have this problem: /* 0(0) fixing /usr/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/lib/ser/modules/rr.so record_route 0(0) fixing /usr/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/lib/ser/modules/registrar.so save 0(0) DBT:dbt_load_file: request for table [location] 0(0) DBT:dbt_load_file: db is [/tmp/serdb] 0(0) DBT:dbt_load_file: loading file [/tmp/serdb/location] 0(0) DBT: column[0] is STR! 0(0) DBT: column[1] is STR! 0(0) DBT: column[2] is STR! 0(0) DTB:dbt_db_get_table: --------------- 0(0) DBT:db_query: table not loaded! 0(0) DBT:db_query: error while quering table! 0(0) preload_udomain(): Error while doing db_query 0(0) register_udomain(): Error while preloading domain 'location' 0(0) domain_fixup(): Error while registering domain ERROR: error ffffffff while trying to fix configuration 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: tm_shutdown : empting DELETE list 0(0) DEBUG: tm_shutdown : empting hash table 0(0) DEBUG: tm_shutdown: releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : done 0(0) DBT:destroy ... 0(1410) shm_mem_destroy 0(1410) destroying the shared memory lock */ I don`t know if i have to create any table, or something like that, i supose location file is something like this: /* username(str) domain(str,null) password(str) garcia:iptel.org:1234 */ Thanks, Carlos

3 2

RE: [Serusers] Routing multiple numbers to 1 account
by Igor Balk 26 Aug '05

26 Aug '05

The idea is not to query any tables as they are at different server but use radius to provision aliases and then use lookup("aliases") I believe there was a patch a way back for 8.12 but seems it did not found it way to CVS -----Original Message----- From: Chris St Denis [mailto:chris@aebc.com] Sent: Wednesday, August 24, 2005 3:44 PM To: Igor Balk; serusers(a)lists.iptel.org Subject: RE: [Serusers] Routing multiple numbers to 1 account This is basically the same as aliases. Query the registration table. If its not found there push the AVP alias into the ruri/username and lookup based on that. All aliases are is a static registration table anyway I believe. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Igor Balk Sent: Wednesday, August 24, 2005 11:59 AM To: serusers(a)lists.iptel.org Subject: RE: [Serusers] Routing multiple numbers to 1 account Guys, do any of you know if there is a way in SER to dynamically set up aliases using radius? I'm trying to achieve similar results but with aliases. Thank you. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Chris St Denis Sent: Wednesday, August 24, 2005 1:26 PM To: 'Erik Versaevel'; 'SER Users' Subject: RE: [Serusers] Routing multiple numbers to 1 account I am doing this by pulling the translation out of a database using radius AVP (could be done direct to database too) and sticking it as the destination. It looks like this # Try to lookup the number directly registered. if(!lookup("ser_location")) { #Number isn't found, so try to do an avp translation on it avp_pushto("$ruri/username", "s:callee_vdid"); if(!lookup("ser_location")) { #Look up the new value #Still can't find the user. Give up. sl_send_reply("404", "I can't find 'em"); break; } } -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Erik Versaevel Sent: Wednesday, August 24, 2005 6:49 AM To: SER Users Subject: [Serusers] Routing multiple numbers to 1 account Hello all, At this moment i'm using the full e164 numbers as account's in ser, that is for every number someone wants delivered to them they would have to register it using the account details. This however poses a problem with a 100 bock of numbers for example while a device can only register a few of them so what i would like to do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way the devices only has to register once and recieves the number dialed in the incomming INVITE. If i use aliases to alias numbers to 1 account the account allways recieves it's own registered contact instead of the number dialed, is there any way to support number blocks in this way? For example: Device registeres as device_1@sipserver with contact device_1@device_ip:port SER gets invite for number in the block of device_1 and sends the invite as dialed_number@device_ip:port instead of device_1@device_ip:port Kind reagards, Erik Versaevel _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 2

Re: [Serusers] Checking URI
by Ryan Pagquil 26 Aug '05

26 Aug '05

Hi Steve, Here is what happens with my current configuration. When I call from my iptel.org account going to my local users it prompts me "407, Proxy Authenctication Required". So what I like to do is to allow those calls going to my local users from other domain. BTW, here is my ser.cfg's INVITE message handler. ##INVITE message Handler## route[3] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; # if (!proxy_authorize("", "subscriber")) { # proxy_challenge("", "0"); # break; # } else if (!check_from() && method=="INVITE") { sl_send_reply("403", "User From=ID"); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; if (method=="CANCEL") { route(1); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } I misse call with your SIP phone, just testing it. =) Thanks, Ryan Steve Blair wrote: > > Set the alias=<your sip domain> parameter then use if (uri==myself) > construct. > > -Steve > > Ryan Pagquil wrote: > >> Hello, >> How can I check if the call is for my domain? What specific >> module should I use? >> >> 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

1 0

[Serusers] failure_route question
by Java Rockx 25 Aug '05

25 Aug '05

Hi All. I'm using t_check_status() to do call forwarding on 486 Busy responses. I have an odd thing happening where SER receives the 486 Busy message (for which SER properly sends an ACK response) and then the failure_route calls append_branch() and t_relay() to try another destination. After the new leg of the call is setup and then torn down, the original 486 Busy seems to be hung in the tm module, because it magically appears in ngrep, which shows it being sent back to the caller as if the append_branch() never took place. Is there anything special that I should be doing to "swallow" the 486 Busy response to prevent it from being sent to the caller UA? Regards, Paul

10 20

[Serusers] is that compliant?
by Olivier Taylor 25 Aug '05

25 Aug '05

Hi all, I have a problem with ringing message coming from our provider. Associate with the message, i find a SDP. Some UACs doesn't like that and doesn't give a ringback (Snom and Grandstream ata 486) SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.101;rport=1039;received=81.247.182.42;branch=z9hG4bKfeb085ba0caf55 99..To: <sip:0495283361@finalcut.be>;tag=507b8a49..From: " Family Taylor" <sip:7321073@finalcut.be>;tag=cf8413e3df61a5a5..Call-ID: ae6f02682bb43efd@192.168.2.101..CSeq: 4667 INVITE..Record-Route: <sip:82.146.123.252>..Content-Type: application/sdp..Content-Length: 217....v=0..o=VoipSwitch 7772 7772 IN IP4 213.246.41.194..s=VoipSIP..i=Audio Session..c=IN IP4 213.246.41.194..t=0 0..m=audio 6772 RTP/AVP 18 101..a=rtpmap:18 G729/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. Is that correct and compliant? Thanks, Olivier

1 0

[Serusers] Forward to busy and no answer
by Hernando Maisano 25 Aug '05

25 Aug '05

I need its aid, like setting the SER, I have still the problem with call forwarding on busy or no answer? ....I am to use a module AVPOP, but it does not walk. Please help Me..... Thanks

1 0

[Users] printer friendly documentation
by Gary Mart 25 Aug '05

25 Aug '05

First, I like what you are doing, but is their a better way to get printed copies of the documentation than to individually display and print the documentation for each module? Gary

2 1

[Users] Faster to do avp_printf();avp_check OR search()?
by Kanakatti M. Subramanya 25 Aug '05

25 Aug '05

e.g., if I want to do something if there is a Refer-To header in the message, am I better off doing avp_printf("$someAvp", "$rt"); avp_check("$rt", "re/.../i"); or search("/.../"); Just want ot know whether to get started rewriting all the "search()" strings i have in my ser.cfg.... cheers

5 7

[Users] LCR module
by Ozan Blotter 25 Aug '05

25 Aug '05

Dear List, I cannot find enough documentation about LCR Module, anyone using and successful SER.CFG for it ? Thanks, Ozan

2 2

[Users] mediaproxy logging
by Klaus Darilion 25 Aug '05

25 Aug '05

Hi! Is it possible to redirect the syslog output of mediaproxy and proxydispatcher to a certain file? e.g. as openser can: log_facility=LOG_LOCAL4 regards klaus

1 0

[Serusers] Checking URI
by Ryan Pagquil 25 Aug '05

25 Aug '05

Hello, How can I check if the call is for my domain? What specific module should I use? 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

2 1

[Serusers] SERWEB
by Colin Jordan 25 Aug '05

25 Aug '05

I am having trouble to write to mysql from /htdocs/serweb/html/styles/user_interface/index.php. Where am I going wrong DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

1 0

[Serusers] RPM serweb
by Marc Khayat 25 Aug '05

25 Aug '05

Anybody knows where I can get the rpm package of the serweb for ser-0.9.3? Thanks,

1 0

[Serusers] Active and standby gateway
by Sam Lee 25 Aug '05

25 Aug '05

I have 2 voice gateway which apparently both routes the same path. I'm trying to use 1 as an active gateway where my SER routes the voice traffic to, the other used as a hot standby. Are there anyway i can specify this implementation in my configuration? Regards, Sam

2 1

[Serusers] Access Denied Error in SER setup
by Bhaskar Bhumkar 25 Aug '05

25 Aug '05

Hi, In the process of setting up the SER proxy server, I always get the following error : # /usr/sfw/bin/mysqladmin -h <my_machine_name> -u mysql password 'viperdodge' /usr/sfw/bin/mysqladmin: unable to change password; error: 'Access denied for user: '@<my_machine_name>' to database 'mysql'' I am referring to the following document for setting up the proxy server : http://developers.sun.com/solaris/articles/solaris_as_sip/solaris_as_sip.ht… I could finish upto step 5.3 i.e. I finished the test run successfully, however I am having a problem in setting up the root password. If I skip this step and go ahead, I get the following error whenever I try to use serweb using http://<my_machine_name>/admin : Database error: pconnect(<my_machine_name>, ser, $Password) failed. MySQL Error: () Session halted. I tried all the options as per the MYSQL manual which is available at : http://dev.mysql.com/doc/mysql/en/access-denied.html But it's not of much help to me. I will appreciate if somebody has any pointers for this. Thanks, Bhaskar...

2 1

[Serusers] If else using call source
by Sam Lee 25 Aug '05

25 Aug '05

Dear All, Can i do my call routing based on the calling party ? So far from what i see in the examples in iptel.org, it is using the request URI , which is the call destination. Any help ? eg. If i want to route all calls coming from 1234 to a number, how do i do it ? Please let me know if you don't understand my question. Regards, Sam

2 1

[Serusers] Radius Module acc_radius.so Loading Problem
by Fuchs Gernot 25 Aug '05

25 Aug '05

Hello Serusers, After deploying the binary distribution of SER 0.9.3 for Debian from the repository "deb http://apt.sip-router.org/debian/ stable main contrib non-free" I was able to install the RADIUS based AUTH against a FreeRADIUS server. However after loading the acc_radius.so module I encounter module loading errors as in: # WARNING: no fork mode and more than one listen address found(will use only the the first one) stateless - initializing textops - initializing acc - initializing ERROR: error while initializing modules # No other errors are encountered for "ser -c". When I remove the module acc_radius.so, everything is OK again and AUTH works fine against the RADIUS server. My ser.cfg for the modules looks like: # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_radius.so" loadmodule "/usr/lib/ser/modules/acc_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- RADIUS params -- modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # # Your help is very much appreciated! Kind regards, --Gernot

2 1

[Serusers] PSTN gateway - Sipura 3000
by feedbak 25 Aug '05

25 Aug '05

I am using a Sipura 3000 with Ser 8.13 When I forward a PSTN call to the gatway. It disregards the fact that I have stripped the initial 9 using strip(1) and dials the 9 on the PSTN side. I have set the device for one stage Dialing. and have the following in my my ser.cfg # Check for PSTN1 if (uri~"^sip:9") { strip(1); t_relay_to_udp ("192.168.100.15", "5061"); break; }; Any ideas ? Thanks, Dinesh

1 0

Re: [Serusers] Routing multiple numbers to 1 account
by Erik Versaevel 25 Aug '05

25 Aug '05

Disticintive ring is not an option as the device won't handle that, replacing the from would break Caller ID :) However, i might have an idea now. * Create aliases in alias table Invite comes in if(lookup(alias)) {#alias excists Append-hf("Called number: ruri\username") if(lookup(location)) { #user found, replace username with the called number from the Called number header field } That might work, i'll see if i can set this up in my test machine. Thanks again for your help, Erik Versaevel Chris St Denis wrote: >Ah, I see your problem. > >One option there is to either: >1. use distinctive ring to differentiate. Something like: > append_hf("Alert Info: Bellcore-r1\r\n"); >Or >2. Set the caller ID to be the called id with uac_replace_from or something. > >Other than that I can't suggest anything to help you. > >-----Original Message----- >From: Erik Versaevel [mailto:erik@infopact.nl] >Sent: Wednesday, August 24, 2005 2:01 PM >To: Chris St Denis >Cc: 'SER Users' >Subject: Re: [Serusers] Routing multiple numbers to 1 account > >Chris, > >So the invite still gets send to the contact uri sip_user registered in >ser ( in this case sip_user@1.2.3.4:5060), that's not what i want since >the end device won't know which number has been dialed in order to reach >it (it has no way to see that 5551234 has been dialed to get to it). >This looks like alias done by radius instead of the standard ser mysql >aliases. > >Kind regards and thanks for your help, > >Erik > > >Chris St Denis wrote: > > > >>Using the numbers from your example >> >>Call comes in to 5551234 >>5551234 is looked up in registration table. >>5551234 is not found in registration table. >>5551234 is looked up via radius which returns sip_user >>Invite is rewritten to be for sip_user >>sip_user is looked up in the registration table and the usual lookup action >>is taken. >> >>-----Original Message----- >>From: Erik Versaevel [mailto:erik@infopact.nl] >>Sent: Wednesday, August 24, 2005 1:52 PM >>To: Chris St Denis >>Cc: 'SER Users' >>Subject: Re: [Serusers] Routing multiple numbers to 1 account >> >>Chris, >> >>What exactly do you replace in this case? what i need to do is the >>following: >> >>user sip_user registers his contact in ser (ie sip_user@1.2.3.4:5060) >> >>SER recieves invite for 5551234(a)mysipserver.com, makes a lookup in a db >>table, finds that 5551234(a)mysipserver.com belongs to sip_user and >>forwards the message to 5551234@1.2.3.4:5060 (not sip_user@1.2.3.4:5060) >> >>This: >> avp_pushto("$ruri/username", "s:callee_vdid"); >>looks like you replace the username part of the request uri (in this >>case 5551234(a)mysipserver.com) with something else (which is what i kinda >>want to do), >> >>Kind regards, >> >>Erik >> >> >>Chris St Denis wrote: >> >> >> >> >> >>>I am doing this by pulling the translation out of a database using radius >>>AVP (could be done direct to database too) and sticking it as the >>>destination. >>> >>>It looks like this >>> >>> # Try to lookup the number directly registered. >>> if(!lookup("ser_location")) { >>> #Number isn't found, so try to do an avp translation on it >>> avp_pushto("$ruri/username", "s:callee_vdid"); >>> if(!lookup("ser_location")) { #Look up the new value >>> #Still can't find the user. Give up. >>> sl_send_reply("404", "I can't find 'em"); >>> break; >>> } >>> } >>> >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On >>>Behalf Of Erik Versaevel >>>Sent: Wednesday, August 24, 2005 6:49 AM >>>To: SER Users >>>Subject: [Serusers] Routing multiple numbers to 1 account >>> >>>Hello all, >>> >>>At this moment i'm using the full e164 numbers as account's in ser, that >>>is for every number someone wants delivered to them they would have to >>>register it using the account details. >>>This however poses a problem with a 100 bock of numbers for example >>>while a device can only register a few of them so what i would like to >>>do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way >>>the devices only has to register once and recieves the number dialed in >>>the incomming INVITE. If i use aliases to alias numbers to 1 account >>>the account allways recieves it's own registered contact instead of the >>>number dialed, is there any way to support number blocks in this way? >>> >>>For example: >>>Device registeres as device_1@sipserver with contact >>> >>> >>> >>> >>device_1@device_ip:port >> >> >> >> >>>SER gets invite for number in the block of device_1 and sends the invite >>>as dialed_number@device_ip:port instead of device_1@device_ip:port >>> >>>Kind reagards, >>> >>>Erik Versaevel >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >>> >>> >>> >>> >> >> >> >> > > > > -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99

1 0

[Serusers] Routing multiple numbers to 1 account
by Erik Versaevel 24 Aug '05

24 Aug '05

Hello all, At this moment i'm using the full e164 numbers as account's in ser, that is for every number someone wants delivered to them they would have to register it using the account details. This however poses a problem with a 100 bock of numbers for example while a device can only register a few of them so what i would like to do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way the devices only has to register once and recieves the number dialed in the incomming INVITE. If i use aliases to alias numbers to 1 account the account allways recieves it's own registered contact instead of the number dialed, is there any way to support number blocks in this way? For example: Device registeres as device_1@sipserver with contact device_1@device_ip:port SER gets invite for number in the block of device_1 and sends the invite as dialed_number@device_ip:port instead of device_1@device_ip:port Kind reagards, Erik Versaevel

2 4

[Serusers] as it is the logica of scripts?
by Walter Willis 24 Aug '05

24 Aug '05

I was testing with scripts of serr one receives calls of asteriskk and the other can make calls to asterisk? I see that metodo INVITES in one of scripts is repeated twice, in any case podrian to give a serious guide to me of as script in form generates . my idea is to cause that being can receive and send calls to asterisk. thanks

1 0

RE: [Serusers] Routing multiple numbers to 1 account
by Igor Balk 24 Aug '05

24 Aug '05

Guys, do any of you know if there is a way in SER to dynamically set up aliases using radius? I'm trying to achieve similar results but with aliases. Thank you. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Chris St Denis Sent: Wednesday, August 24, 2005 1:26 PM To: 'Erik Versaevel'; 'SER Users' Subject: RE: [Serusers] Routing multiple numbers to 1 account I am doing this by pulling the translation out of a database using radius AVP (could be done direct to database too) and sticking it as the destination. It looks like this # Try to lookup the number directly registered. if(!lookup("ser_location")) { #Number isn't found, so try to do an avp translation on it avp_pushto("$ruri/username", "s:callee_vdid"); if(!lookup("ser_location")) { #Look up the new value #Still can't find the user. Give up. sl_send_reply("404", "I can't find 'em"); break; } } -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Erik Versaevel Sent: Wednesday, August 24, 2005 6:49 AM To: SER Users Subject: [Serusers] Routing multiple numbers to 1 account Hello all, At this moment i'm using the full e164 numbers as account's in ser, that is for every number someone wants delivered to them they would have to register it using the account details. This however poses a problem with a 100 bock of numbers for example while a device can only register a few of them so what i would like to do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way the devices only has to register once and recieves the number dialed in the incomming INVITE. If i use aliases to alias numbers to 1 account the account allways recieves it's own registered contact instead of the number dialed, is there any way to support number blocks in this way? For example: Device registeres as device_1@sipserver with contact device_1@device_ip:port SER gets invite for number in the block of device_1 and sends the invite as dialed_number@device_ip:port instead of device_1@device_ip:port Kind reagards, Erik Versaevel _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] VoIp and Java
by javabuddy 24 Aug '05

24 Aug '05

Is there any available api for java on VoIp. - javabuddy. --------------------------------------------------------------------------- Broadband/rich interface + mail box saftey = Roomity.com. ~~1124911739820~~ SIP_Router_Users_List.roomity.com no sign up to read, ad supported. ---------------------------------------------------------------------------

2 1

[Serusers] call transfer process
by Iqbal 24 Aug '05

24 Aug '05

Hi If I have a inbound call frompstn which hits SER, and then the call is passed to asterisk, the receiver then decides to transfer the call, can this be controlled within asterisk/ser, without the PSTN gateway being involved, I was thiking that although REFER messages are sent, couldnt SER take care of this and just transfer the call to a new sip contact. The problem I have got is that I have a upstream gateway which doesnt support REFER :-) Iqbal

1 0

[Serusers] SER default p-time 20 ms
by Frank Kostin 24 Aug '05

24 Aug '05

Hello, I try debugging and get this error "found match with different intervals (20 != 30)" Is it possible to change the p-time into SER from 20 to something else like 30 or 40 ? Thanks, frank --------------------------------- Start your day with Yahoo! - make it your home page

2 1

[Serusers] ACKs not getting routed?
by Chris St Denis 24 Aug '05

24 Aug '05

I am having a problem with incoming calls where ACK messages dont seem to be getting (loose) routed. Here is the relevant part of my config ====================== route { xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n"); # 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 -- probebly a loop or something"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy for accounting purposes record_route(); #Accounting if a call is started or ended if (method=="INVITE" || method=="BYE" || method=="CANCEL") { acc_rad_request("200"); } # grant Route routing if route headers present # loose-route processing. Decreases security slightly regarding # forged messages, but decreases load. if (loose_route()) { xlog("L_NOTICE", "Loose Routing\n"); t_relay(); break; }; ====================== My understand is this should cause the ACK messages that are part of the transaction to get routed. They should not pass beyond this block of code right? However, the SIP conversation goes as follows. Gateway invites proxy Proxy says 100 trying Proxy says 100 trying Proxy invites user User says 100 trying User says 180 Ringing User says 180 Ringing Thus far all is well When the user picks up the phone User tells proxy 200 ok Proxy tells gateway 200 ok Gateway tells proxy ACK At this point the proxy should be relaying the ACK to the user, but it does not. Instead due to timeout the user sends another 200 ok and that last block of conversation repeats. Any ideas why this would not be working? (I tried to send this with a full SIP dialog but it got rejected as being too big so this email is the chopped down version)

1 1

[Users] configuring multi-domain support
by jayesh nambiar 24 Aug '05

24 Aug '05

Hi all, I was trying to configure my script for hosting multiple domains. I went through the admin's guide and understood the procedure of creating new subscriber table for each domain and serve them. But I would be very thankful if someone let me know how does automated multi-domain support works and how do we maintain multiple domains using automated support. I think using domain table will help, but any information regarding serving multiple domains in an efficient manner will be very helpful. Thanx a lot. jayesh.. --------------------------------- Start your day with Yahoo! - make it your home page

3 2

[Users] BRANCH route support
by Bogdan-Andrei Iancu 24 Aug '05

24 Aug '05

Hi everybody, I just finished adding support for BRANCH routes. The idea and frame were partial ported from SER (thanks to Juha for extracting and adapting patches). For those who are not familiar with this concept: branch route is a route to be execute separately for each branch before being sent out - changes in that route should reflect only on that branch. I say partial because the features, implementation were improved. Shortly here is a list: - remove from grammar default BRANCH ROUTE definition - since this is a 100% TM route, make no sense - RURI may be changed from a branch route (the routing will accordingly change); This capability is essential, since there are core function which alter the RURI, functions which cannot be blocked in specific route types - not handling this, may easyly lead to memory faults or leaks. - flags per branch - if flags are changed in branch_route, the change will not be visible for other branches, but it will be visible for all replies of that branch (it might be useful for per branch NAT traversal) - full AVP support in branch route - more efficient handling of lumps (changes in request) - instead of cloning and removing the entire lump tree for each branch, using lump flags, after each branch, the original lump tree will be just cleaned only what was added during branch route will be removed). still pending: how to handle append_branch() core function.....I will prefer to handle properly all core functions, since their execution cannot be banned from specific routes. For the moment only textops, avpops and xlog modules were enabled for branch route. Other will come in the future. regards, Bogdan PS: testing done only superficial (is quite late now).....any help/reports are welcomed :D

3 7

[Serusers] SER IP PBX for multiple clients
by Waldo Rubinstein 24 Aug '05

24 Aug '05

Hello, I'm still trying to learn more about SER. I've been using Asterisk to manage virtual PBX services for different companies by using multiple contexts within Asterisk. However, since I only use Asterisk with SIP UAs and to communicate with ITSPs, I don't have the need to have all the fancy features Asterisk offers, plus I have the additional advantage of having the built-in NAT support in SER. The question I have is if someone can point me to the right place where I can see some sample configs that do more or less the things I need or if someone would be willing to share some of those configs on the list. Basically, I need to have the ability to manage any number of virtual PBX services where each virtual entity can manage their own extension numbering, DIDs, outbound CLID. I would probably just continue using Asterisk for voicemail services since I get the feeling that SEMS is still "unstable" to go into production, even for voicemail services only (unless told otherwise). The idea is that even if two companies define two extensions 101 and two extensions 102, when each company calls ext 101 or 102 internally, the call will be maintained within that company and would not cross over to the other company. Also, having features like one main number (DID) that could ring in multiple extensions simultaneously (forking) and if no one answers, leave a message in the general mailbox, or that each extension could have, additionally, a DID so that they can be reached directly from the outside. Any help would be greatly appreciated. Thanks, Waldo

3 5

[Serusers] SEMS + SER
by c waddy 24 Aug '05

24 Aug '05

Hi, I have version 0.8.14 of ser up and running and would now like to allow conference call bridging. >From what i have read is to use SEMS? Can someone point me to some documentation or provide me with an example to integrate this into my SER server. I cant find much information on this module.

1 0

[Serusers] invite timers
by Jeremy George 24 Aug '05

24 Aug '05

I need to set inv_timer to at least 30 seconds to account for the long failover times apparently used by cellular providers. Not hard to understand; it takes a while to dig that cell phone out of the pocket. But this seems to cause inbound calls to take the same long time to failover to voicemail (an Asterisk system running on a separate box.) Do I understand the timers correctly? Is there a way to cause a relatively short inbound failover and keep a longish outbound failure? - Jeremy

2 2

RE: [Fwd: Re: [Serusers] ul_add flag and serctl]
by Aisling 24 Aug '05

24 Aug '05

Hi Iqbal, Jan, I changed my log to /tmp/fifo in my ser.cfg and the serctl.dat file. I then tried adding an alias: Serctl alias add 7890 sip:aisling@x.x.x.x Again I got the 404 flags expected error and I checked the /tmp/fifo file. It was empty. I then did /opt/ser/sbin/serctl fifo ul_add aliases aisling(a)x.x.x.x 7890 0 1.00 0 128. (It would work unless I added those 4 parameters) It said 200 Added to table and the following appeared in my new log file: :ul_add:ser_receiver_28642 aliases aisling(a)x.x.x.x 7890 However while this said added above. When I dialed 7890 to reach the aisling(a)x.x.x.x client, a 404 was sent to my phone. Many Thanks, Aisling. -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: 24 August 2005 12:29 To: Aisling O'Driscoll Subject: [Fwd: Re: [Serusers] ul_add flag and serctl] could try this -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] lcr module
by Sebastian Kühner 24 Aug '05

24 Aug '05

Hello, I have a problem with the lcr module if a gateway has "big" IP Address. I calculate the integer of the IP like this: a.b.c.d ==> IP-Integer = d*pow(2,24) + c*pow(2,26) + b*pow(2,8) + a If the d-value is very big (100 or more), the resulting IP Address doesn't work anymore. What is the greatest that an integer can have? I use postgres with this module. I hope that this won't be a problem! Thanks! Sebastian

4 4

[Serusers] www_challange and proxy_challange
by Thomas Britis 24 Aug '05

24 Aug '05

Hi all, Is it sane to use: if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; } else { if (!www_authorize("", "subscriber")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; }; }; }; Or anything here appears to be wrong ? Thank you. -- Thomas Storino Britis TCNet Inf e Telecom LTDA

4 5

[Serusers] Best bersion of radiusclient to use with ser-0.9.0
by Ashutosh Kumar 24 Aug '05

24 Aug '05

Hi, Which is the best known stable version of radiusclient, radiusclient4.x or radisuclient5.x, or radiusclient-ng to use with ser-0.9.x, more specifically, ser-0.9.0. I have been bewildered by the earlier archived posts of this list some of which advocate radiusclient5.x, and some advocate radisuclient4.x, while in some , it is mentioned that radiusclient-ng works very fine with ser-0.9.x, but my ser is unable to even detect radiusclient-ng (it keeps loading the old radiusclient5.1 so's, despite the fact that rc-ng has been compiled from source and ldconfig'ed ). An authoritative response will be appreciated. Regards, Ashutosh Kumar

2 1

[Serusers] ser reinstall problem connected with mysql module load
by Richard Luo 24 Aug '05

24 Aug '05

Hi all, Thanks for the tips. I tried to reinstall my ser. not by rpm file. but compiling source code and make install. Then I met the problem when i running the ser: ================================================== [root@pc5 ~]# /usr/local/sbin/ser : command not found: line 6: : command not found: line 8: : command not found: line 12: /usr/local/sbin/ser: line 13: /bin: is a directory : No such file or directory6: */ : command not found: line 17: : command not found: line 24: : command not found: line 26: : command not found: line 29: /usr/local/sbin/ser: line 30: loadmodule: command not found /usr/local/sbin/ser: line 31: loadmodule: command not found /usr/local/sbin/ser: line 32: loadmodule: command not found /usr/local/sbin/ser: line 33: loadmodule: command not found /usr/local/sbin/ser: line 34: loadmodule: command not found /usr/local/sbin/ser: line 35: loadmodule: command not found /usr/local/sbin/ser: line 36: loadmodule: command not found : command not found: line 37: : command not found: line 42: : command not found: line 44: : command not found: line 46: /usr/local/sbin/ser: line 47: syntax error near unexpected token `"usrloc",' 'usr/local/sbin/ser: line 47: `modparam("usrloc", "db_mode", 0) =================================== but my ser.cfg is already revised to the required one. as: ========================= [root@pc5 ~]# cd /usr/local/etc/ser [root@pc5 ser]# ls dictionary.ser old ser.cfg ser.cfg~ ser.cfg.edited ser.cfg.old ser.cfg.sample [root@pc5 ser]# cat ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (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=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//usr/local/lib/ser/modules/auth.so" loadmodule "//usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("iptel.org", "subscriber")) { www_challenge("iptel.org", "0"); break; }; save("location"); 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(); }; } ================================ Sorry about too much copy files. But could anyone can tell me where the problem is and what the solution? Thanks a lot. -- _______________________________________________ Search for businesses by name, location, or phone number. -Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.as…

1 0

[Serusers] radiusclient-ng/SER attributes for accounting
by Alberto Cruz 24 Aug '05

24 Aug '05

Hi guys, does anybody knows if it's possible to give a format to the "calling-from" and "calling-to" values that radiusclient-ng/SER send to the Radius server? I have a licensed billing application that is expecting to receive the "calling-from" and "calling-to" values like in a number format in order to apply the rates this mean: if "calling-from" or "calling-to" value is 836572345(a)sip.domain the billing application needs to receive 836572345 without the @sip.domain What should i do at SER or radiusclient-ng in order to achieve this format? Regards Alberto Cruz

3 4

[Serusers] User-Agent: Asterisk PBX.
by fedora＠voice.dyndns.tv 24 Aug '05

24 Aug '05

# U 2005/08/20 00:13:30.111494 221.186.150.67:5060 -> 221.186.150.68:5060 BYE sip:8000@221.186.150.68:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:221.186.150.67;ftag=as1cbee789;lr=on>. Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0. Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. Contact: <sip:500@221.186.150.67:5062>. Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. Route: <sip:8000@221.186.150.68:5060>. . # U 2005/08/20 00:13:30.147446 221.186.150.68:5060 -> 221.186.150.67:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0. Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. Contact: <sip:8000@221.186.150.68:5060>. Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. CSeq: 102 BYE. Server: X-Lite release 1103m. Content-Length: 0. . # U 2005/08/20 00:13:30.149017 221.186.150.67:5060 -> 221.186.150.67:5062 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. Contact: <sip:8000@221.186.150.68:5060>. Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. CSeq: 102 BYE. Server: X-Lite release 1103m. Content-Length: 0.

4 3

[Serusers] ul_add flag and serctl
by Iqbal 24 Aug '05

24 Aug '05

Hi been having a offlist discussion with Aisling, who has a problem with using serctl to add alias. The error we get is ul_add flag expected error 1. We have looked at mysql logs, and there is nothing sent, hence error must be in script/fifo 2. I have tested the same script on my server, and it works with no errors 3. We have used serctl ul_add to send raw comannds, and again it works on aisling's server also 4. Ran a diff between my working sercl and aisling's and there is no diff. The flow I think of info is serctl ---> fifo --> DB is there anyway of seeing what fifo is doing, or any other thoughts, this error DOES NOT invlove serweb, or its config scripts, its just serctl on its own Iqbal

2 1

[Serusers] t_on_failure and t_on_reply in the same block
by maka 24 Aug '05

24 Aug '05

Hi all, Can I have t_on_reply() and t_on_failure() in the same ser route block, i.e. in the route block processing INVITEs? On-replied transactions are processed in the on_reply route block for nat traversal, but how do I specifically catch the 486 busy message if want to play a voice message for them? cheers

2 1

[Serusers] speeding up db update
by Johan 24 Aug '05

24 Aug '05

I noticed that SER will take some time to update location table everytime I quit from my UA. Is there anything that I can do to speedup this update ? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] Adding "Allow" to the reply
by Federico Giannici 24 Aug '05

24 Aug '05

According to RFC3261, a reply of code 405 (Method Not Allowed) MUST include an "Allow" header field. How can I add this header field? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 1

Re: [Serusers] Distributing "OSP enable SER" binary file w/ source
by vmathur＠transnexus.com 24 Aug '05

24 Aug '05

Hi Jan, You had mentioned in your mail that you provide binary tarballs and packages, which are kept in local repositories. Further to that, I have the following questions: 1. Could you tell me the location of the repository. 2. I am guessing that in order for me to post our package there, I would need to send it to you or is it possible for me to post it directly. 3. When you say "packages", you mean rpm ? 4. Finally, is it possible for me to create a binary of OSP enabled SER and post it there ? Many Thanks, Vikrant ------------------------------------------------------------------------------------------------------------- Yes, we provide binary tarballs and binary packages. In case of binary packages we even have our own repositories for Redhat and Debian distributions (so if you create packages of the library then you could put it there). Jan.

2 1

[Serusers] generating password by hand
by Roger Thomas 24 Aug '05

24 Aug '05

Hi, I have a user registration that writes user info to ldap database. Whenever a new user registers, I would like to populate SER's subscriber table too. How to I generate the HA1 password with PHP ? Please advise. ../zhza --------------------------------------------------- Sign Up for free Email at http://ureg.home.net.my/ ---------------------------------------------------

4 4

[Serusers] unix socket interface documentation?
by Federico Giannici 24 Aug '05

24 Aug '05

I have seen that, besides the fifo interface, SER has an unix socket interface too. Is there any documention or at least any example of usage? What are the advantages over the fifo interface? If I'm correct, serctl uses only the fifo interface, right? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 1

[Serusers] Sems troubleshooting
by Pavol Segeč 24 Aug '05

24 Aug '05

Hi, May somebody give me some suggestion how to troubleshoot sems? I need to troubleshhoot sems voicemail, I configured ser (0.9.3) + sems. I'd like use sems voicemail, I configured ser and sems following Configure-Sems-Ser-HOWTO with pasing mail info through fifo_db_url. I'm receiving voicemail announcement about leaving voice message, but e-mail with voice mail is not sent. In syslog I don't see any error, mailq is empty, message wasn't received by postix. I tried also configure mail passing through tw_append, but setting described in manual is probably out of date, because I have some database reading problems with it. Thanks pavol

1 0

[Serusers] (no subject)
by share phone 23 Aug '05

23 Aug '05

Hi All, Mediaproxy help needed is this working with openser. Regards, Sunil --------------------------------- Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

1 0

[Serusers] missed_calls username and domain
by Ah khng 23 Aug '05

23 Aug '05

Hi all, I want to display missed calls in SERWEB, so i use acc_db_request to store the entry into table missed_calls. However, username and domain weren't recorded into database. While i noticed that serweb is comparing username and domain in sql WHERE cluase (method.get_missed_calls.php) which cause no records being display. Thus, how do i make domain and username recorded into database? # -- accounting parameters -- modparam("acc", "log_level", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) failure_route[1] { # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted acc_db_request("missed calls","missed_calls"); revert_uri(); if (t_check_status("486") || t_check_status("603")) { log(1,"User busy\n"); } else if (t_check_status("480") || t_check_status("408")) { log(1,"User is unavailable\n"); } else { log(1,"User is unavailable for unknown reason\n"); } rewritehostport("202.73.10.7:5060"); append_branch(); t_relay_to_udp("202.73.10.7", "5060"); } mysql> select sip_from,sip_status,sip_method,username,domain from missed_calls; +----------------------------------------------+--------------+------------+----------+--------+ | sip_from | sip_status | sip_method | username | domain | +----------------------------------------------+--------------+------------+----------+--------+ | <sip:10001@202.73.10.8>;tag=ff2015eeb8c23171 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=ff2015eeb8c23171 | missed calls | CANCEL | n/a | n/a | | <sip:10001@202.73.10.8>;tag=688a799e25d70d76 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=9da5279e291e12cc | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=1fbde47420f30ff2 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=bde153a0cf74ce23 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=bc3362f1cb8b0f64 | missed calls | INVITE | n/a | n/a | +----------------------------------------------+--------------+------------+----------+--------+ 9 rows in set (0.00 sec) Regards Seong

2 1

[Serusers] call through
by Iqbal 23 Aug '05

23 Aug '05

Has anyone implemented a call through scenario eg dial 001 234512367, and thats hits ser, then press # (am thinking this will not work just with SER) dial the next number, and the call is terminated at the second number. 2 stage dialing. Iqbal

2 1

[Serusers] "stun" parameters in Contact HF
by Massimo Barletta 23 Aug '05

23 Aug '05

Hi all, I'm testing a new UA and I've got a Contact Header Field format in REGISTER msg I'm not aware of: REGISTER sip:/sip.mydomain.com/:5060 SIP/2.0. Via: SIP/2.0/UDP *0.0.0.0:0;*branch=z9hG4bK2497750963. From: <sip:/pstn_number/@/sip.mydomain.com/>;tag=3670695360. To: <sip:/pstn_number/@/sip.mydomain.com/>. Call-ID: 2377110845(a)192.168.1.4. CSeq: 1 REGISTER. Contact: <sip:/pstn_number/*@0.0.0.0:0;**stun=4*>. max-forwards: 70. expires: 3600. user-agent: oSIP/Linphone-1.1.0-IMVP. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO. Content-Length: 0. i.e. ip:port is no longer in Contact or Via Header Fields and a stun=4 parameter is specified in Contact HF Have you got any clue on RFC reference for this behaviour? How can I handle this on SER? (SER fills location table as follows: username: /pstn_number/ domain: /sip.mydomain.com/ contact: sip:/pstn_number/@0.0.0.0:0;stun=4 received: sip:/UA_ip_address/:5060 ) THX, Mety

2 1

[Serusers] Developers' Guide
by Ahmed Kellal 23 Aug '05

23 Aug '05

Hello, I was reading the SER DG when I came upon a sentence (twice on the same page) saying that there is a chapter dealing with, and I quote, SIP Message Modifications. However, the said chapter does not exist in the 0.8.8 version of the document. Can someone shed some light on the subject? :-) Cause I really need to understand the workings of that part of the server. Thanks in advance Ahmed _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] Compiling SER to use radiusclient-ng library
by Ashutosh Kumar 23 Aug '05

23 Aug '05

Hi, I want to use radiusclient-ng library with my SER. Previously, I was using radiusclient. Now that I have installed from source radiusclient-ng on the machine and also configured SER to sue, it still seems to be using old radiusclient, and not radiusclient-ng, because when I delete the modules libradius.so.* from my machine, then I get the error by SER "libradiusclient.so.2: cannot open shared object file: No such file or directory", which surely indicates that SER is configure to use only radiusclient at compile time . Is there any switch/ directive while compile time to indicate that SER should use radiusclient-ng , and not radiusclient? Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

[Users] avp_write
by Richard Z 23 Aug '05

23 Aug '05

Hi, If there are multiple branches, will avp_write("$ruri/username","i:123") generate multiple avps with each for one branch? Or it only grabs the first one? Thanks, Richard

3 8

[Serusers] help: fix_nated_contact_f
by Ahmed Kellal 23 Aug '05

23 Aug '05

Hello, I am trying to add a function to SER, in order to only alter the host part of the Contact URI. To do that, I'm trying to draw inspiration from the existing fix_nated_contact_f function written in nathelper.c Problem is there are various points I just don't seem to understand. If anyone can help me answer these questions, I would sure appreciate it!! 1- the parameters declared for the fix_nated_contact_f function are (struct sip_msg* msg, char* str1, char* str2) But the code itself only exploits the first parameter, if I am not mistaken. If that is the case, what are the others here for? 2- the lines: if (uri.port.len==0) uri.port.s = uri.host.s + uri.host.len; What I understand from this is the following: If there isn't a port specified in the original message (the port pointer is not pointing to anything), set the pointer's value (the pointer in the port part of the URI structure) to point to the memory location right after where the host part in the URI ends. Is that correct? And, if so, is that done in order to prepare for an eventual insertion of a port when going through the NAT? 3- the line: offset = c->uri.s - msg->buf; I won't even venture a guess here. A total mystery to me. 4- the line: len = c->uri.len + strlen(cp) + 6 /* :port */ - (uri.port.s + uri.port.len - uri.host.s) + 1; What I gather here is that we're computing the relative difference in size (length) between the already existing ip:port field in the message, and the one we're about to replace it with (source address of the packet), is that right? 5- the part: temp[0] = uri.host.s[0]; temp[1] = c->uri.s[c->uri.len]; Euh!! Is he taking just the first and last characters from the URI host string, putting them in a "safe place" (to be later restored) and putting '\0' characters instead to "delimit" something or other? Or am I totally off mark??? 6- the line: len1 = snprintf(buf, len, "%s%s:%d%s", c->uri.s, cp, msg->rcv.src_port,uri.port.s + uri.port.len); Here, len1 will hold the size of the memory size (buffer) needed to insert the new data, right? 7- the condition: if (len1 < len) len = len1; You'll have to enlighten me on that one, cause I don't see it... I'm sorry to bother you with such a long mail, but I need the help! And if someone has a better approach to my original problem(cf. first line of the mail) please do not hesitate :-) Thank you in advance _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] ser register at external SIP-Server
by Müller Thorsten 23 Aug '05

23 Aug '05

Hello all, is it possible to send a REGISTER with username and password to an external SIP-Server right after starting up of ser? I thought at uac-Module or tm-Module, any other suggestions? Thanks Thorsten

2 1

[Serusers] serweb on different server
by Johan 23 Aug '05

23 Aug '05

I have ser running on svr1. Would it be possible to run serweb from another server, say, svr2 ? What do I need to achieve this ? TIA. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] openser :
by share phone 23 Aug '05

23 Aug '05

Hi All, lookup(): '@192.168.1.66' Not found in usrloc my SIP_DOMAIN was sharephone.networks.com and my IP Address is 192.168.1.66 and i added the FQDN in the /etc/hosts file and when i use the UA to register it registers but after a while i get the folloing message in the log. Thanks, Sunil. 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) parse_headers: flags=ffffffffffffffff 2(14038) check_via_address(192.168.1.65, 192.168.1.65, 0) 2(14038) DEBUG:destroy_avp_list: destroying list (nil) 2(14038) receive_msg: cleaning up 2(14038) SIP Request: 2(14038) method: <OPTIONS> 2(14038) uri: <sip:192.168.1.66:5060> 2(14038) version: <SIP/2.0> 2(14038) parse_headers: flags=2 2(14038) Found param type 235, <rport> = <n/a>; state=6 2(14038) Found param type 232, <branch> = <z9hG4bKc0a801410000000b430a4e92000005dd000005ec>; state=16 2(14038) end of header reached, state=5 2(14038) parse_headers: Via found, flags=2 2(14038) parse_headers: this is the first via 2(14038) After parse_msg... 2(14038) preparing to run routing scripts... 2(14038) parse_headers: flags=100 2(14038) DEBUG: get_hdr_body : content_length=0 2(14038) get_hdr_field: cseq <CSeq>: <456> <OPTIONS> 2(14038) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(14038) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 2(14038) DEBUG: add_param: tag=286578432823134 2(14038) end of header reached, state=29 2(14038) parse_headers: flags=200 2(14038) end of header reached, state=9 2(14038) DEBUG: get_hdr_field: <To> [25]; uri=[sip:192.168.1.66:5060] 2(14038) DEBUG: to body [<sip:192.168.1.66:5060> ] 2(14038) found end of header 2(14038) find_first_route: No Route headers found 2(14038) loose_route: There is no Route HF 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) parse_headers: flags=ffffffffffffffff 2(14038) check_via_address(192.168.1.65, 192.168.1.65, 0) 2(14038) DEBUG:destroy_avp_list: destroying list (nil) 2(14038) receive_msg: cleaning up --------------------------------- Start your day with Yahoo! - make it your home page --------------------------------- Start your day with Yahoo! - make it your home page

2 1

[Users] ruri, dset, branches ...
by Klaus Darilion 23 Aug '05

23 Aug '05

Daniel-Constantin Mierla wrote: > Looking at this issue, I am wondering if worth to add the > pseudovariables $dh and $dp for destination host and port ... Also I > am considering to change a bit the $du reference so when the dst_uri > is not set to point to request URI. Would be these changes usefully? Hi Daniel! This reveals another question: What is the difference between: request URI, first branch, all branches, destionation sets ..... I used xlog to log all this pseude variables just before t_relay. first branch: $br=sip:klaus@10.10.0.50:8718 all branches: $bR=sip:klaus@10.10.0.50:8718 destination set: $ds=Contact: sip:klaus@83.136.33.19:5060, sip:klaus@10.10.0.50:8718 destination uri: $du=<null> request uri: $ru=sip:klaus@83.136.33.19:5060 In the above call scenario, there is just lookup() which reveals 2 contacts, one with public IP and one with NATed IP (received structure used). Can you please describe the difference of these variables? - Why does the destination set contains the NATed IP instead of the received-IP? - Why does all branches only shows the first one? - Why is the request URI different to the first branch? regards klaus

4 5

[Serusers] ser.cfg problem cont.
by Richard Luo 23 Aug '05

23 Aug '05

Hi, I tried to uncomment the ser.cfg file as the instruction to use mysql. but i met some problem. the sys log file shows: Aug 22 19:13:52 pc5 ser: ERROR: load_module: could not open module </usr/lib/ser/modules/mysql.so>: /usr/lib/ser/modules/mysql.so: cannot open shared object$ Aug 22 19:13:52 pc5 ser: parse error (28,13-14): failed to load module Aug 22 19:13:52 pc5 ser: set_mod_param_regex: No module matching auth_db found | Aug 22 19:13:52 pc5 ser: parse error (56,21-22): Can't set module parameter Aug 22 19:13:52 pc5 ser: set_mod_param_regex: No module matching auth_db found | Aug 22 19:13:52 pc5 ser: parse error (61,20-21): Can't set module parameter could anyone can help? ============================ You won't admit you love me, and so How am I ever to know You always tell me... Perhaps, perhaps, perhaps -- _______________________________________________ Search for businesses by name, location, or phone number. -Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.as…

4 3

[Serusers] ser+mediaproxy behind nat
by jeff kwong 23 Aug '05

23 Aug '05

Hi everyone! I am currently trying out the examples from the Getting Started manual version 04a from Onsip.org. The reference design there shows that SER itself is nated. My question is how will a UA from the public internet or somewhere outside the SER's nat register with the SIP server? Will I put the SER's nat public ip as the sip proxy and do I have to set some kind of port forwarding on the SER's nat? Also, doesn't mediaproxy have to listen on a public ip? The document is actually a good tutorial and just need additional clarifications for issues like nat. Thanks Guys!

2 2

[Users] UAC module: parameter not found
by Franz Edler 22 Aug '05

22 Aug '05

Hi, I am just testing UAC module (development version). Using the parameter "from_restore_mode" is not possible because it is not found in module uac. Debug log: set_mod_param_regex: parameter <from_restore_mode> not found in module <uac> This is strange, because it is defined in the list of exported parameters of uac.c module v 1.4. Any ideas? Franz

4 6

[Serusers] Radius packets not being sent to radius server
by Ashutosh Kumar 22 Aug '05

22 Aug '05

Hi, I m using radius with SER-0.9.0 and also using all the reqd functions viz. www_raidus_authorize etc... , but when I ngrep for any connection with my radius server, I don't get any radius-packets in the result. Also the log file contains "rc_send_server: no reply from RADIUS server a.b.c.d.:1812" (a.b.c.d is the radius server's IP). This is despite the fact that the radius server is running and hosting another running service, also radisuclient is configured fine. Kindly guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 3

[Serusers] Mediaproxy
by Juan 22 Aug '05

22 Aug '05

Hi. I am trying to have mediaproxy starting on reboot but when doing chkconfig --add mediaproxy I am getting: service mediaproxy does not support chkconfig. How can I make it to start on reboot, then? Thanks Juan

1 0

[Serusers] Missed Calls not being displayed in Serweb
by Rafael R. GV 22 Aug '05

22 Aug '05

Hi I have ser-0.9.3 and Serweb-0.9.3, mysql4 and accounting works correcty but missed calls not being displayed in serweb. I have read in the list that method.get_missed_calls.php script from serweb require domain column in aliases tables and username/domain columns in missed_calls table and I have the values "n/a" in those columns so if that the case, How can I set these values correctly? any ideas? thanks Rafael Risco -- rrgv

1 1

[Serusers] Timeout
by Sebastian Kühner 22 Aug '05

22 Aug '05

Hi! Does anyone of you know how to stop the timeout timer manually? Thanks! Sebastian

3 7

[Serusers] reroute
by Jason Mohler 22 Aug '05

22 Aug '05

Hello everyone, I am novice with SER reroute. Can anyone help explain to me how I can set my SER up to reroute when I get a particular SIP message back (e.g. 503 ) and route to another IP. Your help with this matter will be greatly appreciated.

1 0

[Users] trusting src_ips using AVP
by jayesh nambiar 22 Aug '05

22 Aug '05

hi all, I went through the avp docs in voicesystem.ro and i thought the source_ip trusting in multi domain environment would be useful to me. I used permissions to do this before, but i think multidomain support is not possible in permissions module. But I jus wanted to know that on updating the table how does openser get updated without restarting the server. Is there a FIFO command or something. pls let me kno cause I did not understand the example mentioned in the docs properly. One more thing, Does Openser 0.9.5 support use_domain parameter mentioned in the docs. thanx.. jayesh ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

3 4

[Serusers] Problem with natted X-lite and public gateway
by Carlo 22 Aug '05

22 Aug '05

Hi all, I've a problem beetween X-lite softphone and Cisco Sip Gateways. Both SER and Cisco Gateways are configured with public ip addresse, while the x-lite is behind a nat router. It seems that the x-lite doesn't take into account the ser ip address in SDP connection information field but it sends rtp to the ip of the cisco gateway. Can someone help me ? How can I try to solve the problem? Thanks Carlo

1 0

[Serusers] fifo aliases working, serctl aliases not working
by Aisling 22 Aug '05

22 Aug '05

Hello, I need help troubleshooting a problem I am having adding aliases. It works fine using direct FIFO commands but not with my serctl script. I have not modified the serctl script so its just the standard script. Basically I am continually told that flags are expected. See below: 1) serctl alias add test1 sip:3456@serveraddress sip:3456@serveraddress 400 ul_add: flags expected 2) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 400 ul_add: expires expected 3) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 400 ul_add: q expected 4) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 400 ul_add: replicate expected 5) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 0 400 ul_add: flags expected 6) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 0 128 400 ul_add: flags expected 7) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 0 128 200 Added to table ('sip:3456', 'test1') to 'aliases' Any help appreciated, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] problem with serctl command
by Richard Luo 22 Aug '05

22 Aug '05

Hi guys, i met the problem with using serctl to add the new users. I have the FC4 on my pc. I have uncommented the ser.cfg file for using mysql. and the error information is like follow: [cluo@pc5 ~]$ /usr/sbin/serctl add richard 000000 richard_luo(a)lycos.com /usr/sbin/serctl: line 764: gen_ha1: command not found HA1 calculation failed And if i changed to the root user, the error information is changed like these: /usr/sbin/serctl: line 401: mysql: command not found could anyone help me for this problems? best richard ============================ You won't admit you love me, and so How am I ever to know You always tell me... Perhaps, perhaps, perhaps -- _______________________________________________ Search for businesses by name, location, or phone number. -Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.as…

1 0

[Serusers] configuration for public ser and natted pstn gateway
by Simone Cittadini 22 Aug '05

22 Aug '05

I've the following configuration : internet --- ser -||-- asterisk where ser has two network interfaces, one with public ip and one with 192.168.1.1, asterisk sits in the 192.168.1.0 net, authorization and forwarding of calls from internet works well with this ser.cfg (I omit parameters and obvious stuff): if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; lookup("aliases"); }; if (method=="INVITE") { record_route(); }; rewritehost("192.168.1.101"); <--- asterisk ip if (!t_relay()) { sl_reply_error(); }; but of course you can't hear nothing with this config... Googling I only find configurations relative to the ser machine also being natted, can someone tell me what I need to do in my particular case ?

4 3

[Serusers] accounting question
by Nhadie 22 Aug '05

22 Aug '05

Hi All, Is my query correct SELECT t1.from_uri as orig_number, t1.to_uri as term_number, TIMEDIFF(t2.time, t1.time) as duration, t1.timestamp as calldate FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND ((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR (t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND t1.sip_method='INVITE' AND t2.sip_method='BYE' also this query displays all calls fron all user and it shows also all the previos month, what can I add here to show only calls from a certain user e.g 2009 and the current month? I tried this: SELECT t1.from_uri as orig_number, t1.to_uri as term_number, TIMEDIFF(t2.time, t1.time) as duration, t1.timestamp as calldate FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND ((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR (t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND t1.sip_method='INVITE' AND t2.sip_method='BYE' AND t1.username="2009" and t1.time REGEXP '-08-'; <--- I added this line but i had an error, hope someone can help me. thank you. regards, nhadie

1 0

[Serusers] Module Avpops
by Hernando Maisano 22 Aug '05

22 Aug '05

Hello, I need to do, group of line, rotating lines. Somebody I advise myself that it uses I modulate AVPOPS. Somebody this using? Somebody can help me? I am reading http://www.voicesistem.ro/index.phplang=ro&page=tehnologii&sub_page=diferit… but it does not obtain it. Thanks Nando

2 4

[Users] OpenSER and SEMS ?
by Ozan Blotter 22 Aug '05

22 Aug '05

Dear List, Does anyone playing with SEMS 0.9.0 with OpenSER 0.9.X ? I couldn't find VM module to use with SEMS. Thanks, Ozan

3 2

[Users] Proper FROM mangling
by Bogdan-Andrei Iancu 22 Aug '05

22 Aug '05

Hi everybody, Based on the latest dialog supported (which was added in RR module), I was able to implement in the "uac" module proper FROM replacement. The missing issue was correct restoring/replacement of FROM in the sequential requests. As restoring information was kept in FROM param, this information didn't show up in all sequential requests (for example in ones from original caller to callee). Now, based on the new RR API, the module keeps the restoring info in the Route header which will show up for sure in all sequential request; so the module will be able to track and fix all sequential replies and requests. This implies fixing FROM or TO hdr, depending of the request direction. For people not interested in implementation details, I put first the usage description: There are 3 restore mode for the module to work (for FROM mangling): "none" -> NO RESTORE; the FROM URI will be changed and no further effort or info will be done/store in order to restore its value neither is sequential replies, nor requests; "manual" -> information needed for restoring the value will stored in RR parameter; all the replies and local generated requests (ACK and CANCEL) will be automatically fixed, but no sequential request will be restored; the admin has full liberty to do it from script by using the restore_from() function. "auto" -> FULL AUTO RESTORING -> information needed for restoring the value will stored in RR parameter and all sequential request and replies will be fixed *automatically* NOTE: for restore modes different than "none", the module will try to bind itself to RR and TM modules, so take care and load them. For people interested in implementation details. Shortly, here is how it works: Let's consider a call initiated from UA A to UA B (or from A to B) Now, since sequential requests may flow from A to B or B to A we need to be able both to restore original TO in requests from B to A and to replace with new FROM in request from A to B; which means that during the dialog we need to know both old and new FROM uris. As in each situation on of the value is known (depending of the direction, either the new, either the new URI appears), the simplest and most efficient way to store both in minimum size is to store the XOR value of them; having the XOR value and one of the uris, we can determine the other URI ( (x^y)^y=x :) ) A Proxy B 1) initial request: From:OLD_URI FROM:NEW_URI RR:(OLD_URI^NEW_URI) ---------------------------------> -----------------------------------> 2) final reply From:OLD_URI FROM:NEW_URI RR:(OLD_URI^NEW_URI) RR:(OLD_URI^NEW_URI) <--------------------------------- <----------------------------------- 3) re-invite (direction downstream) From:OLD_URI FROM:NEW_URI Route:(OLD_URI^NEW_URI) ---------------------------------> -----------------------------------> 4) final reply From:OLD_URI FROM:NEW_URI <--------------------------------- <----------------------------------- 5) BYE (direction upstream) TO:OLD_URI TO:NEW_URI Route:(OLD_URI^NEW_URI) <--------------------------------- <----------------------------------- 6) final reply TO:OLD_URI TO:NEW_URI ---------------------------------> -----------------------------------> 1) initial equest has the FROM URI changed from scrip via replace_from() ; XOR between NEW_URI and OLD_URI is added to RR as parameter 2) the reply (actualy all replies) is fixed via TM callbacks 3) the re-invite is detected via RR callback as having the RR param; as the direction is downstream,it will have the OLD_URI *FROM* changed with the NEW_URI = (OLD_URI^NEW_URI)^OLD_URI 2) the reply is fixed via TM callbacks 3) the BYE is detected via RR callback as having the RR param; as the direction is upstream,it will have the NEW_URI *TO* changed with the OLD_URI = (OLD_URI^NEW_URI)^NEW_URI and this is all the magic behind :) regards, bogdan

3 4

[Serusers] relaying fifo
by Iqbal 22 Aug '05

22 Aug '05

Hi I cant recall the name of the proggie to help relay ser_fifo from one machine to another, has anyone installed it, and does it work properly, I want to move fifo away from the central install of ser iqbal

1 0

[Users] SDP
by ggtel 22 Aug '05

22 Aug '05

Hello, Any use of SDP media attribute in conjunction with SIP /SER ? I would appreciate any insight! George ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] 2 SER network using 1 PSTN provider
by smy＠addm.com 22 Aug '05

22 Aug '05

Hi all... Just came across a problem with running 2 individual SER for 2 separate network. They both work fine independently, yet I have subscribed for PSTN landing of calls for only 1 SER server (IP restrictions). Now, for my second SER server, I have forwarded calls to the first server (much like pointing a call to it using "if uri=~" command........ and of course, in the trusted table of SER 1, I have placed the IP of the SER 2. Now... the call goes through and the can be landed, yet the PSTN user can not hear anything while the IP Phone user can hear perfectly from the PSTN user. The 2 SER servers are setup based on onsip.org with mediaproxy configuration. Everything works fine, except for this 1 way audio problem. Could some please point me to the right direction to where to solve this?!?! By the way, if my IP phone were to use a REAL IP on SER 2 and try to land calls via SER 1, everything works perfectly!!! There will be 2 way audio. So I would imagine this has something to do with mediaproxy or NAT issues. PLEASE HELP!! :>

2 1

[Serusers] ser configuration issues
by AChin63457＠aol.com 22 Aug '05

22 Aug '05

I just started installing ser the other day and the installation was smooth and painless. I am know not quite sure about about the configuration files and how to set them. Here is my current setup -- I have a fedora 3 linux box obtaining a dhcp assigned address with currently no hostname set and no FQDN-- localhost.localdomain. I signed up the other day to iptel.org to register for a username and password -- ie, _example(a)iptel.org_ (mailto:example@iptel.org) . So i have 2 issues. 1. What should i specify in my /etc/profile for SIP_DOMAIN -- should i use my _example(a)iptel.org_ (mailto:example@iptel.org) or my internal lan address or my wan address available to the outside? 2. Same question when it comes to configuring the /usr/local/etc/ser/ser.cfg what should be listed on the listen parameters listen=127.0.0.1 listen=192.168.x.x or should it be the wan address also can i use the alias alias=_example(a)iptel.org_ (mailto:example@iptel.org) or should this be my machines hostname. Regards, Alfred

2 1

[Users] Re: Re: Re: Re: MediaProxy & PSTN GW
by Ozan Blotter 22 Aug '05

22 Aug '05

Dear Klaus, As i get ideas from three people here i should close this subject. Codec i use is g729 (it supports g711 too) as it comes default with Zyxel Prestige 2002, even i have a special firmware (engineered one) for this product so it supports g723 too :P, i think problem is high latencies in ADSL or CableNet services here, even they're 512/128, 1024/256, 2048/512 in ADSL and 512/256, 1024/512, 2048/1024 in CableNet... Anyway i will try to find another solution and thanks for your replies... Ozan... ------------------------------ Message: 3 Date: Sat, 20 Aug 2005 23:03:59 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [Users] Re: Re: MediaProxy & PSTN GW To: cosmocid(a)ispro.net.tr Cc: users(a)openser.org Message-ID: <43079ABF.9030405(a)pernau.at> Content-Type: text/plain; charset=us-ascii; format=flowed That does not help you - inserting another hop can not compensate the bottleneck between the UAs and the Internet (the ADSL connection). BTW: 600km is not far away. Which Codec you you use? Try low-babdwidth codecs like G729, ilbc, GSM and G723 klaus Ozan Blotter wrote: > Dear Iqbal and Bogdan, > > That is exactly what i need to improve QoS, because the UAs ADSL > connection is not fast enough, 512/128k in general, OpenSER and UA's are > located in same city but PSTN GW is nearly 600 km (350 miles nearly) away > from this point. I thought since OpenSER is on a good connection (1Mbit > Fiber) and can relay to PSTN faster then UA's themselves. > > Thanks, > Ozan > > Also in terms of ping times how far is the PSTN gateway, as Bogdan said, > media proxy will add even more delay, as for sending pstn via > mediaproxy, you could do all yourprocessing, and once ser detects that > call must goto PSTN, then do all the NAT checking ...this is a > oversimplified view... > > Iqbal > > Bogdan-Andrei Iancu wrote: > > >>Hi Ozan, >> >>so, if I get it right, you have a PSTN gw locate far away from the >>UAs...and instead of letting the media traffic to do directly between >>GW and UAs, you want to insert in the middle a RTP relay (mediaproxy) >>in order to improve the QoS, right? >>If so, I would say this will not help you - having a media relay on >>your media path leads to QoS degradation due additional delays.... >> >>If you still want to go for it....I will take a look on tour script. >> >>regards, >>bogdan >> >>Ozan Blotter wrote: >> >> >>>Dear List, >>> >>>At below config. i setted up OpenSER on Public IP and PSTN Gateway is >>>again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL >>>modems on both client side and note that they have SIP ALG, also i've >>>completed port forwarding in each modem, so SIP2SIP call happens >>>(voice traffic goes from UA to UA) without need of anything extra as >>>STUN or RTP RELAY. But since those are very away from each other if a >>>UA needs to call PSTN the voice traffic flows from UA to PSTN and >>>that lowers QoS :( >>> >>>I would like to activate MediaProxy again but just for one case which >>>is PSTN, so whenever a UA needs to call PSTN the voice traffic flows >>>like : >>> >>>UA (ATA) >>> OpenSER + MediaProxy >>> PSTN >>> >>>How can i setup this like above without touching SIP2SIP calls >>>between UA's... >>>or adding the lines beginning with # these to my existing config is >>>enough ? >>> >>>Thanks, >>>Ozan Blotter

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[Users] Re: Re: MediaProxy & PSTN GW
by Ozan Blotter 22 Aug '05

22 Aug '05

Dear Iqbal and Bogdan, That is exactly what i need to improve QoS, because the UAs ADSL connection is not fast enough, 512/128k in general, OpenSER and UA's are located in same city but PSTN GW is nearly 600 km (350 miles nearly) away from this point. I thought since OpenSER is on a good connection (1Mbit Fiber) and can relay to PSTN faster then UA's themselves. Thanks, Ozan Also in terms of ping times how far is the PSTN gateway, as Bogdan said, media proxy will add even more delay, as for sending pstn via mediaproxy, you could do all yourprocessing, and once ser detects that call must goto PSTN, then do all the NAT checking ...this is a oversimplified view... Iqbal Bogdan-Andrei Iancu wrote: > Hi Ozan, > > so, if I get it right, you have a PSTN gw locate far away from the > UAs...and instead of letting the media traffic to do directly between > GW and UAs, you want to insert in the middle a RTP relay (mediaproxy) > in order to improve the QoS, right? > If so, I would say this will not help you - having a media relay on > your media path leads to QoS degradation due additional delays.... > > If you still want to go for it....I will take a look on tour script. > > regards, > bogdan > > Ozan Blotter wrote: > >> Dear List, >> >> At below config. i setted up OpenSER on Public IP and PSTN Gateway is >> again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL >> modems on both client side and note that they have SIP ALG, also i've >> completed port forwarding in each modem, so SIP2SIP call happens >> (voice traffic goes from UA to UA) without need of anything extra as >> STUN or RTP RELAY. But since those are very away from each other if a >> UA needs to call PSTN the voice traffic flows from UA to PSTN and >> that lowers QoS :( >> >> I would like to activate MediaProxy again but just for one case which >> is PSTN, so whenever a UA needs to call PSTN the voice traffic flows >> like : >> >> UA (ATA) >>> OpenSER + MediaProxy >>> PSTN >> >> How can i setup this like above without touching SIP2SIP calls >> between UA's... >> or adding the lines beginning with # these to my existing config is >> enough ? >> >> Thanks, >> Ozan Blotter > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > . >

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Re: [Users] SDP
by Bogdan-Andrei Iancu 22 Aug '05

22 Aug '05

Hi George, OpenSER is a signalling SIP proxy and it has very few implications over the SDP (media) part - only regarding NAT traversal. the SDP info is mostly used end-to-end (only by user-agents) regards, Bogdan ggtel wrote: >Hello, > >Any use of SDP media attribute in conjunction with SIP /SER ? >I would appreciate any insight! > >George > > > > > > > >___________________________________________________________________________ >Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger >Téléchargez cette version sur http://fr.messenger.yahoo.com > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

Re:[Serusers] Centralized routing for managing multiple SER proxies
by Samuel Osorio Calvo 22 Aug '05

22 Aug '05

Take a look at RFC3263 Locating SIP Servers.The idea is to use DNS to locate the right server. If you configure your DNS to serve your different SER proxies you just have to use t_relay and the request will be forwarded to the right destination. Hope it helps. Samuel Unclassified. >>> <vmathur(a)transnexus.com> 08/19/05 06:20PM >>> Hi Ashutosh, Thanks for your response. The readme file of mediaproxy talks about load balancing the RTP streams across multiple proxies. Does the signaling also follow the same path as the media (for the same call). Also, The readme file of the mediaproxy says that: "Once it has determined which domain to use to obtain the list of available proxy servers, it will make a DNS lookup for the following SRV records: _mediaproxy._tcp.selected.domain.name" Does this mean that the DNS lookup serves as the routing mechanism for the call? Also, the document says: "However it should be fairly easy to implement another dispatcher that uses a different mean to determine the proxy server (like using a database mapping), or even to distribute the traffic by other means than SIP domain name." I guess that this means that I could use a 3rd party routing engine to route the calls. Is that correct? If so, is there an interface in the dispatcher that I could use to make the dispatcher talk to my routing DB and not do a DNS lookup. Thanks, Vikrant

1 0

Re: [Users] Bus error
by Samuel Osorio Calvo 22 Aug '05

22 Aug '05

Try to compile SER without any optimization parameter set. Bus error usually refers to a bad alignemnt and usually the optimization paramters of the compiler sometimes do not properly behave with the alignemnt. Hope it helps. Samuel. Unclassified. >>> Marc Haisenko <haisenko(a)be-ok.com> 08/17/05 06:07PM >>> On Tuesday 16 August 2005 22:53, Andréa Giordanna wrote: > Hello, does anybody here already have this problem > while starting openser? > And, what does this Aliases exactly means? > > [...] > > Bus error > > Thanks, > Andrea Giordanna "Bus error" is a very critical OS error message, it has nothing to do with OpenSER. You either have faulty hardware or hit a kernel bug ;-) C'ya, Marc -- Marc Haisenko Linux Solutions Be O.K. service group GmbH Rüdesheimer Straße 7 D-80686 München Tel: +49 (0)89 - 548 43 33 21 Fax: +49 (0)89 - 548 43 33 29 e-mail: haisenko(a)be-ok.com http://www.be-ok.com _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

RE: [Serusers] Serweb
by Marc Khayat 22 Aug '05

22 Aug '05

Anybody got the Log.php file from serweb (ser-0.9.3), coz it seems like it's missing from my package. Thx, Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marc Khayat Sent: Friday, August 19, 2005 11:25 AM To: Karel Kozlik Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Serweb /tmp/ser_fifo/ is wrong since it'll be looking for a folder... Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marc Khayat Sent: Friday, August 19, 2005 11:03 AM To: Karel Kozlik Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Serweb Hi Karel, Thx for bearing with me :) In config_paths.php, I changed it from "/tmp/ser_fifo" to "tmp/ser_fifo/" The errors at the top disappeared but I'm still getting the "FIFO not running or bad path to it" message. Is it wrong to put "tmp/ser_fifo/"? knowing that in ser.cfg, it's /tmp/ser_fifo Thanks again, Marc -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: Friday, August 19, 2005 9:28 AM To: Marc Khayat Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Serweb sorry Marc, haven't idea what may be wrong. Please check again setting of fifo in serweb. Check variable $config->fifo_server in config_paths.php. It's value must be '/tmp/ser_fifo'. Try enable logging in serweb ($config->enable_loging = true in config.php), set log level to debug ($config->log_level = "PEAR_LOG_DEBUG") and examine the log file if there are some messages refered fifo. Also try search for string 'fifo_server' throught log files (or whole serweb) and check if it's value isn't changed somewhere. You should find this string only in file config_paths.php (one times) and in file functions.php (four times). Karel Marc Khayat napsal(a): > Hi Karel, > I already tried this one by manually changing the permissions on > ser_fifo to 666; I just tried adding fifo_mode=0666 to ser.cfg, but > still the same problem. > > Marc > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Friday, August 19, 2005 8:21 AM > To: Marc Khayat > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Serweb > > Hi, > try add: > fifo_mode=0666 > > to your ser.cfg > > Karel > > Marc Khayat napsal(a): > >>/tmp/ser_fifo exists, and is pipe type >>User running Apache is "apache". Owner of ser_fifo is User: "root", >>Group: "root". >>I joined user "apache" to group "root", same. >> >>I created group "ser" and joined "root" "apache" to that group, and >>added fifo_group=ser to the ser.cfg, but still same problem: >> >>Warning: filetype(): Lstat failed for (null) (errno=13 - Permission >>denied) in /var/www/html/serweb/html/functions.php on line 306 >> >>Warning: Cannot modify header information - headers already sent by >>(output started at /var/www/html/serweb/html/functions.php:306) in >>/var/www/html/serweb/html/page.php on line 35 >> >> >>Thanks, >> >>Marc >> >>-----Original Message----- >>From: Karel Kozlik [mailto:karel@iptel.org] >>Sent: Thursday, August 18, 2005 1:38 PM >>To: Marc Khayat >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Serweb >> >>Hello, >>please check if file /tmp/ser_fifo exists and if it's type is pipe. > > Also > >>check rights - if user under which apache is running have access to > > read > >>and write to this file. >> >>Karel >> >>Marc Khayat napsal(a): >> >> >>>The domain thing is working, but problem 2 is still bugging me. >>>Help anyone? >>> >>>Marc >>> >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 11:15 AM >>>To: serusers(a)lists.iptel.org >>>Subject: RE: [Serusers] Serweb >>> >>>Never mind guys... it's working! >>> >>>Marc >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 10:53 AM >>>To: serusers(a)lists.iptel.org >>>Subject: [Serusers] Serweb >>> >>>Hi all, >>>I'm using SER-0.9.3 and its serweb version. >>>I have 2 problems with it: >>>1- On the login page, when I enter the username, it automatically >>>completes it with the FQDN, while I need only the domain name; i.e. I >>>login to http://ser.xyz.com, I enter the username aloha, it's >> >>completed >> >> >>>as aloha(a)ser.xyz.com while I need it as aloha(a)xyz.com. Currently, I'm >>>writing in manually >>> >>>2- When I login, I'm getting an error on the first page only as "FIFO >>>not running or bad path to it". The path to it (in config_paths.php >> >>file >> >> >>>is /tmp/ser_fifo, which is correct), and when I enter the command >> >>serctl >> >> >>>monitor, the counter isn't constant, and same path. >>> >>>Can anyone help me plz, >>> >>>Thanks, >>> >>>Marc >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] Error during accounting - val2str()
by Carlo 22 Aug '05

22 Aug '05

1 0

[Serusers] STUN and public ip address change
by Kevin Brennan 22 Aug '05

22 Aug '05

When testing with X-lite and GPX-2000 behind NAT using only STUN for NAT traversal we loose registration and calls fail if the public ip address changes i.e. if the ISP is assigning with DHCP. The reason it fails is because the UA's fail to recognise the change in public IP and continue to use the old IP. Just wondering if anybody has suggestions to avoid this (other than use an RTP proxy) and/or experiences with UA's which have a STUN implementation which solves this. Br /Kev/

1 1

[Serusers] urgent help
by Abdul Lateef 21 Aug '05

21 Aug '05

Hi, Does anyone know how to configure RTC Client to register with ser. i was doing in VB6 but i am not able to register. the RTC is always sending SUBSCRIBE CSeq not REGISTER. or is there any possibilite to use SUBSCRIBE instead of REGISTER? Please let me know if someone got successfull configuration with RTC Client. Regards, Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

1 0

[Serusers] LCR Module - Does it support PSTN prefix digits?
by Paul Hazlett 21 Aug '05

21 Aug '05

Hi All. We have multiple PSTN gateways in our network. Some are our own and some are owned and operated by 3rd parties. We are also use ser-0.9 stable with the LCR module for call routing. Some of these 3rd party PSTN gateways require us to prefix accounting digits to the R-URI before sending the call to their network. So my questions is this; does the LCR module support a scenerio whereby it can prefix digits for specific PSTN gateways? For example, suppose I have 3 PSTN gateways. The the second gateway requires prefixed digits on the R-URI. When the fist GW fails, the LCR module would prefix the required digits and then send to the send GW. If the second GW fails, then the LCR module would remove the previously prefixed digits and then send the call to the 3rd GW. Is this possible is the LCR module Juha provides for ser-0.9 stable? If not, is it possible with the LCR module in the unstable CVS HEAD? Regards, Paul

3 4

[Users] allow_trusted() in cvs-head?
by Kanakatti Mahesh Subramanya 21 Aug '05

21 Aug '05

Its present in permissions.c, but not in the associated docs. Oversight? Nefarious plot? Devious plan to eliminate it in future releases? Enquiring minds want to know....

2 1

[Serusers] know is a user is registered
by Gustavo Villegas 20 Aug '05

20 Aug '05

Dear Users I'd like to know how can i know is a user is registerd, so he can make calls. and users who aren't registered, send a error and disable make calls the point is the users aren't registered can't make calls. Regards GV

1 0

[Serusers] call timeouts
by Iqbal 20 Aug '05

20 Aug '05

Hi Is there any standard or best user experience for inv_timeouts and fr_timer for IP calls and ip to pstn calls. Also can you have different timers for different types of calls and destinations, maybe even based on prefix/gateway lookup iqbal

2 1

[Serusers] Update
by Lazy.charles 20 Aug '05

20 Aug '05

1 0

[Serusers] SER Launch problems
by Kyle Johnson 20 Aug '05

20 Aug '05

Thanks Luba- I built from source.. And I can't seem to find any references to ser-mysql-module later than v. 0.8.12 (I'm using 0.9.3) I've tried making the whole package with- #make exclude_modules="" modules But I still get that error.. I'm not really sure where to find/compile the ser-mysql-module from... Thanks again for your assistance- Kyle On 8/19/05 11:50 AM, "Luba Vincent" <vincent_luba(a)yahoo.fr> wrote: > You should install the package "ser-mysql-module" > Vincent > > > Le vendredi 19 août 2005 à 11:30 -1100, Kyle Johnson a écrit : > >> All- >> >> I've just compiled SER on RHEL 4, 32 bit, and I'm having this error when >> trying to launch it. >> >> Aug 19 16:28:25 sip1 ser: ERROR: load_module: could not open module >> </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: >> undefined symbol: PL_memory_wrap >> >> I've got these MySQL RPMs instlaled.. >> >> [root@sip1 src]# rpm -qa |grep -i mysql >> mod_auth_mysql-2.6.1-2.1 >> mysql-4.1.10a-1.RHEL4.1 >> mysql-devel-4.1.10a-1.RHEL4.1 >> perl-DBD-MySQL-2.9004-3.1 >> php-mysql-4.3.9-3.2 >> mysql-server-4.1.10a-1.RHEL4.1 >> MySQL-shared-standard-4.1.13-0.rhel4 >> >> My version of SER is- >> >> [root@sip1 src]# /usr/local/sbin/ser -V >> version: ser 0.9.3 (i386/linux) >> flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, >> SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ >> main.c compiled on 16:27:07 Aug 19 2005 with gcc 3.4 >> >> Can anyone offer a hint to get me past this point? >> >> Many thanks in advance- >> >> Kyle >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] pa module
by Mark Aiken 19 Aug '05

19 Aug '05

Hi, Is anyone using the pa module in 0.9.3? I have a problem where the NOTIFY always sends 'closed', even though the user is registered. Also, the NOTIFY is only sent when a SUBSCRIPTION request is received and does not seem to reflect changes to the registrar. Mark

1 0

[Serusers] SER Launch problems
by Kyle Johnson 19 Aug '05

19 Aug '05

All- I've just compiled SER on RHEL 4, 32 bit, and I'm having this error when trying to launch it. Aug 19 16:28:25 sip1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: PL_memory_wrap I've got these MySQL RPMs instlaled.. [root@sip1 src]# rpm -qa |grep -i mysql mod_auth_mysql-2.6.1-2.1 mysql-4.1.10a-1.RHEL4.1 mysql-devel-4.1.10a-1.RHEL4.1 perl-DBD-MySQL-2.9004-3.1 php-mysql-4.3.9-3.2 mysql-server-4.1.10a-1.RHEL4.1 MySQL-shared-standard-4.1.13-0.rhel4 My version of SER is- [root@sip1 src]# /usr/local/sbin/ser -V version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 16:27:07 Aug 19 2005 with gcc 3.4 Can anyone offer a hint to get me past this point? Many thanks in advance- Kyle

1 0

[Users] More, but different crashes (in fastlock.h?)
by Kanakatti Mahesh Subramanya 19 Aug '05

19 Aug '05

Trying openser-head, and have noticed that after an hour or so of heavy usage, it just stops responding to incoming requests. The only way out is to kill it, and restart a backtrace on the resulting core shows #0 destroy_kill () at fastlock.h:76 76 fastlock.h: No such file or directory. in fastlock.h (gdb) bt #0 destroy_kill () at fastlock.h:76 #1 0xb7d1dc7a in exec_shutdown () at exec_mod.c:92 #2 0x08078502 in destroy_modules () at sr_module.c:355 #3 0x0805fb5d in cleanup (show_status=0) at main.c:351 #4 0x080614ea in main (argc=0, argv=0xb7fb9800) at main.c:1580 Something obvious out of whack? cheers

2 1

[Serusers] Openser 407
by share phone 19 Aug '05

19 Aug '05

Digit Authentication working with openser.Any user has this info i am using 0.9.5 ver. --------------------------------- Start your day with Yahoo! - make it your home page

2 1

[Serusers] cancal
by Sebastian Kühner 19 Aug '05

19 Aug '05

Hello, I'm using ser to forward to PSTN Gatways. Before the t_relay I put the command: t_on_failure("2"); So, after a timeout ser goes to: failure_route[2] { log (1, "next gateway...\n"); if (t_check_status("408|500|503")) { if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); break; } log (1, "gateway changed...\n"); } t_on_failure("2"); t_relay(); } This is working very good... if the user doesn't make a hangup. If caller hangs up, the CANCEL hits SER and the call is cleanly terminated. SER however continues to failure_route after timeout of the initial INVITE... ser sends out a new INVITE message to another PSTN-Server... and the phone rings (with nobody in the line) Many thanks for your help! Sebastian

2 2

Re: [Serusers] cancal
by Sebastian Kühner 19 Aug '05

19 Aug '05

Hello, I made a ngrep and I noticed that NO "487 Request Cancelled" is reaching my ser on CANCEL. That could be the problem. I also tried to use the avpops module: if (method == "CANCEL") { setflag(1); avp_write("cancel", "s:failover"); log(1, "-CANCEL PSTN-\n"); }; ...and then with avp_check in the failover route. Without result. Could it be that the variables in the avp don't reach another route? Thanks for your help Sebastian ----- Original Message ----- From: "Bayan William Towfiq" <william(a)telepacket.com> To: "Sebastian Kühner" <skuehner(a)veraza.com> Cc: <serusers(a)lists.iptel.org> Sent: Thursday, August 18, 2005 3:39 PM Subject: Re: [Serusers] cancal > Hi Sebastian, > Sorry about that, I misread the code. I will test your code and dig up > an example of how I do it in my failure routes to show you. > > William > > Sebastian Kühner wrote: > > >Hi William, > > > >Thanks for your help. > > > >In my failover route I have the following code: > > > >failure_route[2] { > > if (t_check_status("408|500|503")) > > { > > log (1, "next gateway...\n"); > > if (!next_gw()) > > { > > t_reply("503", "Service not available, no more gateways"); > > break; > > } > > log (1, "gateway changed...\n"); > > t_on_failure("2"); > > t_relay(); > > } > >} > > > >So the status is already filtered. I tried your code but without result... > > > >Does anybody know how I can stop the failover timer? > > > >Thanks! > > > >Sebastian > > > > > > > >----- Original Message ----- > >From: "Bayan William Towfiq" <william(a)telepacket.com> > >To: "Sebastian Kühner" <skuehner(a)veraza.com> > >Cc: <serusers(a)lists.iptel.org> > >Sent: Thursday, August 18, 2005 1:28 AM > >Subject: Re: [Serusers] cancal > > > > > > > > > >>Hi Sebastian, > >>If the message is a cancel you it will have status 487 > >> > >>so you can just add > >> > >> if (t_check_status("487")) { > >> break; > >> }; > >> > >>before the rest of the code in the failure route. You can mail me > >>privately if you have any more questions about this issue. > >> > >>William > >> > >>Sebastian Kühner wrote: > >> > >> > >> > >>>Hello, > >>> > >>>I'm using ser to forward to PSTN Gatways. > >>> > >>>Before the t_relay I put the command: > >>> > >>>t_on_failure("2"); > >>> > >>>So, after a timeout ser goes to: > >>> > >>>failure_route[2] { > >>> log (1, "next gateway...\n"); > >>> if (t_check_status("408|500|503")) > >>> { > >>> if (!next_gw()) > >>> { > >>> t_reply("503", "Service not available, no more gateways"); > >>> break; > >>> } > >>> log (1, "gateway changed...\n"); > >>> } > >>> t_on_failure("2"); > >>> t_relay(); > >>>} > >>> > >>>This is working very good... if the user doesn't make a hangup. > >>> > >>>If caller hangs up, the CANCEL hits SER and the call is cleanly > >>> > >>> > >terminated. > > > > > >>>SER however continues to failure_route after timeout of the initial > >>>INVITE... ser sends out a new INVITE message to another PSTN-Server... > >>> > >>> > >and > > > > > >>>the phone rings (with nobody in the line) > >>> > >>>Many thanks for your help! > >>> > >>>Sebastian > >>> > >>> > >>> > >>>_______________________________________________ > >>>Serusers mailing list > >>>serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > >>> > >>> > >>> > >>> > >> > >> > >> > > > > > > > > > > >

2 6

[Serusers] Upgrade
by Marc Khayat 19 Aug '05

19 Aug '05

Hi all, I have ser-0.8.14 installed on my Linux with MySQL support. I want to upgrade it to ser-0.9.3. What do you advise me to do? Use bin or src? Thanks, Marc

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[Serusers] Re: checking called number
by Kamran Ahmad 19 Aug '05

19 Aug '05

hello i m using ser-0.9.0 on 5060 and asterisk-1.0.9 on 5970 but problem is that i cannot send my ser back to asterisk. the problem is that it always goto route(4) second else if is not properly checked how to check this condition. i m adding 0 when asterisk is sending call back. if(uri=="^sip:00[1-9]+@.*") { route(4); break; } else if (uri=="^sip:0[1-9]+@.*") { strip(1); route(5); break; } else { route(4); break; } __________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail

2 9

[Serusers] call transfer without REFER on end gateway
by Iqbal 19 Aug '05

19 Aug '05

Hi I have alittle problem, I am looking at doing call transfers from inbound PSTN to IP phones, who can then transfer across their department, BUT my pstn gateway is currently not supporting REFER. This is what seems to be causing the problem (tks Steve), now if this is the case, can I a) get transfer to work without this b) use asteriisk somehow to take the call inbound into asterisk , and then do transfer Iqbal

1 0

[Serusers] "No memory left" error!
by Federico Giannici 19 Aug '05

19 Aug '05

I need to load in memory a large table (currently about 10.000 entry). During the sql query (MySQL) the following error happens and SER quit! The error is due to the failure of a pkg_malloc(). I tried to increase the PKG_MEM_POOL_SIZE and SHM_MEM_SIZE but nothing changed. What I have to do to succed the large query? SER is 0.9.3 under OpenBSD 3.7 amd64. Thanks. Aug 19 17:56:09 eowyn ./ser[27075]: convert_row(): No memory left Aug 19 17:56:09 eowyn ./ser[27075]: convert_rows(): Error while converting row #7014 Aug 19 17:56:09 eowyn ./ser[27075]: convert_result(): Error while converting rows Aug 19 17:56:09 eowyn ./ser[27075]: store_result(): Error while converting result -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 1

[Serusers] ngrep
by Sebastian Kühner 19 Aug '05

19 Aug '05

Hi! Here is a ngrep of my cancel message of a call that wasn't established (cancel on ring, not while talking): U 2005/08/19 09:45:46.747447 xxx.xxx.xxx.xxx:1024 -> xxx.xxx.xxx.xxx:5060 CANCEL sip:0054261xxxxxxxx@pbx2.test.com:5060 SIP/2.0. Content-Length: 0. Call-ID: 2E863532-7483-4585-B9FA-C6EC3340203B(a)192.168.1.101. Max-Forwards: 70. From: "Administrator"<sip:44441@pbx2.test.com:5060>;tag=257139028539. CSeq: 1 CANCEL. To: <sip:0054261xxxxxxx@pbx2.test.com:5060>. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:1024;rport;branch=z9hG4bKc0a801650131c9b14305d51d00002639000 000b4. User-Agent: SJLabs-SJphone/1.30.252. . # U 2005/08/19 09:45:46.997466 xxx.xxx.xxx.xxx:5060 -> xxx.xxx.xxx.xxx:1024 SIP/2.0 200 ok -- no more pending branches. Call-ID: 2E863532-7483-4585-B9FA-C6EC3340203B(a)192.168.1.101. From: "Administrator"<sip:44441@pbx2.test.com:5060>;tag=257139028539. CSeq: 1 CANCEL. To: <sip:0054261xxxxxxx@pbx2.test.com:5060>;tag=2fb8a6135db5d855a493c61ec9633675 -1e47. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:1024;rport=1024;branch=z9hG4bKc0a801650131c9b14305d51d000026 39000000b4. Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=31972 req_src_ip=xxx.xxx.xxx.xxx req_src_port=1024 in_uri=sip:00542 . Shouldn't that be 486 Request Terminated instead of 200 ok? Thanks!

1 0

Re: [Serusers] User-Agent: Asterisk PBX.
by Iqbal 19 Aug '05

19 Aug '05

and I thought I was privileged :-) Iqbal Juan wrote: >No. You are not the only to receive them. As I don't think I am the only one >receiving the mails from Lazy.charlie. >I would like to see both blocked. >Regards > >Juan > >-----Original Message----- >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On >Behalf Of Iqbal >Sent: Friday, August 19, 2005 11:22 AM >To: fedora(a)voice.dyndns.tv >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] User-Agent: Asterisk PBX. > >am I the only one who gets these messages, if so, although I dont mind being >part of the privileged few...I would prefer t if I didnt get these :-), if >everyone is getting them, mailing list admin, can we delete/remove/ban them > >Iqbal > >fedora(a)voice.dyndns.tv wrote: > > > >># >>U 2005/08/20 00:13:30.111494 221.186.150.67:5060 -> 221.186.150.68:5060 >>BYE sip:8000@221.186.150.68:5060 SIP/2.0. >>Max-Forwards: 10. >>Record-Route: <sip:221.186.150.67;ftag=as1cbee789;lr=on>. >>Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0. >>Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. >>From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. >>To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. >>Contact: <sip:500@221.186.150.67:5062>. >>Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. >>CSeq: 102 BYE. >>User-Agent: Asterisk PBX. >>Content-Length: 0. >>Route: <sip:8000@221.186.150.68:5060>. >>. >> >># >>U 2005/08/20 00:13:30.147446 221.186.150.68:5060 -> 221.186.150.67:5060 >>SIP/2.0 200 Ok. >>Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0. >>Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. >>From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. >>To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. >>Contact: <sip:8000@221.186.150.68:5060>. >>Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. >>CSeq: 102 BYE. >>Server: X-Lite release 1103m. >>Content-Length: 0. >>. >> >># >>U 2005/08/20 00:13:30.149017 221.186.150.67:5060 -> 221.186.150.67:5062 >>SIP/2.0 200 Ok. >>Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. >>From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. >>To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. >>Contact: <sip:8000@221.186.150.68:5060>. >>Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. >>CSeq: 102 BYE. >>Server: X-Lite release 1103m. >>Content-Length: 0. >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >>. >> >> >> >> >> > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > >. > > >

1 0

Re:[Serusers] Centralized routing for managing multiple SER proxies
by vmathur＠transnexus.com 19 Aug '05

19 Aug '05

Hi Ashutosh, Thanks for your response. The readme file of mediaproxy talks about load balancing the RTP streams across multiple proxies. Does the signaling also follow the same path as the media (for the same call). Also, The readme file of the mediaproxy says that: "Once it has determined which domain to use to obtain the list of available proxy servers, it will make a DNS lookup for the following SRV records: _mediaproxy._tcp.selected.domain.name" Does this mean that the DNS lookup serves as the routing mechanism for the call? Also, the document says: "However it should be fairly easy to implement another dispatcher that uses a different mean to determine the proxy server (like using a database mapping), or even to distribute the traffic by other means than SIP domain name." I guess that this means that I could use a 3rd party routing engine to route the calls. Is that correct? If so, is there an interface in the dispatcher that I could use to make the dispatcher talk to my routing DB and not do a DNS lookup. Thanks, Vikrant

1 0

[Users] MediaProxy & PSTN
by Ozan Blotter 19 Aug '05

19 Aug '05

Dear List, At below config. i setted up OpenSER on Public IP and PSTN Gateway is again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL modems on both client side and note that they have SIP ALG, also i've completed port forwarding in each modem, so SIP2SIP call happens (voice traffic goes from UA to UA) without need of anything extra as STUN or RTP RELAY. But since those are very away from each other if a UA needs to call PSTN the voice traffic flows from UA to PSTN and that lowers QoS :( I would like to activate MediaProxy again but just for one case which is PSTN, so whenever a UA needs to call PSTN the voice traffic flows like : UA (ATA) >>> OpenSER + MediaProxy >>> PSTN How can i setup this like above without touching SIP2SIP calls between UA's... or adding the lines beginning with # these to my existing config is enough ? Thanks, Ozan Blotter debug=3 fork=yes log_stderror=no check_via=no listen=212.XXX.104.XXX # This is OpenSER's Public IP port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" # fifo_db_url="mysql://openser:openserrw@localhost/openser" 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/mediaproxy.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # 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") route { 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; }; if (method!="REGISTER") { record_route(); }; if (loose_route()) { route(1); break; }; if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # This is PSTN Gateways's Public IP # use_media_proxy(); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { if (!save("location")) { sl_reply_error(); }; }

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Re: [Serusers] routing by codec
by Daryl Sanders 19 Aug '05

19 Aug '05

SER does not handle the media, so I don't think this possible. Codec negotiation is handled by the gateways. - Daryl On 8/19/05, Daryl Sanders <daryl.sanders(a)gmail.com> wrote: > SER does not handle the media, so I don't think this possible. Codec > negotiation is handled by the gateways. > > - Daryl > > > On 8/18/05, Frank Kostin <frankostin(a)yahoo.com> wrote: > > Hi everybody, > > > > Is it possible routing by Codec type with SER and how would you do that ? > > > > Thanks in advance, > > Frank > > > > ________________________________ > > Do you Yahoo!? > > Read only the mail you want - Yahoo! Mail SpamGuard. > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > >

1 0

[Serusers] Missed BYE
by Ryan Pagquil 19 Aug '05

19 Aug '05

Guys, I'm accounting INVITE and BYE but what I get is INVITE and ACK only, BYE doesn't get in my database. I ngrep 5060 and I saw BYE on the request but it doesn't get accounted. Please Help! This is my ser.cfg... debug=3 fork=yes log_stderror=no listen=202.84.24.107 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" #load module part loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/domain.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/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" #module parameter setup modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) 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("registrar", "nat_flag", 6) modparam("acc", "log_level", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "report_ack", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_flag", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") #our routing logic route { 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#### if (method=="INVITE" && client_nat_test("3")) { record_route_preset("202.84.24.107:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; ###call tear down section### if (method=="BYE" || method=="CANCEL") { end_media_session(); }; ###loose route### 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### if (uri!=myself) { route(2); break; }; if (uri==myself) { if (method=="INVITE" || method=="ACK" || method=="BYE") record_route(); setflag(1); 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); } ##Default message handler## route[1] { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } ##Register message handler## route[2] { sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; 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(); }; } ##INVITE message Handler## route[3] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { 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", "User 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(); }; } Please correct me if any wrong configuration in my config. 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

1 1

[Serusers] PSTN gateway cluster - how to implement a backup procedure (maybe FAQ)
by Pietro Ravasio 19 Aug '05

19 Aug '05

Hi, I'm sorry this is maybe a FAQ, but I can't configure SER in order to statefully forward all method for "PSTN uris" to a backup gateway only when the primary is not reachable or down; I have two gateways: - primary gateway (installed into a remote site, connected to 2 PRIs, IP connected via VPN) - backup gateway (installed locally, connected to a few BRIs) Is there a way to handle "gateway failover" directly into ser? If not possible, I could use DNS round robin, but I'd prefer to handle failover into SER. I've tried with t_on_failure but I can't insert any t_relay logic into onreply_route blocks: .... } else if ((uri=~"sip:0.*")||(uri=~"sip:3.*")) { route(2); } .... route[2] { t_on_failure("2"); prefix("99"); rewritehost("MY.PRIMARY.GW.IP"); t_relay_to_udp("MY.PRIMARY.GW.IP","5060"); } onreply_route[2] { xlog("L_ERR", "Can't connect to primary gateway, redirecting to backup one, uri was %ru"); revert_uri(); xlog("L_ERR", "Uri reverted back to %ru, forwarding to Cisco gw"); t_relay_to_udp("MY.BACKUP.GW.IP","5060"); # <----------- CAN'T DO THIS HERE!!! } Any ideas? Thanks! Pietro -- Ing. Pietro Ravasio - pietro.ravasio(a)abbeynet.it Enterprise Market Manager Abbeynet S.p.A. - http://www.abbeynet.it TEL: +390702109044 GSM: +393282172055 FAX: +390702109055

2 1

[Serusers] 407 Proxy Authentication Required?
by zhaomin 19 Aug '05

19 Aug '05

Hello! I install asterisk and ser in same host, They are working well alone , I want to integrated them together,I used the pstn-gw.cfg from onsip.org .I used the asterisk as a pstn gateway and voicemail.my asterisk's account is 200 201 ,when I dial the number 011201 forward the PSTN gateway ,UA show me error"407 Proxy Authentication Required" How can I solve it? BTW I want to know how does I plan the ser and asterisk dialplan. Ser and asterisk have differ users authentication,How we know them are differ? thanks advance! Zhaomin

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RE: [Serusers] Serweb
by Marc Khayat 19 Aug '05

19 Aug '05

/tmp/ser_fifo/ is wrong since it'll be looking for a folder... Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marc Khayat Sent: Friday, August 19, 2005 11:03 AM To: Karel Kozlik Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Serweb Hi Karel, Thx for bearing with me :) In config_paths.php, I changed it from "/tmp/ser_fifo" to "tmp/ser_fifo/" The errors at the top disappeared but I'm still getting the "FIFO not running or bad path to it" message. Is it wrong to put "tmp/ser_fifo/"? knowing that in ser.cfg, it's /tmp/ser_fifo Thanks again, Marc -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: Friday, August 19, 2005 9:28 AM To: Marc Khayat Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Serweb sorry Marc, haven't idea what may be wrong. Please check again setting of fifo in serweb. Check variable $config->fifo_server in config_paths.php. It's value must be '/tmp/ser_fifo'. Try enable logging in serweb ($config->enable_loging = true in config.php), set log level to debug ($config->log_level = "PEAR_LOG_DEBUG") and examine the log file if there are some messages refered fifo. Also try search for string 'fifo_server' throught log files (or whole serweb) and check if it's value isn't changed somewhere. You should find this string only in file config_paths.php (one times) and in file functions.php (four times). Karel Marc Khayat napsal(a): > Hi Karel, > I already tried this one by manually changing the permissions on > ser_fifo to 666; I just tried adding fifo_mode=0666 to ser.cfg, but > still the same problem. > > Marc > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Friday, August 19, 2005 8:21 AM > To: Marc Khayat > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Serweb > > Hi, > try add: > fifo_mode=0666 > > to your ser.cfg > > Karel > > Marc Khayat napsal(a): > >>/tmp/ser_fifo exists, and is pipe type >>User running Apache is "apache". Owner of ser_fifo is User: "root", >>Group: "root". >>I joined user "apache" to group "root", same. >> >>I created group "ser" and joined "root" "apache" to that group, and >>added fifo_group=ser to the ser.cfg, but still same problem: >> >>Warning: filetype(): Lstat failed for (null) (errno=13 - Permission >>denied) in /var/www/html/serweb/html/functions.php on line 306 >> >>Warning: Cannot modify header information - headers already sent by >>(output started at /var/www/html/serweb/html/functions.php:306) in >>/var/www/html/serweb/html/page.php on line 35 >> >> >>Thanks, >> >>Marc >> >>-----Original Message----- >>From: Karel Kozlik [mailto:karel@iptel.org] >>Sent: Thursday, August 18, 2005 1:38 PM >>To: Marc Khayat >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Serweb >> >>Hello, >>please check if file /tmp/ser_fifo exists and if it's type is pipe. > > Also > >>check rights - if user under which apache is running have access to > > read > >>and write to this file. >> >>Karel >> >>Marc Khayat napsal(a): >> >> >>>The domain thing is working, but problem 2 is still bugging me. >>>Help anyone? >>> >>>Marc >>> >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 11:15 AM >>>To: serusers(a)lists.iptel.org >>>Subject: RE: [Serusers] Serweb >>> >>>Never mind guys... it's working! >>> >>>Marc >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 10:53 AM >>>To: serusers(a)lists.iptel.org >>>Subject: [Serusers] Serweb >>> >>>Hi all, >>>I'm using SER-0.9.3 and its serweb version. >>>I have 2 problems with it: >>>1- On the login page, when I enter the username, it automatically >>>completes it with the FQDN, while I need only the domain name; i.e. I >>>login to http://ser.xyz.com, I enter the username aloha, it's >> >>completed >> >> >>>as aloha(a)ser.xyz.com while I need it as aloha(a)xyz.com. Currently, I'm >>>writing in manually >>> >>>2- When I login, I'm getting an error on the first page only as "FIFO >>>not running or bad path to it". The path to it (in config_paths.php >> >>file >> >> >>>is /tmp/ser_fifo, which is correct), and when I enter the command >> >>serctl >> >> >>>monitor, the counter isn't constant, and same path. >>> >>>Can anyone help me plz, >>> >>>Thanks, >>> >>>Marc >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] Serweb
by Marc Khayat 19 Aug '05

19 Aug '05

Hi Karel, Thx for bearing with me :) In config_paths.php, I changed it from "/tmp/ser_fifo" to "tmp/ser_fifo/" The errors at the top disappeared but I'm still getting the "FIFO not running or bad path to it" message. Is it wrong to put "tmp/ser_fifo/"? knowing that in ser.cfg, it's /tmp/ser_fifo Thanks again, Marc -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: Friday, August 19, 2005 9:28 AM To: Marc Khayat Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Serweb sorry Marc, haven't idea what may be wrong. Please check again setting of fifo in serweb. Check variable $config->fifo_server in config_paths.php. It's value must be '/tmp/ser_fifo'. Try enable logging in serweb ($config->enable_loging = true in config.php), set log level to debug ($config->log_level = "PEAR_LOG_DEBUG") and examine the log file if there are some messages refered fifo. Also try search for string 'fifo_server' throught log files (or whole serweb) and check if it's value isn't changed somewhere. You should find this string only in file config_paths.php (one times) and in file functions.php (four times). Karel Marc Khayat napsal(a): > Hi Karel, > I already tried this one by manually changing the permissions on > ser_fifo to 666; I just tried adding fifo_mode=0666 to ser.cfg, but > still the same problem. > > Marc > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Friday, August 19, 2005 8:21 AM > To: Marc Khayat > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Serweb > > Hi, > try add: > fifo_mode=0666 > > to your ser.cfg > > Karel > > Marc Khayat napsal(a): > >>/tmp/ser_fifo exists, and is pipe type >>User running Apache is "apache". Owner of ser_fifo is User: "root", >>Group: "root". >>I joined user "apache" to group "root", same. >> >>I created group "ser" and joined "root" "apache" to that group, and >>added fifo_group=ser to the ser.cfg, but still same problem: >> >>Warning: filetype(): Lstat failed for (null) (errno=13 - Permission >>denied) in /var/www/html/serweb/html/functions.php on line 306 >> >>Warning: Cannot modify header information - headers already sent by >>(output started at /var/www/html/serweb/html/functions.php:306) in >>/var/www/html/serweb/html/page.php on line 35 >> >> >>Thanks, >> >>Marc >> >>-----Original Message----- >>From: Karel Kozlik [mailto:karel@iptel.org] >>Sent: Thursday, August 18, 2005 1:38 PM >>To: Marc Khayat >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Serweb >> >>Hello, >>please check if file /tmp/ser_fifo exists and if it's type is pipe. > > Also > >>check rights - if user under which apache is running have access to > > read > >>and write to this file. >> >>Karel >> >>Marc Khayat napsal(a): >> >> >>>The domain thing is working, but problem 2 is still bugging me. >>>Help anyone? >>> >>>Marc >>> >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 11:15 AM >>>To: serusers(a)lists.iptel.org >>>Subject: RE: [Serusers] Serweb >>> >>>Never mind guys... it's working! >>> >>>Marc >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 10:53 AM >>>To: serusers(a)lists.iptel.org >>>Subject: [Serusers] Serweb >>> >>>Hi all, >>>I'm using SER-0.9.3 and its serweb version. >>>I have 2 problems with it: >>>1- On the login page, when I enter the username, it automatically >>>completes it with the FQDN, while I need only the domain name; i.e. I >>>login to http://ser.xyz.com, I enter the username aloha, it's >> >>completed >> >> >>>as aloha(a)ser.xyz.com while I need it as aloha(a)xyz.com. Currently, I'm >>>writing in manually >>> >>>2- When I login, I'm getting an error on the first page only as "FIFO >>>not running or bad path to it". The path to it (in config_paths.php >> >>file >> >> >>>is /tmp/ser_fifo, which is correct), and when I enter the command >> >>serctl >> >> >>>monitor, the counter isn't constant, and same path. >>> >>>Can anyone help me plz, >>> >>>Thanks, >>> >>>Marc >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > >

1 0

[Serusers] Re: Document
by Lazy.charles 19 Aug '05

19 Aug '05

1 0

[Serusers] nathelper/rtpproxy + MOH
by harry gaillac 19 Aug '05

19 Aug '05

Hello, Sorry I post the same mail I 've posted a few days ago ! Anybody could explain me how nathelper module/rtpproxy will support music on hold ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Replication problem
by Andreas Granig 19 Aug '05

19 Aug '05

Hi all, we use DNSSRV balancing and forward_tcp() to replicate registrations from one SER to the other SERs in the system. Now when one machine completely crashes, all other SER processes on all other machines hang when processing a REGISTER until tcp-connect times out, leading to a system load of ~16 per machine assuming 16 child processes per SER, and no other messages can be processed. I understand that replicating using UDP would solve this issue, but then replicated registrations get lost every now and then because of unreliable transmission, and as far as I found out t_replicate() can only be used for replicating to *one* other SER. This really gets me thinking about patching out the internal location cache and lookup every location from memory, because this additional lookup really doesn't hurt because of ~10 other DB queries per call. IMHO in systems with more than two SERs this cache is just a big pain. Andy

5 21

[Serusers] User-Agent: Asterisk PBX
by fedora＠voice.dyndns.tv 19 Aug '05

19 Aug '05

BYE sip:8000@221.186.150.68:5060 SIP/2.0 Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK11fd3baa;rport Route: <sip:8000@221.186.150.68:5060> From: <sip:5000@voice.dyndns.tv>;tag=as41dc7302 To: 8000 <sip:8000@voice.dyndns.tv>;tag=3486931228 Contact: <sip:5000@221.186.150.67:5062> Call-ID: CFAB7CE0-4B0D-4892-A3F0-5B0E3F3A9C59(a)221.186.150.68 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0

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