- sr-users - kamailio.org

[Users] migrating DB to OpenSER
by Stefano Capitanio 06 Dec '06

06 Dec '06

Hi, i want to migrate from SER 0.9.4 to OpenSER 1.1.0 and i would like to mantain the MySQL DB actually in use. is it possible? which tables i have to alter and how? thanks, Stefano

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[Users] identifying what an ACK is in response to
by Mark Price 06 Dec '06

06 Dec '06

Hi, How does one determine which part of a SIP conversation an ACK packet is responding to? I would have guessed that it would be in tne CSeq, but that field just says, for example, CSeq: 3 ACK The only method I have come up with so far is looking at the source and destination ports + the destination of the Request-LIne and try to determine the most recent message it might be responding to. Thanks, Mark Price

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[Users] bug in create DB scripts?
by Klaus Darilion 06 Dec '06

06 Dec '06

Hi! I wonder if the creation of the serweb default admin user works. Because the function credentials() is never called inside postgresqldb.sh and mysql.... regards klaus -- Klaus Darilion nic.at

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RE: [Users] Load Balancing
by Watkins, Bradley 06 Dec '06

06 Dec '06

> -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Andreas Granig > Sent: Tuesday, December 05, 2006 8:20 AM > To: Marcos Hack > Cc: Thomas Deillon; users(a)openser.org > Subject: Re: [Users] Load Balancing > > Marcos Hack wrote: > > I don't know these modules, but here we solve this problem using a > > unique VIP (Virtual IP) for all openSER cluster nodes. > > How do you ensure that all messages belonging to one > transaction are balanced to the same physical server? Is > there now SIP support in LVS? > I use UltraMonkey (www.ultramonkey.org), but it's basically a packaging of Linux-HA and LVS. LVS supports connection persistence, and that works very well. The level of granularity of load balancing is on a per-IP basis, not per-transaction (LVS only does L3 load-balancing, not L4-7). But assuming a reasonable value for the persistence timeout, there should be no problems with SIP transactions (and I haven't experienced anything problematic). Ldirectord also supports SIP health checks (they're pretty basic OPTIONS requests, but it's something), and will fail over to working SIP proxies. Of course, in-progress transactions will fail as the OpenSER process on one box doesn't have the stateful knowledge of the other. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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[Users] forwarding requests...
by Frogger 06 Dec '06

06 Dec '06

I am trying to forward requests so that the request is completely gone and the UA is forwarded to the final destination with no signaling from the forwarding system. For example: "5551212@" would be forwarded to 1.1.1.2 but 1.1.1.2 would not get a redirect request from 1.1.1.1 rather, directly from the UA. Any thoughts? Thanks! FR ____________________________________________________________________________________ Access over 1 million songs - Yahoo! Music Unlimited (http://music.yahoo.com/unlimited)

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[Users] Jabber (Authentication to the Jabber server failed)
by Vanderson Rocha 05 Dec '06

05 Dec '06

Hi all, I'm trying to configure jabber module in OpenSER and this is what I get when sending a message from Ekiga to a Jabber user: "0(26030) XJAB:xj_jcon_connect: socket [7] 0(26030) XJAB:xj_worker: auth to jabber as: [teste2] / [xxx] 0(26030) XJAB:xj_worker:26154: Authentication to the Jabber server failed ... 0(26030) XJAB:xj_jcon_disconnect: -----START----- 0(26030) XJAB:xj_jcon_disconnect: socket [7] 0(26030) XJAB:xj_jcon_send_presence: -----START----- 0(26030) INFO: signal 13 received 0(26030) XJAB:xj_jcon_send_presence: Error - presence not sent 0(26030) INFO: signal 13 received 0(26030) XJAB:xj_jcon_disconnect: error closing stream 0(26030) XJAB:xj_jcon_disconnect: -----END----- " "jusers" table has only one entry: juid => 1 jab_id => teste2 jab_passwd => teste2 sip_id => sip:teste2@voip.mydomain ( This is the Ekiga user) type => 0 "openser.cfg" modparam("jabber", "db_url","postgres://openseruser:mypass@192.168.95.78/sip_jab") modparam("jabber", "jaddress", "192.168.95.135") modparam("jabber", "jport", 5222) modparam("jabber", "jdomain", "xmpp.mydomain=*") modparam("jabber", "registrar", "jabber_gateway(a)127.0.0.1") modparam("jabber", "workers", 2) modparam("jabber", "max_jobs", 10) modparam("jabber", "cache_time", 600) modparam("jabber", "delay_time", 90) modparam("jabber", "sleep_time", 20) modparam("jabber", "check_time", 20) Any sugestion? Thanks Vanderson.

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[Users] SER/OpenSER + Asterisk + Queue
by lists＠infoway.net 05 Dec '06

05 Dec '06

We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER ("SER") server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel

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[Serusers] SER/OpenSER + Asterisk + Queue
by lists＠infoway.net 05 Dec '06

05 Dec '06

We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER ("SER") server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel

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RE: [Users] Using Openser's CPL module...
by Marvin Ottley 05 Dec '06

05 Dec '06

Good day Bogdan and Nicolas, Thanks for your prompt response. I tried the recommended command: "openserctl fifo LOAD_CPL sip:user@domain /path/his_cpl_file.xml" And the response I received was: "500 command 'LOAD_CPL' not available." I'm using the release version of openser 1.1. Marvin -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 05, 2006 5:43 AM To: Nicolas Palix Cc: users(a)openser.org Subject: Re: [Users] Using Openser's CPL module... Hi Nicolas, the DB upload must be done via the proxy because the DB contains not only the CPL script in XML format, but also in a pre-compiled format of it in binary format. This pre-compiling is done at upload time. You can take it as limitation, but you also have full liberty to implement any transport mechanism for the CPL scripts (from user to server) (like CGI and HTTP) - but on server side, the DB storage must be done via openser. regarding the docs - you are right, it is a bug there - the module requires a DB module (that's why db_url is mandatory), but this is not properly documented. I will take care of it. thanks and regards, bogdan Nicolas Palix wrote: >Hi all, > >Is there some limitations ? >What about email tag ? >The documentation does mention anything. > >Futhermore, "db_url" is mandatory but DB module are not mentioned as mandatory >in the "Dependencies" section. > >Regards, > >Le mardi 05 décembre 2006 09:14, Bogdan-Andrei Iancu a écrit : > > >>Hi Marvin, >> >>you have two options to upload a CPL script into DB (both require your >>server up and running) >> >>1) use FIFO/MI command "LOAD_CPL" : >>:LOAD_CPL: >> >>username >>cpl_filename >>( openserctl fifo LOAD_CPL sip:user@domain /path/his_cpl_file.xml ) >> >>2) use the REGISTER request >>http://www.openser.org/docs/modules/1.2.x/cpl-c.html#AEN272 >> >>regards, >>bogdan >> >>Marvin Ottley wrote: >> >> >>>Good day all, >>> >>>I'm trying to use the CPL module and I'm having trouble figuring out >>>how to load the cpl xml script into the cpl database. >>> >>>Thanks for your pointers >>> >>>Marvin >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Couple of usagequestions (trunking/alias, )
by Andreas Sikkema 05 Dec '06

05 Dec '06

Hi, I'm running an OpenSER server and I've got a couple of usage questions I'd like some pointers on how I can implement them. - I want to link multiple incoming telephone numbers to one single account so 123456 and 45678 and 999933 can all point to the account customer1 Basically I want to do SIP trunking, but with requiring the CPE to authenticate itself. How do I do this? - I'm now using an Asterisk server to do some incoming call routing (mainly some call forwarding scenarios) and I'd like to remove the Asterisk dependency. How do I retrieve values based on the called destination from a database? I've tried looking at AVPops, but haven't yet found any clear documentation on how to use it. Thanks! -- Andreas Sikkema

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[Serusers] modify the From header
by Zeeshan Sajid 05 Dec '06

05 Dec '06

Is there any way in which i can rewrite the From header efficiently like i want to change this From header From: "Zeeshan"<sip:shani@localhost.localdomain>;tag=4a6dec6f to From: "Zeeshan"<sip:shani@128.242.111.222:5060>;tag=4a6dec6f how can i do that using subst function ???? please help me thanks in advance regards Zeeshan --------------------------------- Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.

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[Users] Load Balancing
by Thomas Deillon 05 Dec '06

05 Dec '06

Hi all, I try to use SER as a load balancer with the dispatcher module. All SIP/RTP request are load balanced and, in a LAN, it's working. With phone with a public address ip, it's work too. But my problem it's with NATed phones... The ser change the source ip address and so, the nat=yes option in asterisk cannot work!! (I make registration on ser servers) I wanted to change the "from header field" in the ser and put the IP source, but it seams impossible. Do you have an idea for my problem? Thanks a lot, Thomas Deillon

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[Users] qop authentication support
by Klaus Darilion 05 Dec '06

05 Dec '06

Hi! Reading openser's docs about qop, it looks like openser supports qop. turning qop on with www_challenge("", "1") activates the qop parameter: WWW-Authenticate: Digest realm="foo.bar", nonce="457553154ed7b9d93effa4118b4fe21f11b7f887", qop="auth". Openser advertises only qop=auth. Looking at the source code I get the impression that auth-int is supported too. How can this be activated? regards klaus -- Klaus Darilion nic.at

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[Users] Openser Clustering ?
by Jengjr Lee 李正之 05 Dec '06

05 Dec '06

Dear All Guru : Is there any experience about LVS , RHCS (RedHat Cluster Suite) , or OpenSSI working with openser ? Or any other hardware L4 switch with openser ? I have tried LVS with openser , but failed. Thanks for your comments. This message (and any attachments) may contain information that is confidential, proprietary, privileged or otherwise protected by law. The message is intended solely for the named addressee (or a person responsible for delivering it to the addressee). If you are not the intended reciptient of this message, you are not authorized to read, print, retain, copy or disseminate this message or any part of it. If you have received this message in error, please destroy the message or delete it from your system immediately and notify the sender. 本郵件(及任何附件)可能載有機密、專有、具有特權或受法律保護的資料，並僅供收件人(或負責將資料遞交給收件人的人士)使用。如閣下不是本郵件的預定收件人，便無權閱讀、列印、保留、複製或傳佈本郵件或其任何部分。如閣下錯誤地收到本郵件，請立即將之銷毀或從閣下的系統中刪除，並通知寄件人

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[Serusers] About field of subscriber
by chungyu 05 Dec '06

05 Dec '06

Hi: I add a field in subscriber and using the field routing process example: field name:mode format :ture or false if mode is true route[2] else route[3] how can write ser'configuration that call field of subscriber? AVP can do? Thank you! changyu 2006/12/04

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[Users] Installing Openser with Multiple Asterisk
by Naveen 05 Dec '06

05 Dec '06

Hi all, Can some one please point me to the doc's that how to install Openser as Central server and multiple asterisks' are as PSTN gateways. Thanks and regards Naveen

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[Serusers] SER Oracle Support
by Abdul Qadir 05 Dec '06

05 Dec '06

Hi All, We are currently using radius server for authentication. Is it possible to use oracle database with SER without using radius server. We have no problem if need to develop our own module for it, but the concern is whether we still be able to do digest authentication if we take out the radius server. Best Regards, Abdul Qadir --------------------------------- Want to start your own business? Learn how on Yahoo! Small Business.

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[Users] TLS
by Sebastian Murray-Roberts 05 Dec '06

05 Dec '06

Hi, This may seem like a silly question, but I would just like to know if any modifications need to be made to the routing block(s) of openser.cfg to allow calls to be established using TLS? I'm running openser v 1.1.0 and I have just setup TLS. I'm using SIPp to test it and it seems to be working as I can register using a SIPp scenario. However when I try and make a call the invites are challenged and authenticated, but once that is complete the invite is not forwarded to the uas. The uas is registered and listening using TLS. The SIPp scenarios also work perfectly if I don't use TLS. Am I missing anything? Many Thanks Sebastian -- Sebastian Murray-Roberts (B.Sc Hons) Server Developer CeDux Technologies Fax: +27 21 409 7050 Mobile: +27 72 183 5998 CONFIDENTIALITY NOTICE: E-MAIL This e-mail is confidential. If you receive it by mistake, please delete it and notify the sender immediately. If you are not the intended recipient, you may not use, disclose, distribute, copy or publish any information contained herein.

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[Users] Certificate failure
by Li Stephen 05 Dec '06

05 Dec '06

I setup openser with TLS, and configured eyebeam as client. I installed rootCA on one laptap and one desktop. Strange thing is the eyebeam on desktop works well, but the same setup (Same setting in eyebeam, same rootCA installed) does not work on Laptop, It always returns 503 certificate failure. The log on eyebeam shows errors below. Any idea what is going on? Thanks, Stephen [06-12-01]10:25:17.611 Error (min) RESIP "Error when verifying server's chain of certificates: self signed certificate in certificate chaindepth=1 /CN=Stephen_Li/ST=IL/C=US/emailAddress=w13738(a)email.mot.com/O=MultiMediaIMS " [06-12-01]10:25:17.612 Error (min) RESIP:TRANSPORT TLS connection failed ok=-1 err=1 error:00000001:lib(0):func(0):reason(1) [06-12-01]10:25:17.612 Error (min) RESIP:TRANSPORT (SSL Error ssl) [06-12-01]10:25:17.634 Error (min) RESIP:TRANSPORT error:14090086:SSL routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed [06-12-01]10:25:17.634 Error (min) RESIP:TRANSPORT Error code = 336134278 file=.\ssl\s3_clnt.c line=844 [06-12-01]10:25:17.634 Error (min) RESIP:TRANSPORT Couldn't TLS connect

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[Users] presence inferamation
by venkid 05 Dec '06

05 Dec '06

Hi, i am new to openser.i am using openser1.1.0.tls . i am getting problem with PA module while try to get presence information.i am using windows messenger as sip client . while am calling handle_subscription("registrar") 8(32566) ERROR: parse_uri: bad uri, state 0 parsed: <> (4) / <> (135224384) error is appears on openser screen . please help me on this .i am attaching openser.cfg also 8(32566) SIP Request: 8(32566) method: <SUBSCRIBE> 8(32566) uri: <sip:ramesh.b@192.168.0.56> 8(32566) version: <SIP/2.0> 8(32566) parse_headers: flags=2 8(32566) end of header reached, state=5 8(32566) parse_headers: Via found, flags=2 8(32566) parse_headers: this is the first via 8(32566) After parse_msg... 8(32566) preparing to run routing scripts... 8(32566) parse_headers: flags=100 8(32566) DEBUG:maxfwd:is_maxfwd_present: value = 70 8(32566) parse_headers: flags=10 8(32566) DEBUG: add_param: tag=4728ca770b8542babb6c8be2e07558d1 8(32566) DEBUG: add_param: epid=7abbed3e7b 8(32566) DEBUG:parse_to:end of header reached, state=29 8(32566) DBUG:parse_to: display={"ramesh.b(a)192.168.0.56"}, ruri={sip:ramesh.b@192.168.0.56} 8(32566) parse_headers: flags=200 8(32566) DEBUG:parse_to:end of header reached, state=10 8(32566) DBUG:parse_to: display={}, ruri={sip:ramesh.b@192.168.0.56} 8(32566) DEBUG: get_hdr_field: <To> [29]; uri=[sip:ramesh.b@192.168.0.56] 8(32566) DEBUG: to body [<sip:ramesh.b@192.168.0.56> ] 8(32566) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> 8(32566) DEBUG: get_hdr_body : content_length=0 8(32566) found end of header 8(32566) find_first_route: No Route headers found 8(32566) loose_route: There is no Route HF 8(32566) grep_sock_info - checking if host==us: 12==9 && [192.168.0.56] == [127.0.0.1] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) grep_sock_info - checking if host==us: 12==12 && [192.168.0.56] == [192.168.0.56] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) grep_sock_info - checking if host==us: 12==9 && [192.168.0.56] == [127.0.0.1] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) grep_sock_info - checking if host==us: 12==12 && [192.168.0.56] == [192.168.0.56] 8(32566) grep_sock_info - checking if port 5060 matches port 5060 8(32566) DEBUG: t_newtran: msg id=5 , global msg id=2 , T on entrance=0xffffffff 8(32566) parse_headers: flags=ffffffffffffffff 8(32566) parse_headers: flags=78 8(32566) t_lookup_request: start searching: hash=20651, isACK=0 8(32566) DEBUG: proceeding to pre-RFC3261 transaction matching 8(32566) DEBUG: t_lookup_request: no transaction found 8(32566) handle_subscription() entered 8(32566) parse_headers: flags=608010 8(32566) check_message -0- _m=0x812fec8 8(32566) check_message -1- 8(32566) check_message -2- accepts_mimes=(nil) 8(32566) check_message -3- *8(32566) check_message -4- parsed_event=(nil) 8(32566) check_message -5- 8(32566) get_pres_uri: _puri= 8(32566) ERROR: parse_uri: bad uri, state 0 parsed: <> (4) / <> (135224384) 8(32566) extract_plain_uri(): Error while parsing URI 6(32559) SIP Request:* 6(32559) method: <SUBSCRIBE> 6(32559) uri: <sip:mahesh@192.168.0.56> 6(32559) version: <SIP/2.0> 6(32559) parse_headers: flags=2 6(32559) end of header reached, state=5 6(32559) 7(32560) SIP Request: 7(32560) method: <SUBSCRIBE> 7(32560) uri: <sip:ramesh.b@192.168.0.56> 7(32560) version: <SIP/2.0> 7(32560) parse_headers: flags=2 7(32560) end of header reached, state=5 7(32560) parse_headers: Via found, flags=2 7(32560) parse_headers: this is the first via 7(32560) After parse_msg... 7(32560) preparing to run routing scripts... 7(32560) parse_headers: flags=100 7(32560) DEBUG:maxfwd:is_maxfwd_present: value = 70 7(32560) parse_headers: flags=10 Thanks and Regards venkatesh.d #This configaration file for routing the failure calls another destination (like VMS) # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/pa.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") modparam("registrar", "default_expires", 1800) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 2) modparam("auth_db","db_url","mysql://openser:openser@venkid/openser") modparam("usrloc","db_url","mysql://openser:openser@venkid/openser") modparam("pa", "db_url", "mysql://openser:openser@venkid/openser") # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS # if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; # } else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # 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("venkid", "subscriber")) { www_challenge("venkid", "0"); exit; }; save("location"); exit; }; if( method=="SUBSCRIBE" ) { route(2); }; if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { log("not found while look up *********************************"); exit; }else{ log("found while look up*************************************"); exit; }; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); } exit; } route[2] { # absorb retransmissions if (! t_newtran()) { sl_reply_error(); exit; }; # append to reply the contact of your server append_to_reply("Contact: <sip:192.168.0.56:5060>\r\n"); handle_subscription("registrar"); sl_send_reply("200","ok "); t_release(); exit; }

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[Users] t_replicate: Invalid nonce balue received
by Mark Price 05 Dec '06

05 Dec '06

Hi, I'm attempting to use t_replicate in order to load balance the registration proxy. However, when the REGISTER packet goes to proxy2, it says "401 Unauthorized". Both openser installations are working off of the same mysql database. The first proxy has a srv record for the domain being served, the second does not. The first openser.cfg looks like this: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voipcall.com", "subscriber")) { www_challenge("voipcall.com", "0"); exit; }; save("location"); t_replicate('proxy2'); exit; }; the second looks like this: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voipcall.com", "subscriber")) { www_challenge("voipcall.com", "0"); exit; }; save("location"); exit; }; Both openser instances have aliases for the domain, and as I said before, are operating off of the same database backend. Furthermore, if I set the sip client's sip domain so that it will register directly with proxy2, the phone registers with now problem. However, when the t_replicate executes, syslog says this: Dec 4 23:26:01 /usr/sbin/openser[15365]: check_nonce(): comparing [4574af56734dbd1d5c2c81242a74ba5cb5703c79] and [4574af56989148c2191ea66c5c3c1d75726da244] Dec 4 23:26:01 /usr/sbin/openser[15365]: pre_auth(): Invalid nonce value received Thanks, Mark

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[Users] BYE from call recipient not passing the firewall
by Mark Price 05 Dec '06

05 Dec '06

Hi, I am experiencing the following: openser 1.1 and asterisk are on public IP addresses, with openser acting as a sip proxy in front of asterisk. sip clients A and B live behind the same firewall. Let A be a TCP UAC. Let B be a UAC using TCP or UDP. If A calls B and B hangs up, A never sees the BYE packet, and never hangs up. If A calls B and A hangs up, the BYE packet is transmitted just fine, and B hangs up as it should. tcpdump at the sip proxy shows that when B hangs up, the SIP hangup dialog is as expected between B<->openser<->asterisk Then, asterisk sends a BYE through openser which is addressed to A, but A never responds. To be more exact, A never responds because the BYE message is not passing through the firewall. Perhaps someone has seen such a problem? I have included a collection of facts below about the situation. If there are others I can provide you with, let me know. If A and B are both TCP, then clients A and B each have exactly one tcp stream between themselves and openser. I.e. tcpdump shows precisely 2 each of syn and syn/ack packets. The source and destination ports of the final BYE packet are the same as the destination and source ports of the original INVITE packet. Furthermore, the time between phone registration and the sending of the last BYE packet is on the order of 20 seconds, and the behavior is consistent, so it is unlikely that the TCP stream timed out. The following four packets were cut and pasted from tcpdump. Would more information be helpful? The following packets represent: 1. invite from asterisk to openser 2. invite from openser to client B 3. bye from asterisk to openser 4. bye from openser to client B Recall that INVITE(tcp.src, tcp.dst) == BYE (tcp.dst,tcp.src). INVITE sip:9043067733@joinuneta.com:5065 SIP/2.0\r\n From: "Mark Price2" <sip:9043060000@66.129.95.24>;tag=as50a4e684\r\n To: <sip:9043067733@joinuneta.com:5065>\r\n Contact: <sip:9043060000@66.129.95.24>\r\n Call-ID: 6e9c8160194ea4eb59afa0da0c22633d(a)66.129.95.24\r\n CSeq: 102 INVITE\r\n INVITE sip:9043067733@66.177.61.238:17522;transport=TLS;rinstance=58c12cb6506504ae SIP/2.0\r\n From: "Mark Price2" <sip:9043060000@66.129.95.24>;tag=as50a4e684\r\n To: <sip:9043067733@joinuneta.com:5065>\r\n Contact: <sip:9043060000@66.129.95.24>\r\n Call-ID: 6e9c8160194ea4eb59afa0da0c22633d(a)66.129.95.24\r\n CSeq: 102 INVITE\r\n BYE sip:9043060000@66.177.61.238:17559 SIP/2.0\r\n From: "9043067733 (Softphone)"<sip:9043067733@joinuneta.com >;tag=as3f8914c4\r\n To: "Mark Price"<sip:9043060000@joinuneta.com>;tag=3c60f44e\r\n Contact: <sip:9043067733@66.129.95.24>\r\n Call-ID: ZDFkMjBlMWJlOGZlZWE4NmZlMzQ2NWE0OWNiOGYzYzU.\r\n CSeq: 102 BYE\r\n BYE sip:9043060000@66.177.61.238:17559 SIP/2.0\r\n From: "9043067733 (Softphone)"<sip:9043067733@joinuneta.com >;tag=as3f8914c4\r\n To: "Mark Price"<sip:9043060000@joinuneta.com>;tag=3c60f44e\r\n Contact: <sip:9043067733@66.129.95.24>\r\n Call-ID: ZDFkMjBlMWJlOGZlZWE4NmZlMzQ2NWE0OWNiOGYzYzU.\r\n Thanks, Mark Price

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[Users] forward_reply error
by johnk＠koreanet.us 05 Dec '06

05 Dec '06

Hi all. After UA makes a PSTN call, if remote PSTN peer hangs up first, server correctly receives BYE from gateway, hit BYE section in config and CDR records accordingly through radius. But if UA hangs up first, BYE section in config never gets hit, instead server logs "ERROR: forward_reply: no 2nd via found in reply" right after " ACC: transaction answered:...", which was logged when the call was answered by called PSTN. CDR accounting only records without STOP time and duration being zero. Any comments will be appreciated.

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[Users] regexp and performance...
by Mahesh Paolini-Subramanya 05 Dec '06

05 Dec '06

Which is more efficient? if (uri=~"^sip:foo.*") { or if (avp_check("$rU", "re/^foo/i")) { And, in the long run, given Amdahl's law, does it even matter? Enquiring minds want to know -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] 302 routing?
by John Peters 05 Dec '06

05 Dec '06

Not sure why this is happening,

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[Users] radius auth change attribute
by tele＠plexia.com 05 Dec '06

05 Dec '06

Hello all, this was already asked on the SER mailinglist without response. is there a way to rewrite radius auth request information packet before sending to radius? i need to change User-Name attribute of the authentication request. reading the source of auth_radius module i think this is not possible right now. the User-Name is filled by the digest credential username + realm. could be usefull to have the possibility of customize this. thank you, :tele

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RE: [Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

no that was the TM module docs I used. It was an sgml file. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 3:34 PM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites ?? Was that a question? Did you read the documentation I referred to? g-) Paul Antinori (pantinor) wrote: I am using the timers as below: modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) There do not seem to be any max count parameters to set according to the TM module docs. Thanks, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:51 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites See the FAQs on iptel.org for docs on 0.9.x timers and What's new in 0.10.x for docs on the new timers in SER Ottendorf. g-) Paul Antinori (pantinor) wrote: Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

I am using the timers as below: modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) There do not seem to be any max count parameters to set according to the TM module docs. Thanks, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:51 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites See the FAQs on iptel.org for docs on 0.9.x timers and What's new in 0.10.x for docs on the new timers in SER Ottendorf. g-) Paul Antinori (pantinor) wrote: Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Hi Steve, Yes I am trying to get SER to resend the INVITE to priority 2 after priority 1 timed out due to connection down scenario. Thank you, Paul -----Original Message----- From: Steve Blair [mailto:blairs@isc.upenn.edu] Sent: Monday, December 04, 2006 9:37 AM To: Paul Antinori (pantinor) Cc: Greger V. Teigre; serusers(a)iptel.org Subject: Re: [Serusers] SRV failover question Paul: Are you trying to get the proxy server to re-send the invite based on the SRV list or the phone? _Steve Paul Antinori (pantinor) wrote: > Greger and all, > > The URL does not really help. You will see my cfg script below is > already using rewritehostport with no port and the SRV name. > I also am not using t_on_failure as it should not be necessary > according the post by Klaus. > > Heres my SRV table with the ranking priorities: > > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 1 > weight = 1 > port = 5060 > svr hostname = vxml-1.pats.cisco.com > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 2 > weight = 1 > port = 5060 > svr hostname = vxml-2.pats.cisco.com > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 3 > weight = 1 > port = 5060 > svr hostname = vxml-3.pats.cisco.com > pats.cisco.com nameserver = ns1.pats.cisco.com > vxml-1.pats.cisco.com internet address = 10.86.129.x > vxml-2.pats.cisco.com internet address = 10.86.129.x > vxml-3.pats.cisco.com internet address = 10.86.129.x > ns1.pats.cisco.com internet address = 10.86.129.x > > > Thanks anyone for any help, > > Paul > > > > ---------------------------------------------------------------------- > -- > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Monday, December 04, 2006 6:41 AM > *To:* Paul Antinori (pantinor) > *Cc:* serusers(a)iptel.org > *Subject:* Re: [Serusers] SRV failover question > > http://lists.iptel.org/pipermail/serusers/2006-October/030813.html > > Paul Antinori (pantinor) wrote: >> >> Hi, >> >> Regarding the SRV query addition in (ser-0.9.7-pre7), when the call >> fails on the first priority destination, it never seems to attempt to >> send the INVITE to the second priority destination in the SRV >> response list. Is there anything else to script in the ser.cfg file >> to get this to work? >> >> Thanks for help, >> >> Paul >> >> [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on >> eth0 >> 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE >> sip:77710107773365@10.86.129.17:5060;transport=udp >> 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- >> your call is important to us >> 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request >> Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK >> sip:77710107773365@10.86.129.17:5060;transport=udp >> 2139 packets captured >> >> [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on >> eth0 >> 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV >> _sip._udp.vxml.pats.cisco.com >> 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response >> SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com >> SRV 3 1 5060 vxml-3.pats.cisco.com >> >> 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A >> vxml-1.pats.cisco.com >> 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A >> 10.86.129.2 <--this is unplugged from the network >> >> >> debug=3 >> fork=yes >> log_stderror=yes >> rev_dns=no >> port=5060 >> children=4 >> check_via=no >> sip_warning=yes >> fifo="/tmp/ser_fifo" >> uid="nobody" >> gid="nobody" >> >> #aliases for this proxy server >> #ie hostnames/domains that it routes for alias=sox.cisco.com >> alias=pats.cisco.com >> >> 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" >> >> modparam("usrloc", "db_mode", 0) >> modparam("rr", "enable_full_lr", 1) >> modparam("tm","fr_timer",5) >> modparam("tm","fr_inv_timer",5) >> >> route{ >> >> >> if (!mf_process_maxfwd_header("10")) { >> sl_send_reply("483","Too Many Hops"); >> break; >> }; >> >> if (uri=~"^sip:1") { >> log(1, "SER found 1*\n"); >> rewritehostport("ccm.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:2") { >> log(1, "SER Found 2*\n"); >> rewritehostport("ccm.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:7") { >> log(1, "SER Found 7*\n"); >> rewritehostport("vxml.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:8") { >> log(1, "SER Found 8*\n"); >> rewritehostport("SER.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:9") { >> log(1, "SER Found 9*\n"); >> >> rewritehostport("ringtone.pats.cisco.com");route(1);break; >> } >> >> log(1,"Could Not Match DN to Route\n"); >> route(1); >> } >> >> route[1] >> { >> if (!t_relay()) { >> sl_reply_error(); >> }; >> } >> >> --------------------------------------------------------------------- >> --- >> >> _______________________________________________ >> 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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[Serusers] need help with AVP module (print_sattr())
by Kostas Marneris 04 Dec '06

04 Dec '06

Hello, I have the following config : loadmodule "/opt/ser-0.9.6/lib/ser/modules/avp.so" and in main route block, after the first Sanity Check Section : uri2attr("ruri"); print_sattr("ruri"); # if (is_sattr_set("ruri")) { xlog("L_NOTICE", "SET"); print_sattr("ruri"); } else { xlog("L_NOTICE", "NOT SET"); }; It seems that 'print_sattr()' simply does not work. On first 'print_sattr("ruri")' command I have nothing at my syslog. On 'if' block, the function 'is_sattr_set()' works ok, I get the "SET" message @ syslog, but not the 'ruri' attribute. Am I miss something ? thanks for any help, Kostas -- e-mail: K.Marneris(a)otenet.gr

2 1

[Serusers] ASterisk and SER
by Arun Kumar 04 Dec '06

04 Dec '06

HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 62222 asterisk passes this is ser and then again ser passes this no 2222 (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun

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RE: [Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Greger and all, The URL does not really help. You will see my cfg script below is already using rewritehostport with no port and the SRV name. I also am not using t_on_failure as it should not be necessary according the post by Klaus. Heres my SRV table with the ranking priorities: _sip._udp.vxml.pats.cisco.com SRV service location: priority = 1 weight = 1 port = 5060 svr hostname = vxml-1.pats.cisco.com _sip._udp.vxml.pats.cisco.com SRV service location: priority = 2 weight = 1 port = 5060 svr hostname = vxml-2.pats.cisco.com _sip._udp.vxml.pats.cisco.com SRV service location: priority = 3 weight = 1 port = 5060 svr hostname = vxml-3.pats.cisco.com pats.cisco.com nameserver = ns1.pats.cisco.com vxml-1.pats.cisco.com internet address = 10.86.129.x vxml-2.pats.cisco.com internet address = 10.86.129.x vxml-3.pats.cisco.com internet address = 10.86.129.x ns1.pats.cisco.com internet address = 10.86.129.x Thanks anyone for any help, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:41 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] SRV failover question http://lists.iptel.org/pipermail/serusers/2006-October/030813.html Paul Antinori (pantinor) wrote: Hi, Regarding the SRV query addition in (ser-0.9.7-pre7), when the call fails on the first priority destination, it never seems to attempt to send the INVITE to the second priority destination in the SRV response list. Is there anything else to script in the ser.cfg file to get this to work? Thanks for help, Paul [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on eth0 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE sip:77710107773365@10.86.129.17:5060;transport=udp 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- your call is important to us 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK sip:77710107773365@10.86.129.17:5060;transport=udp 2139 packets captured [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on eth0 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV _sip._udp.vxml.pats.cisco.com 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com SRV 3 1 5060 vxml-3.pats.cisco.com 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A vxml-1.pats.cisco.com 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A 10.86.129.2 <--this is unplugged from the network debug=3 fork=yes log_stderror=yes rev_dns=no port=5060 children=4 check_via=no sip_warning=yes fifo="/tmp/ser_fifo" uid="nobody" gid="nobody" #aliases for this proxy server #ie hostnames/domains that it routes for alias=sox.cisco.com alias=pats.cisco.com 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" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (uri=~"^sip:1") { log(1, "SER found 1*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:2") { log(1, "SER Found 2*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:7") { log(1, "SER Found 7*\n"); rewritehostport("vxml.pats.cisco.com");route(1);break; } if (uri=~"^sip:8") { log(1, "SER Found 8*\n"); rewritehostport("SER.pats.cisco.com");route(1);break; } if (uri=~"^sip:9") { log(1, "SER Found 9*\n"); rewritehostport("ringtone.pats.cisco.com");route(1);break; } log(1,"Could Not Match DN to Route\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; } ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] use of SIP_DOMAIN
by Zeeshan Sajid 04 Dec '06

04 Dec '06

hey ppl, i new to ser and i am struck at a point. I have 3 local machines and i have set them up as shown dialer -----> ser ---------> pstn gateway when i register to ser i use user at IPaddress:Port and then simply dial a pstn number in the script i am simply forwarding it to a pstn gateway .... Now my question is when SER forwards the request to the PSTN gateway it adds a new via header in which it uses IPaddress:Port whereas i want ser to use the domain that has been specified on the ser machine using export SIP_DOMAIN = "test domain" command First of all can i do that if yes please guide me how to do that ... and how do i make ser use example.domain ... is there any other way ??? I am using the hello world cfg file .... with no db / media proxy support. please correct me and help me thanks in advance Zeeshan --------------------------------- Everyone is raving about the all-new Yahoo! Mail beta.

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RE: [Users] t_replicate to multiple destinations
by Thomas Deillon 04 Dec '06

04 Dec '06

Thanks for your answer. I'm just trying to register a phone to multiple openser, for load balancing, instead of having a single usrloc database. The doc of tm module about t_replicate says that: "No information due the replicated request (like reply code) will be forwarded to the original SIP UAC." So it's seems that there should be no problems about reply code, but if ser makes a fork, only a single t_replicate is possible. What am I missing? > You just have to make a tcpdump and see what happen to be sure. > > Thomas

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RE: [Users] t_replicate to multiple destinations
by Thomas Deillon 04 Dec '06

04 Dec '06

Yes, but as you know, to REGISTER a phone, you have more than one step. Phone -> Asterisk: REGISTER without authentification Phone <- Asterisk: 401 / unauthaurized Phone -> Asterisk: REGISTER with authentification Phone <- Asterisk: 200 (OK) So, if you duplicate a message REGISTER, SER make a fork. The phone has to response to the 3 (if you have 3 asterisk servers) 401 messages (because of the different nonce) with 3 "REGISTER with authentification". All phone are not able to do this... be careful, Thomas Deillon -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Marco Envoyé : lundi, 4. décembre 2006 11:53 À : users(a)openser.org Objet : [Users] t_replicate to multiple destinations Hi all, I read on the docs of the tm module that I can use t_replicate to send, for example, REGISTER to multiple destinations. Can a I replicate to more than one server, such as t_replicate("sip:1.2.3.4") t_replicate("sip:1.2.3.5") t_replicate("sip:1.2.3.6") and so on ? >From the docs it seems possible, but I don't know how to set branches. TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] MSILO question.
by Meidiana Ten 04 Dec '06

04 Dec '06

Hi, Is it true that MSILO module only can handle text message? So what is the usage of voice_silo table that written on message_store.php on serweb then? Thanx Regards, Meidiana --------------------------------- Access over 1 million songs - Yahoo! Music Unlimited.

3 3

[Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul

2 1

[Serusers] Fwd: [Sems] About field of subscriber
by Raphael Coeffic 04 Dec '06

04 Dec '06

---------- Forwarded Message ---------- Subject: [Sems] About field of subscriber Date: Monday 04 December 2006 09:09 From: "chungyu" <chungyu(a)ms11.voip.edu.tw> To: sems(a)lists.iptel.org Hi: I add a field in subscriber and using the field routing process example: field name:mode format :ture or false if mode is true route[2] else route[3] how can write ser'configuration that call field of subscriber? AVP can do? Thank you! changyu 2006/12/04 ------------------------------------------------------- -- Dipl. Inf. Raphael Coeffic iptelorg GmbH Am Borsigturm 11 13507 Berlin Germany rco(a)iptel.org www.iptelorg.com T +49-30-3251-3218 F +49-30-6908-8248 _______________________________________________ Sems mailing list Sems(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/sems

3 2

[Serusers] Call transfer
by jorain 04 Dec '06

04 Dec '06

Hi all, I have some question about call transfer. When I press transfer key follow by the new destination and "send", I can see the REFER message inside ethereal but after that request timeout error(408). Inside REFER message, i can see the refer-to and referred-by values are correct. Do I need to add something inside ser.cfg to enable call transfering? Because I cannot find any sample codings regarding ser call transfer. Thanks for any help. Regards, jorain

2 1

[Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Hi, Regarding the SRV query addition in (ser-0.9.7-pre7), when the call fails on the first priority destination, it never seems to attempt to send the INVITE to the second priority destination in the SRV response list. Is there anything else to script in the ser.cfg file to get this to work? Thanks for help, Paul [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on eth0 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE sip:77710107773365@10.86.129.17:5060;transport=udp 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- your call is important to us 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK sip:77710107773365@10.86.129.17:5060;transport=udp 2139 packets captured [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on eth0 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV _sip._udp.vxml.pats.cisco.com 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com SRV 3 1 5060 vxml-3.pats.cisco.com 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A vxml-1.pats.cisco.com 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A 10.86.129.2 <--this is unplugged from the network debug=3 fork=yes log_stderror=yes rev_dns=no port=5060 children=4 check_via=no sip_warning=yes fifo="/tmp/ser_fifo" uid="nobody" gid="nobody" #aliases for this proxy server #ie hostnames/domains that it routes for alias=sox.cisco.com alias=pats.cisco.com 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" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (uri=~"^sip:1") { log(1, "SER found 1*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:2") { log(1, "SER Found 2*\n"); rewritehostport("ccm.pats.cisco.com");route(1);break; } if (uri=~"^sip:7") { log(1, "SER Found 7*\n"); rewritehostport("vxml.pats.cisco.com");route(1);break; } if (uri=~"^sip:8") { log(1, "SER Found 8*\n"); rewritehostport("SER.pats.cisco.com");route(1);break; } if (uri=~"^sip:9") { log(1, "SER Found 9*\n"); rewritehostport("ringtone.pats.cisco.com");route(1);break; } log(1,"Could Not Match DN to Route\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; }

2 1

[Users] t_replicate to multiple destinations
by Marco 04 Dec '06

04 Dec '06

Hi all, I read on the docs of the tm module that I can use t_replicate to send, for example, REGISTER to multiple destinations. Can a I replicate to more than one server, such as t_replicate("sip:1.2.3.4") t_replicate("sip:1.2.3.5") t_replicate("sip:1.2.3.6") and so on ? >From the docs it seems possible, but I don't know how to set branches. TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain

2 1

[Serusers] Subscriber table decsription
by Kamal.Mann＠t-systems.com 04 Dec '06

04 Dec '06

Hi All I need to know the usage of 'phplib_id, domn & uudi' columns into ser database table subscriber. Please give me your valuable suggestion for the same. Regards Kamal Mann

1 0

[Users] test openser behind NAT not work
by somsak vattanavakin 04 Dec '06

04 Dec '06

Dear All i will setup openser behind NAT like these OPENSER ---> ADSL ROUTER(A) <---> INTERNET <---> ADSL ROUTER(B)<-- UAC(1) <-- UAC(2) for UAC to talk each others in the same network with OPENSER server and try to REGISTER AND INVITE IT OK . but if i use UACs that located behind ADSL ROUTER(B) for REGISTER it OK when UAC(1) INVITE UAC(2) it RING and session established can heard sound but after about 30 second the CALLEE UAC(2) is HANG UP while CALLER UAC(1) still show established i saw some mail said that the problem may occour from ADSL ROUTER ITSELF (AGL problem)or some say that ACK problem. if i connect UAC(1) and UAC(2) through ADSL ROUTER(B) to iptel.org and i use UAC(1) INVITE UAC(2) it work no problem for 30 second disconnection . also if i use UAC(1) and UAC(2) conect to iptel.org through ADSL ROUTER(A) no problem . AT this point can I CONCLUDE THAT BOTH ADSL ROUTER NO PROBLEM FOR SIP CONNECTION. is it TRUE ?? Next i saw from LOG file it seem no ACK MESSAGE and error like these -- LOG FILE ------------- ERROR: tcp_blocking_connect: poll error: flags 18 ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused ERROR: tcpconn_connect: tcp_blocking_connect failed ERROR: tcp_send: connect failed --- for simply NAT testing i use openser.cfg like these ---------- OPENSER.CFG --------- if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("infowavenet.com", "subscriber")) { www_challenge("infowavenet.com", "0"); exit; }; fix_nated_register(); save("location"); exec_msg("cat >> /tmp/test"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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 fix_nated_contact(); if(method == "INVITE") { fix_nated_sdp("3"); } if (!t_relay()) { sl_reply_error(); } else { exec_msg("cat >> /tmp/test");} exit; } ------------ MY QUESTION: 1) can openser server is BEHIND NAT ADSL ROUTER and receive Call from UAC from other NAT ROUTER . 2) if so what the problem that CALLEE HANGUP every 30 seconds ( i use XLITE v.3 for UAC ) 3) if it possible can you fix openser.cfg that work for this situation Best Somsak Vattanavakin

2 4

[Users] two invites, different session description => broken
by John Peters 02 Dec '06

02 Dec '06

Not sure why that's happening. Probably setting canreinvite=no on the asterisk side will eliminate the re-INVITEs as a temporary solution, but still would like to know what is happening... wrote: > Sometimes, a calls b and b hears a, and a hears b for a second but a second > INVITE comes to phone B that causes it to redirect rtp to be point to point. > Sometimes there is no audio. > Sometimes, everything works fine. > At one point, rtp from a was going to asterisk, but asterisk was not sending > the rtp on to b, and b was trying to send traffic point to point.

5 8

[Serusers] forward_reply error
by johnk＠koreanet.us 02 Dec '06

02 Dec '06

Hi all. After UA makes a PSTN call, if remote PSTN peer hangs up first, server correctly receives BYE from gateway, hit BYE section in config and CDR records accordingly through radius. But if UA hangs up first, BYE section in config never gets hit, instead server logs "ERROR: forward_reply: no 2nd via found in reply" right after " ACC: transaction answered:...", which was logged when the call was answered by called PSTN. CDR accounting only records without STOP time and duration being zero. Any comments will be appreciated.

1 0

[Serusers] onreply_route and EXEC module
by Benito Caracuel 02 Dec '06

02 Dec '06

Hello all, I need to execute an exec_msg command inside onreply_route code buy SER doesn't permit this. This is my ser.cfg: route{ ...... if (method=="INVITE"){ .......... t_relay(); route(2); break; }; route[2] { t_on_reply("2"); } onreply_route[2] { if (status=~"200"){ if ( search("msrp:") ){ exec_msg("printenv SRCIP > /tmp/200ok.txt; cat >> /tmp/200ok.txt; /bin/bash /usr/local/ser_snapshot/sbin/controlador2"); replace_all_file("msrp:\/\/[0-9\.:\/;a-z]*\r","/tmp/pathmsrp200ok.txt"); } } } In onreply_route, I need to execute exec_msg to manipulate 200 ok, but SER doesn't permit it. What can I execute exec_msg?? I try to probe this: route[2] { t_on_reply("2"); } onreply_route[2] { if (status=~"200"){ if ( search("msrp:") ){ t_on_failure("2"); } } } failure_route[2] { exec_msg("printenv SRCIP > /tmp/200ok.txt; cat >> /tmp/200ok.txt; /bin/bash /usr/local/ser_snapshot/sbin/controlador2"); replace_all_file("msrp:\/\/[0-9\.:\/;a-z]*\r","/tmp/pathmsrp200ok.txt"); t_relay(); } But never enter in failure route!!! Thank you in advance for any help. _________________________________________________________________ Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo y GRATUITAMENTE. http://www.toolbar.live.com

2 1

[Serusers] Use TCP protocol
by Katty Xiong 01 Dec '06

01 Dec '06

Hi, I am trying to do some tests using TCP protocol with SER as a proxy. SER accepts SIP message through TCP correctly, but after SER accepts SIP messages, SER automatically picks up UDP for transmission. Is there a way to force SER to use TCP for forwarding message? thanks, Joy ____________________________________________________________________________________ Want to start your own business? Learn how on Yahoo! Small Business. http://smallbusiness.yahoo.com/r-index

1 1

[Serusers] Problem with ser-freeradius-mysql
by Lokesh Kumar 01 Dec '06

01 Dec '06

Hello list, I had ser-0.9.6, freeradius-1.1.3 and mysql installed. I am doing AAA for ser by freeradius where freeradius should have to fetch data from default mysql tables setup by freeradius. I am able to do authentication of sip clients without problem, accounting packets also gets logged into mysql database (when group membership checking disabled in ser.cfg).But I am unable to manage with group membership check. I am not sure what should be the AVPs I have to insert into mysql tables for group membership check. I followed various articles about freeradius and ser, and tried all possibilities, but no success. Can anyone give me any clue about that. Below mentioned are some of my conf files. In radiusd.conf I had enabled “sql” and disabled “files” in order to read values from mysql db instead of flat files. Part of ser.cfg for radius group check if (!lookup("location")) { if(uri=~"^sip:0[1-9].*@abc.xyz.pt") { #Towards National PSTN gateway if (!radius_is_user_in("From", "ld")) { sl_send_reply("403", "No permission to call on National numbers..."); break; }; route(4); route(5); break; }; In radcheck table I have 2 entries for each user they are like mentioned below mysql> select * from radcheck; +----+--------------------------------+---------------+----+-------------+ | id | UserName | Attribute | op | Value | +----+--------------------------------+---------------+----+-------------+ | 1 | 69021(a)abc.xyz.pt | User-Password | == | "teste" | 17 | 69021(a)abc.xyz.pt | Auth-Type | := | Digest usergroup table has entry like below mysql> select * from usergroup; +----------------------------+-----------+----------+ | UserName | GroupName | priority | +----------------------------+-----------+----------+ | 69021(a)abc.xyz.pt | ld | 1 | +----------------------------+-----------+----------+ 1 row in set (0.00 sec) And radgroupcheck has entries like mentioned below mysql> select * from radgroupcheck; +----+-----------+--------------+----+-------------+ | id | GroupName | Attribute | op | Value | +----+-----------+--------------+----+-------------+ | 1 | ld | Service-type | == | Group-Check | +----+-----------+--------------+----+-------------+ 1 row in set (0.00 sec) Any clue will be much appreciated.. Thanks Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.15.3/561 - Release Date: 12/1/2006

1 0

[Users] SIP over TCP or TLS - Contact Header - NAT problematic
by Christophe Irles 01 Dec '06

01 Dec '06

Hi, If a SIP device is using TCP, to accept incoming calls it must bind on port 5060. To call someone, this device must create a socket to reach the proxy on port 5060. The port source of this socket cannot be 5060 ! In TCP you cannot bind on a port and in same time using it in another socket. So the port source is generally chosen randomly (XYZ) Now that all sockets and bind are done, what are the correct info used to create the Contact header ? For me, the port used in the contact header in this case must be 5060 since the device accepts incoming calls on this port and never XYZ. But in a NAT context, if this SIP device is behind a NAT, the port 5060 cannot be used to reach it ! It's necessary to reuse the socket previously opensed to reach it. So in conclusion, the Contact header must be set with 5060 by the device and on the proxy side, if the client is detected behind a NAT, this Contact header is modified to used instead XYZ Am I right ? For TLS, the problem is the same but instead of 5060 it's 5061. Thanks, Christophe

2 1

[Users] mysql insert
by Helge Waastad 01 Dec '06

01 Dec '06

Hi, just a little question about mysql insert into location. I might be wrong here, but please be patient whith me. Im not too good to read through the source tree.... :-) Would there be anything wrong with changing the insert statement from insert into..... to: insert into.....on duplicate key update..... (my sipp register stress-test scenario gave me this idea....) br hw

3 3

[Users] Problem Radius Authentication - NO username in DIGEST-URI
by bogos dan-cristian 01 Dec '06

01 Dec '06

Hello Folks, I got some strange issue with RadAuth module. I have a running server which works just great with RadAuth. The problem comes when in the FROM and TO headers are no descriptions. EG: Following request will be properly processed: # INVITE sip:0012345@babble.net SIP/2.0. To: "0012345"<sip:0012345@babble.net>. From: "Some User"<sip:user@babble.net>;tag=5a7e9242. BUT the following one will not: INVITE sip:00123456@babble.net SIP/2.0. To: <sip:0012345@babble.net>. From: <sip:user@babble.net:10101>;tag=1-2963920204. Here is what I will get WRONG in RADIUS Server: Packet-Type = Access-Request Thu Nov 30 22:52:31 2006 User-Name = "user(a)babble.net" Digest-Attributes = "\n\020user(a)babble.net" Digest-Attributes = "\001\014babble.net" Digest-Attributes = "\002*456f61dc0d715ea80fd0ed0431df75b73a4c4092" Digest-Attributes = "\004\020sip:babble.net" Digest-Attributes = "\003\010INVITE" Digest-Response = "c96792da6dfb093215150ccd250fac34" Service-Type = Sip-Session Sip-Uri-User = "user" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Client-IP-Address = 87.102.50.17 The problem is that in the Digest-URI part I will not get the sip:0012345@babble.net, but the sip:babble.net, therefore completly erroneous info. I have checked with xlog the request on openser server just before passing the info to radius_proxy_authorize function, and all the FROM-URI, TO-URI, R-URI avps were just fine. Can somebody from the DEVELOPERS help me? I cannot see other solution, since there is nothing to be tuned in the settings of radauth module. Thank you in advance for your time! Cheers, Dan ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

2 1

[Serusers] Download for later version of SER - "cvs head"?
by Paul Antinori (pantinor) 01 Dec '06

01 Dec '06

I have been using 0.9.6 version of SER that has DNS SRV load balancing, but now want to try the build that has DNS SRV failover/rollover capabilities built in using the SRV priority. I think this is called the "CVS HEAD" build. I cannot see this on the FTP download directories. Is this available? Thank you, Paul

3 2

[Users] Questions about regexp and AVPs
by Mahesh Paolini-Subramanya 30 Nov '06

30 Nov '06

I think I'm missing something pretty obvious here, but is there a reason that avp_check("$avp(tmpVar1)", "re/$avp(myNetwork0)/i") barfs on 1.1.x? I get the error 0(0) ERROR:avpops:fixup_check_avp: regexp operation requires string value 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 3491 In the above, $avp(tmpVar1) == "1.2.3.4" $avp(myNetwork0) == "1.2.3" Now, I'd hoped that I could actually run the following avp_check("$avp(tmpVar1)", "re/^$avp(myNetwork0)/i") but that doesn't work Then, setting $avp(myNetwork0) == "^1.2.3" and running avp_check("$avp(tmpVar1)", "re/$avp(myNetwork0)/i") didn't work either Sooo, whats the solution? avp_check can't have AVPs on both sides of the check? cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

2 2

RE: [Users] regular expression
by Edson 30 Nov '06

30 Nov '06

First thinks first: please don't forget to CC the list.. Any information can be for others help. ;) Well, I would suggest You to make some tests on a command line simulation to be sure what You want to accoplish is even possible. Something like this: # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@" # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]\@" # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]" sip:112345678#@10.10.10.10 # echo "sip:112345678@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]\@" sip:112345678@10.10.10.10 # echo "sip:112345678@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@" sip:112345678@10.10.10.10 # As OpenSER, and SER as origin, use a regular expression as GREP, You could see that it work like You asked for. I also run a test in a SER leaving here. I put this code: } else if (uri=~"^sip:1[0-9]{8}@") { # Test for "1xxxxxxxx" mask numbers strip(9); prefix("600"); route(6); break; } And when I call 112345678 from my Softphone, ngrep shows me this forwarded INVITE: INVITE sip:600@pstn.gw:5060 SIP/2.0..Record-Route: <sip:sip.proxy;ftag=1468362515290;lr=on>..Via: SIP/2.0/UDP sip.proxy;branch=z9hG4bK90b1.5f c55385.0..Via: SIP/2.0/UDP client.ip;received=sip.client;rport=1024;branch=z9hG4bKac1bf8060000010f456ef 12a000028cf00000158..From: "Edson - SJPhone"<sip:8201@sip.proxy>;tag=1468362515290..To: <sip:112345678@sip.proxy>..Contact: <sip:8201@sip.client>..Call-ID: 647DD163-9B1E-45DB-93A2-99 CAC53532B8@client.ip..CSeq: 2 INVITE..Max-Forwards: 16..User-Agent: SJphone/1.61 (SJ Labs)..Content-Length: 217..Content-Type: application/sdp....v=0 ..o=- 3373887402 3373887402 IN IP4 sip.client..s=SJphone..c=IN IP4 sip.client..t=0 0..a=setup:active..m=audio 49186 RTP/AVP 4 101..a=rtpmap:4 G723/8000 ..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11,16.. Which in case is what we are looking for. Identified and translate any 9 digit number, started with '1' to '600'. Hope that this help You understand all the mechanism. Edson. _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: quinta-feira, 30 de novembro de 2006 01:24 To: Edson Subject: Re: [Users] regular expression Hello Edson, On 11/29/06, Edson <4lists(a)gmail.com> wrote: Opppsss.... I think that You make a little typo.... Wouldn't the correct be "^sip:1[0-9]{8}@" to match all numbers with 9 digits initiated by an '1' and followed by any combination of digits (from '0' to '9')? this Regular epression is sip:1[0-9]{8} correct but...... if the number like this ... 112345678# The Red color one must be the digit , not any any special charecter....... According u reg expression .. So,,, i'm sure ..................... like this it must .......... sip:1{8}[0-9}............ Sorry , if i did any thing mistake......excuse me......... Edson _____ From: users-bounces(a)openser.org [mailto: <mailto:users-bounces@openser.org> users-bounces(a)openser.org] On Behalf Of raviprakash sunkara Sent: quarta-feira, 29 de novembro de 2006 11:35 To: Jayesh Nambiar Cc: Users(a)openser.org Subject: Re: [Users] regular expression On 11/29/06, Jayesh Nambiar <voip_freak(a)yahoo.co.in <mailto:voip_freak@yahoo.co.in> > wrote: Hi all, Can someone please help me in formulating a regular expression. I have an expression like this: if(uri=~"^sip:1[0-9]*@) This matches numbers like 15552221212, but also matches number like 15552221212#. when i make it as: if(uri=~"^sip:1[0-9]@*) Use this if (uri=~"^sip:1{8}[0-9]@" ) Again this matches any special characters after the number. I just wanted a regular expression that will match only numbers and no alphabets or special characters. Can someone please help me to formulate a regular expression for this. Thanks in advance. w/regards, Jayesh _____ Find out what India is talking about on - Yahoo! Answers India <http://us.rd.yahoo.com/mail/in/yanswers/*http:/in.answers.yahoo.com/> Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW <http://us.rd.yahoo.com/mail/in/messengertagline/*http:/in.messenger.yahoo.c om> _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com <http://www.hyperion-tech.com> -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Serusers] error while b2b installation
by rupesh boyana 30 Nov '06

30 Nov '06

hi, after downloading latest b2b from vovida (sip-1.5.0.tar.gz), when i tried to install it, got the following error : In file included from Timer.cxx:58: Timer.h: In member function `bool Timer<T>::sleepFor(timeval*)': Timer.h:230: error: `cout' was not declared in this scope Timer.h:230: error: `endl' was not declared in this scope Timer.h: In member function `void Timer<T>::insert(T, int)': Timer.h:267: error: `cout' was not declared in this scope Timer.h:267: error: `endl' was not declared in this scope make[1]: *** [obj.debug.Linux.i686/Timer.o] Error 1 make[1]: Leaving directory `/root/sip-1.5.0/util' make: *** [util] Error 2 Initially i gave ./configure inside the sip-1.5.0 folder. Then when i tried with make b2bua , got the following error and installation stopped. i have ser with mysql,nathelper,rtp proxy. thank you, Rupesh.

2 1

[Users] auth_db subscriber caching
by Klaus Darilion 30 Nov '06

30 Nov '06

Hi! A stupid question: Is the subscriber table also cached inside openser (like location table)? regards klaus -- Klaus Darilion nic.at

2 5

[Serusers] SER/Asterisk inbound problem [corrected]
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 52 AsterB: 51 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9151201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9152205 I have insecure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: < sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15 ;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

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[Serusers] SER/Asterisk inbound problem
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 51 AsterB: 52 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9152201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9151205 I have insecrure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

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RE: [Users] Multiple register message
by Thomas Deillon 30 Nov '06

30 Nov '06

Thanks a lot. It's work. Thomas -----Message d'origine----- De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Envoyé : mercredi, 29. novembre 2006 19:24 À : Thomas Deillon Cc : Klaus Darilion; users(a)openser.org Objet : Re: [Users] Multiple register message Hi Thomas, your script will not correctly work as it is a combination of stateless and stateful processing. t_replicate, t_on_failure() are stateful functions, while forward() does stateless fwd - try t_relay() from TM module. regards, bogdan Thomas Deillon wrote: >Subject: Trying to find a solution to duplicate an incoming REGISTER to >two asterisks boxes. > >1. maybe Asterisk can use the location table of openser too? >2. you can save the contact with save_no_reply and then t_relay to the >asterisk server, or > >1&2 > In fact, SER didn't register phones. It just make forwards > >3. you can use t_replicate ... > > >>I tried but I didn't find the way to do this. >> >> > >I have the configuration below : > > >route{ > > if(method=="REGISTER"){ > t_replicate("192.168.1.119") > forward("192.168.1.120") > } > > ds_select_dst("0", "4"); > forward(); > t_on_failure("1"); >} > >failure_route[1] { > log("next_dst\n"); > ds_next_dst(); >} > > >Thanks a lot for your help. >Thomas > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] how to get source uri?
by jorain 30 Nov '06

30 Nov '06

Hi all, How to get the source uri from exec function? like SIP_USER, SIP_RURI and so on. SIP_HF_FROM give full path, including source username, uri and tag. I just want to get the source uri. Please guide. Thanks for any help. Regards, jorain

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Re: [Users] modifying From/To headers with uac module?
by Mahesh Paolini-Subramanya 30 Nov '06

30 Nov '06

Good to know! Strangely enuff, I got the same response from a handful of people too :-) FWIW, I'm probably still going to mess around with getting uac_replace_to operational, 'cos it does seem to be somewhat useful (especially with some of the stranger carrier requirements that I'm starting to see out there) cheers ----- Original MessagGe ----- From: T.R. Missner <tmissner(a)gmail.com> To: mahesh(a)aptela.com Cc: users openser.org <users(a)openser.org> Sent: Wednesday, November 29, 2006 3:35:47 PM GMT-0600 Subject: Re: [Users] modifying From/To headers with uac module? as a former level3 engineer who wrote some of level3s voip platform parts and current customer i can tell you with certainty that level3 only enforces the e.164 requirement on the from and to header during interop. if you can mock up your headers long enough to pass interop and production turn up you can then safely terminate traffic to l3 without e.164 formating. they will not refuse your calls once you make it to production. of course a better solution may be choosing a meta carrier like bandwidth.com ( where i work ) hope this helps tr On 11/29/06, Mahesh Paolini-Subramanya <mahesh(a)corp.aptela.com> wrote: > FWIW, i've been bemoaning the lack of 'To' rewriting for a while myself. > Level3, in its infinite wisdom, has decided that they need the RURI, From, > and 'To' headers to be in a very specific format (prepended '+', 10 digits, > etc., etc.) > I can get the RURI and 'From' exactly the way they need it, but the 'To' is > definitely beyond my ability. > > Do let me/us know if you intend to go ahead with 'uac_replace_to'. It would > be trememdously useful... > > cheers > ----- Original Message ----- > From: Alan Crosswell <alan(a)columbia.edu> > To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Cc: users openser.org <users(a)openser.org> > Sent: Tuesday, November 28, 2006 3:35:22 PM GMT-0600 > Subject: Re: [Users] modifying From/To headers with uac module? > > Thanks Bogdan. I'll have to double-check with them; They may have only > meant they wanted the From in E.164 so that Caller ID works correctly. > /a > > Bogdan-Andrei Iancu wrote: > > Hi Alan, > > > > my advice is to change the PSTN termination since their service is not > > RFC compliant - TO header has no use in routing, only the RURI being > > used (according to the RFC 3261 ~ 3 years old). > > > > but to answer to your question, yes that is the proper place and the > > mechanism is 95% the same as for FROM hdr. > > > > regards, > > bogdan > > > > Alan Crosswell wrote: > > > >> Hello, > >> > >> I see that UAC allows modifying the From header, but I would also like > >> to modify the To. This is because an ITSP I will be testing with wants > >> the From and To to be written in E.164 form (rewriting the R-URI appears > >> not to be good enough) while my internal registrations use a 5-digit > >> extension. It looks like uac is the right module for this but it > >> appears that it can only modify the From header with uac_replace_from() > >> and uac_restore_from(). Would this be the right place to add > >> uac_replace_to() and uac_resotre_to() as well? > >> > >> /a > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)openser.org > >> http://openser.org/cgi-bin/mailman/listinfo/users > >> > >> > >> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > > -- > ******************************************* > Mahesh Paolini-Subramanya (703) 386-1500 x9100 > CTO mahesh(a)aptela.com > Aptela, Inc. http://www.aptela.com > "Aptela: How Business Answers The Call" > ******************************************* > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] modifying From/To headers with uac module?
by Alan Crosswell 29 Nov '06

29 Nov '06

Hello, I see that UAC allows modifying the From header, but I would also like to modify the To. This is because an ITSP I will be testing with wants the From and To to be written in E.164 form (rewriting the R-URI appears not to be good enough) while my internal registrations use a 5-digit extension. It looks like uac is the right module for this but it appears that it can only modify the From header with uac_replace_from() and uac_restore_from(). Would this be the right place to add uac_replace_to() and uac_resotre_to() as well? /a

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RE: [Users] Multiple register message
by Thomas Deillon 29 Nov '06

29 Nov '06

Subject: Trying to find a solution to duplicate an incoming REGISTER to two asterisks boxes. 1. maybe Asterisk can use the location table of openser too? 2. you can save the contact with save_no_reply and then t_relay to the asterisk server, or 1&2 > In fact, SER didn't register phones. It just make forwards 3. you can use t_replicate ... > I tried but I didn't find the way to do this. I have the configuration below : route{ if(method=="REGISTER"){ t_replicate("192.168.1.119") forward("192.168.1.120") } ds_select_dst("0", "4"); forward(); t_on_failure("1"); } failure_route[1] { log("next_dst\n"); ds_next_dst(); } Thanks a lot for your help. Thomas

2 1

[Users] Re: Billing system with good integration with OpenSER
by M.Emran 29 Nov '06

29 Nov '06

u can check iSoftSwitch.pls visit www.inspiresoftbd.com -- Regards ---------- M Emran InSpiration Software Ltd. E-mail:monemran@gmail.com IM: mohammademran(a)hotmail.com Web: www.inspiresoftbd.com

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[Users] Multiple register message
by Thomas Deillon 29 Nov '06

29 Nov '06

Hi, I'm working on ser and I'm trying to find a solution to duplicate an incoming REGISTER to two asterisks boxes. Do you know how can I do this? Thanks for your help, Thomas

2 1

[Users] OPENSER Load testing
by pardeep rathore 29 Nov '06

29 Nov '06

Hi users; This is my first e-mail to the list as i am new to use openser. I have installed openser V 1.0.0 -Now i have the task for load testing of openser by using SIPp with authentication .I am using MYSQL as a database. I have add some users in the subscriber table .BUT i want to know which module i have to implement so that my openser can responds for INVITES and REGISTER both . I want like like this:SIPpUAC send first REGISTER message to openser , openser replies on the bases if users have entry in the subscriber table.Then INVITE . If any one on the list have already done like this work Kindly share .cfg file or any documentation link . Thanks ans Regards; Pardeep __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] problem after migrating from 0.8.14 to 0.9.7-pre1 (onsip site)
by ram 29 Nov '06

29 Nov '06

Hi all Since long iam running with CDROUTER (0.814) Now i have got another server for testing with new Version of SER So i took the stable version from onsip.org and copied all the users and config files from old server to new server and modified accoring to the new setup when the user not available or busy iam sending to voice mail ( asterisk another server) but i have new problem here with new version of ser is when ever i call from user X to user Y ( once its start ringing) if i disconnect the call on X Side, its going to voice mail. ( and other side iam able to see Y user connection hangup) same setup working with my old server with out any problem, and i ahve changed many things in the config, but no luck, so posting in group for the suggestions. OLD SERVER --------------------- version: ser 0.8.12 (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 1.168 2003/10/12 15:09:08 andrei Exp $ main.c compiled on 19:40:54 Jul 19 2004 with gcc 3.2 ------------------------------------------- NEW SERVER version: ser 0.9.7-pre1 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ main.c compiled on 15:49:21 Nov 3 2006 with gcc 3.4.6 ------------------------------------- here is my config # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-URI", "voicemail")) { setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found route(4); break; }; # if user is on-line and is in voicemail group, enable redirection if (method == "INVITE" || method=="BYE" || method=="ACK" && isflagset(4)) { setflag(1); # for accounting t_on_failure("1"); }; t_relay(); } # ------------- handling of unavailable user ------------------ route[3]{ log(1,"route[3]:no user location: foward to voicemail"); rewritehostport("asteriskip:5090"); t_relay(); } break; } route[4] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber if (!isflagset(4)) { sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; # forward to voicemail now rewritehostport("asteriskip:5090"); t_relay_to_udp("asteriskip", "5090"); } failure_route[1] { revert_uri(); rewritehostport("asteriskip:5090"); append_branch(); t_relay_to_udp("asteriskip", "5090"); } any suggestion will be appriciated Ram

2 1

[Serusers] Multiple Calls
by Abdul Qadir 29 Nov '06

29 Nov '06

How can I prevent SER users to make simultaneous calls using one account. Is there any function which I pass the user id/or some thing and it return true/false based on status if this user is currently making any call. --------------------------------- Sponsored Link Mortgage rates as low as 4.625% - $150,000 loan for $579 a month. Intro-*Terms

4 4

[Serusers] Limiting call duration.
by Ricardo Martinez 29 Nov '06

29 Nov '06

Hello list. I have a quick question. Is there a way to limit the duration of a call using only SER? Thanks! Ricardo.-

3 3

[Serusers] Call forward on no answer in 0.10
by Maciej Żwirski 29 Nov '06

29 Nov '06

Hi, I've been trying to implement call forward on no answer with the new ser 0.10. Now I've the following problem. The call times out as it should, enters the on_failure block, rewrites the address with the one from database and then goes to the standard route blocks. The problem is the last sip message being sent (although t_relay doesn't fail) is the 487 request terminated... Can somebody help me with this? Below are the parts of my ser.cfg involved. Thanks. Maciek route["forward"] { t_on_reply("nat.mangle"); xlog("L_INFO","Ruri: %ru\n"); if(!lookup_user("Request-URI")) { xlog("L_INFO","PSTN\n"); route("PSTN"); return; } xlog("L_INFO","No PSTN\n"); # User found, load profile load_attrs("$tu","t.uid"); if(!lookup_contacts("location")) { sl_reply("483","Temporarily Unavailable"); drop; } if(isflagset("nat.uas")) { $t.nat = true; } else $t.nat = false; route("send"); } failure_route["call_forward"] { if (t_check_status("487")) { drop; } if ($t.call_forward_type) { append_branch(); if (($call_forward_type == "noanswer") && t_check_status("408")) { xlog("L_INFO","Failure\n"); avp_pushto("$ruri","s:call_forward_number_na"); route("forward"); return; } if (($call_forward_type == "onbusy") && t_check_status("486")) { avp_pushto("$ruri","s:call_forward_number_bu"); route("send"); return; } } # end_media_session(); } route["send"] { route("nat.mangle"); t_on_reply("nat.mangle"); if ($record_route) { record_route(); } xlog("L_INFO","Relaying\n"); if (!t_relay()) { sl_reply_error(); drop; } } route["nat.detect"] { if (src_ip == 127.0.0.1) return; if (is_present_hf("^Record-Route:")) return; if (nat_uac_test("3")) { force_rport(); setflag("nat.uac"); $f.nat = true; } else $f.nat = false; return; } onreply_route["nat.mangle"] { if ($t.nat || $tr.nat) { fix_nated_contact(); } if (!$f.nat && (!$t.nat || !$tr.nat)) return; if ($con_realm || $tr.con_realm) return; if (@cseq.method != "INVITE") return; if ((status =~ "(183)|2[0-9][0-9]") && !search("^Content-Length: 0")) { if (!$rtpproxy && !$tr.rtpproxy) { if ($t.nat || $tr.nat) { fix_nated_sdp("8"); return; } else return; } force_rtp_proxy(); } return; } route["nat.mangle"] { if ($f.nat) { if (method == "REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } } if (($tr.rtpproxy || $fr.rtpproxy) && (method == "BYE") && (method == "CANCEL")) { unforce_rtp_proxy(); } else if (method == "INVITE") { if (!$f.nat && (!$t.nat || !$tr.nat)) return; $record_route = true; if ($con_realm || $tr.con_realm) return; if ($f.connectivity_realm == $t.connectivity_realm) { $con_realm = true; return; } if (($f.sym_pass || $fr.comedia) && ($t.nat || $tr.nat) && !$f.nat) { $f.comedia = $f.sym_pass; return; } if (($t.sym_pass || $tr.comedia) && $f.nat && (!$t.nat || !$tr.nat)) { $t.comedia = $t.sym_pass; fix_nated_sdp("8"); return; } if !($tr.rtpproxy || $rtpproxy) { $rtpproxy = true; } force_rtp_proxy(); } return 1; } route["PSTN"] { rewritehost("99.99.99.99"); if (!t_relay()) { sl_reply_error(); drop; } } ---------------------------------------------------------------------- PS. >>> http://link.interia.pl/f19a6

2 3

[Users] regular expression
by Jayesh Nambiar 29 Nov '06

29 Nov '06

Hi all, Can someone please help me in formulating a regular expression. I have an expression like this: if(uri=~"^sip:1[0-9]*@) This matches numbers like 15552221212, but also matches number like 15552221212#. when i make it as: if(uri=~"^sip:1[0-9]@*) Again this matches any special characters after the number. I just wanted a regular expression that will match only numbers and no alphabets or special characters. Can someone please help me to formulate a regular expression for this. Thanks in advance. w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

4 3

[Serusers] route by time of the day?
by Ricardo Martinez 29 Nov '06

29 Nov '06

Hello list. I'm wondering if anyone has implemented a routing rule depending on the time of the day. Is there a way to acomplish this?, maybe using AVP ? or something like that? Hope that someone could help me.. Regards, Ricardo Martinez.-

4 9

[Users] NOTIFY message required fields ??
by Chris Robson 29 Nov '06

29 Nov '06

Is the field "Content-length" a required field in a NOTIFY message? I see that it is part of almost every message from OpenSER except the NOTIFY message. This is causing some issues with Gaim SIMPLe interface which processes the header based on the header string "Content-Length". Note Gaim does not care about the value associated with the header string just the fact that the string "Content-Length" exists. Therefore, since OpenSER isnt sending the string, Gaim is core dumping. FYI on the core dump. Yes, Gaim still has a bug on how it is processing what it sees as unknown SIP messages which actually is causing the core dump, the real problem is the fact that Gaim is processing the header, expecting to see the field "Content-length". So before I report this to Gaim as a bug, I wanted to get insight into whether the field is a required field. Thanks......

4 5

[Serusers] Problem with ser-radius accounting
by Lokesh Kumar 29 Nov '06

29 Nov '06

Hello, I got problem with the radius accounting, I am getting accounting packets but they doesn’t mention start or stop time of call rather all the time saying Acct-Status-Type = Failed Here is one packet Tue Nov 21 17:34:58 2006 Acct-Status-Type = Failed Service-Type = IAPP-AP-Check Sip-Response-Code = 487 Sip-Method = 1 User-Name = "211069021(a)xxx.pt" Calling-Station-Id = "sip:211069021@xxx.pt" Called-Station-Id = "sip:0969830612@xx.pt" Sip-Translated-Request-URI = "sip:09643243612@22.18.31.10" Acct-Session-Id = "597816CF-EFCF-4AD8-B61D-F34301FE0B1B(a)192.168.0.47" Sip-To-Tag = "38CA5520-DF9" Sip-From-Tag = "26232679627249" Sip-Cseq = "2" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "d26959cb062c7786" Timestamp = 1164130498 I looked into the dictionary.ser which is present in /usr/local/etc/ser/dictionary.ser and that dictionary had commented out start and stop attributes. If I uncomment them and comment Failed attribute then ser gets crashed. According to ser-radius how to… .. there should be a dictionary.ser file in /usr/local/etc/radiusclient-ng/ directory, but I tried different versions of radiusclient-ng.. it doesn’t come with any of the version, though documents mention it comes with radiusclient-ng library. Though I tried many options with the dictionary.ser file by including it in radiusclient-ng and in freeradius server, but no success. I followed exactly the ser-radius how to but no luck . infact tried with many other probable configure options but it is just not getting solved. I am using ser-0.9.6 version Please help I will be very thank full Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.11/543 - Release Date: 11/20/2006

2 1

[Serusers] Improper relaying of Invite requests
by Kaushik Kumar 29 Nov '06

29 Nov '06

Hi, I am working on a project which involves testing SER with some dummy clients. I have downloaded and installed SER from iptel.org. I am using the very basic Hello.cfg configuration file available online. I find that when SER gets a INVITE request from the caller, it does not deliver it properly to the client (callee). Some debugging revealed that it relays the request to port 5060 at the client rather than the one which was registered by the callee earlier. I tried to look at the source code of SER and found that the callee's contact information never seem to be looked up at any stage. Is this due to some missing information in my configuration file? Any feedback would be appreciated. Thank you -Kaushik PS: I have attached (inline) the configuration file ser.cfg ------------------------------------ debug=9 fork=yes log_stderror=yes listen=128.42.3.3 port=5060 children=4 check_via=no # added dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/scratch/ser-0.9.7-pre1/modules/sl/sl.so" # Stateless replies loadmodule "/scratch/ser-0.9.7-pre1/modules/tm/tm.so" # Transaction management loadmodule "/scratch/ser-0.9.7-pre1/modules/rr/rr.so" # Routing and record-routing loadmodule "/scratch/ser-0.9.7-pre1/modules/maxfwd/maxfwd.so" # Max fowards check loadmodule "/scratch/ser-0.9.7-pre1/modules/usrloc/usrloc.so" # User location support loadmodule "/scratch/ser-0.9.7-pre1/modules/registrar/registrar.so" # Registrar (requires userloc and sl) loadmodule "/scratch/ser-0.9.7-pre1/modules/textops/textops.so" # Message textual operations modparam("usrloc", "db_mode", 0) # 0 forces registration data to be stored in memory modparam("rr", "enable_full_lr", 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 # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; append_hf("P-hint: userloc applied\r\n"); route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { log(2, "Failed t_relay"); sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; }

2 1

Re: [Users] regular expression
by Jayesh Nambiar 29 Nov '06

29 Nov '06

Hi, Thanks for your reply but it will not match only two digits as I have specified the range( [0-9] ). Actually my primary requirement is it should not match any alphabets or special characters with the numbers. I want that only the number should be matched. w/regards, Jayesh ----- Original Message ---- From: Alexandr Dubovikov <shurik(a)start4.info> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: users(a)openser.org Sent: Wednesday, 29 November, 2006 1:15:51 PM Subject: Re: [Users] regular expression On Wed, Nov 29, 2006 at 12:58:44PM +0530, Jayesh Nambiar wrote: > Hi all, > Can someone please help me in formulating a regular expression. I have an expression like this: > if(uri=~"^sip:1[0-9]*@) > This matches numbers like 15552221212, but also matches number like 15552221212#. > > when i make it as: > if(uri=~"^sip:1[0-9]@*) not really, in this regexp your number can have only 2 digits and first one should ne "1". > Again this matches any special characters after the number. > > I just wanted a regular expression that will match only numbers and no alphabets or special characters. > Can someone please help me to formulate a regular expression for this. if(uri=~"^sip:1[0-9]+@*) > > Thanks in advance. -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

1 0

[Users] Complete Config
by Darryl Ross 29 Nov '06

29 Nov '06

Hey All, I am looking for someone who can give me some assistance with a building an openser configuration that is able to handle the following: 1) Multiple domains. 2) Mediaproxy for all clients (not just clients behind NAT). 3) RADIUS accounting for calls. 4) Least Cost Routing for selecting the cheapest upstream link. 5) Trusted (non-authenticated) upstream providers for receiving calls. 6) Call Redirection (all calls, busy, no answer) with redirected calls billed to the redirecting party. 7) Some network features, like voicemail, talking clock, etc, sent to an Asterisk machine. I currently have a configuration that does (1) through (5), but have had no luck in even getting call redirection to work, let alone getting the billing right. I'm having some issues, like: - Getting two replies to REGISTER messages which causes Asterisk clients to print a message "Got authentication request (401) on unknown REGISTER to <...>" - Sometimes I get a storm of CANCEL or BYE messages when a call is ending. I have absolutely no idea where the messages are coming from, whether its the client or upstream, and what message I'm sending that's causing it. The configuration I have made is just from reading the docs, looking at the howtos and tutorials and from guessing what should be done. I think the problems I'm having are related to not really understanding what should be happening 100% and having a config that is a lot more complicated than it needs to be. Anyone willing to give me some assistance? I can send a password sanitised copy of my current config directly if it would help. Regards Darryl

2 1

[Users] strip
by Sam Hsu 29 Nov '06

29 Nov '06

i use the strip() function to strip the prefix when call out for example, prefix 0 to call out there are my sip invite message ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ INVITE sip:0928117466@202.111.222.76:5060 SIP/2.0 Record-Route: <sip:211.111.222.102;r2=on;lr=on;ftag=694064c4> Record-Route: <sip:211.111.222.102;transport=tcp;r2=on;lr=on;ftag=694064c4> Content-Length: 324 Content-Type: application/sdp Via: SIP/2.0/UDP 211.23.176.102;branch=z9hG4bK4eec.ecf680e4.0;i=1 Via: SIP/2.0/TCP 192.168.123.5:5060;received=220.132.138.7 ;branch=z9hG4bK69486617 To: <sip:00928117466@211.111.222.102> From: "joepass" <sip:joepass@211.111.222.102>;tag=694064c4 Supported: timer Call-ID: 95746504-39004973-1533c430-d8319d29(a)192.168.123.5 CSeq: 26589 INVITE User-Agent: IP SIP Phone/2.1.3 Max-Forwards: 69 Session-Expires: 1800 Allow: UPDATE,INFO,MESSAGE,REFER,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL Authorization: Digest nonce="456ba868fbd62c72ca16fcdd04678168a8fa0683", username="joepass", realm="votel-tech.com", uri=" sip:00928117466@211.111.222.102", response="73a8c869c2a42a12f0d920c2a7d6f068" P-IPRAuth: votel-tech.com Contact: <sip:joepass@220.111.222.7:1070> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ in the example, the real number is 0928117466 add prefix 0, so the final number is 00928117466 I see the INVITE part the id has strip the prefix 0 but To: <sip:00928117466@211.111.222.102> and the Authorization part(uri) still keep on 00928117466 My gateway seems to use this information to call out. So it cause some error respond. If i call to the gateway directly(not through openser with number 0928117466), it works. How can i strip the "To" and "Authorization" part uri. thanks... Sam

5 8

RE: [Users] OpenSER in front of Asterisk
by Thomas Deillon 29 Nov '06

29 Nov '06

Hi, We can use the dispatcher module, but it will only load balance the SIP traffic. Correct? I tried to use this http://mediaproxy.ag-projects.com/ with one ser server in front of two Asterisks. But I guess that this tool will only work with multiple ser servers... Is there a solution to make RTP/SIP load balancing with asterisks ? Thanks for your help, Thomas -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Daniel-Constantin Mierla Envoyé : vendredi, 13. octobre 2006 11:25 À : yusuf Cc : users(a)openser.org Objet : Re: [Users] OpenSER in front of Asterisk Hello, if you want to do load balncing, have a look at dispatcher module, it is intended to do that. Does not mather if behind are asterisks or other boxes. Cheers, Daniel On 10/10/06 10:49, yusuf wrote: > Hi all, > > I just joined this mailing list today :) > I'm sure you guys have heard this question before, but what I want to > do is put OpenSER in front of Asterisk to achieve load balancing. I > realise this entails more than it means, and will probably open up a > world of pain, I just want to know what others experiences of this > are, and any comments. > Also, with 1.1.0 being released, and with this new module called > 'path', will this be of any help > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] how to restrict one sipid login once at the same time
by liu york 29 Nov '06

29 Nov '06

Hello My means is that one sipid cant login again untel the same sipid logout. How to config that? Many thx. Warm regard!! 　　　　　　　　york 　　　　　　　　york(a)goldentek.biz

2 1

[Users] OpenSER Summit 2006: summaries
by Daniel-Constantin Mierla 28 Nov '06

28 Nov '06

Hello, a bit late because of time constraints, and not yet fully complete -- I tried to punctuate the idea behind the talks and presentations at the summit, along with some personal conclusions. It should play along with the slides for those which were not able to attend. We have some short videos, but need to rework the audio. http://openser.blogspot.com/ The posting order is the same as events happened. The afternoon session of the second day will be reviewed soon. Best regards, Daniel

1 0

[Serusers] SER Configuration with Mediatrix/Quintum VOIP Gateways
by Goran Donev 28 Nov '06

28 Nov '06

Can someone please give me an example configuration of how to set up ser to pass calls to pstn gateways? I want to set up SER to be able to pass calls based on certain dial plans to certain gateways I have set up. Please send me links or sample configuration files. It would be greatly appreciated. Thanks

2 1

Re: [Serusers] Re: [Users] TM : retransmission timers
by Jiri Kuthan 28 Nov '06

28 Nov '06

Hi Samuel, thank you for your effort and great beginning. I see too this as a laborous effort but in case you decide to go on, you are most welcome to use the Ottendorf announcement as input. It is not complete, but the major changes are covered (hopefuly :-)) -jiri At 11:34 28/11/2006, samuel wrote: >Hi folks, >I've tried to stay out of this thread but I think it has gone far >beyond technical arguments and I don't think it will help neither SER >or openSER projects. I would just ask to stop this thread before it >get worst or stack to technical discussions (which I'm learning a lot) >and leave personal appreciations aside. > >I still see advantages on each project and depending on the >requirements, one is more suited than the other. I had the hope that >both would merge with the best of them but I see this almost >impossible right now... > >I'll try to summarize differences and I would apreciate *technical* >corrections to it (it's almost impossible to have an update >list....fortunately, i've tried to enumerate diferences in the dev >versions which will soon become stable): >*openSER: >o1)detailled documentation, including howtos >o2)fast reaction in the mailing list from the core developers >o3)incorporate user comments into new features >o4)decentralised developers >o5)improved usrloc (loading on mem, cacheless) >o6)gaining user acceptance->biggest testing comunity > >*SER >s1)improved TCP code >s2)improved timers >s3)easier integration with SEMS/SERWEB > >I have tried to be as opbjective as possible (but everyone knows it's >almost impossible) and it would be great to create a common place to >share experiences with both projects as someone has proposed but I >guess everyone is really really busy right now... > >Samuel. >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 1

[Serusers] B2BUA & Radius
by rupesh boyana 28 Nov '06

28 Nov '06

hi , i have integrated SER with NAT Helper +Media proxy + Mysql +PSTN modules. but i am unable to include B2BUA and Radius modules. can some one please guide me in this issue. moreover when i am calling a PSTN number from my SIP UA, after the call is connected, when the call is disconnected on the PSTN end, the BYE packet is not reaching my SIP UA end.similarly, when the call is cancelled at the PSTN end, it shows still RINGING in the SIP UA end. but when i call from SIP UA-1 to SIP UA-2 (online call) , when the call is terminated on either side, call gets disconnected on both sides. so, can some one throw some light on this ? moreover how r B2B UA and Radius related , i mean how do they communicate with each other? Thank you, Rupesh.

1 0

[Users] HA: Users Digest, Vol 18, Issue 63
by Dmitry Lyubimkov 28 Nov '06

28 Nov '06

We use this billing http://www.netup.biz/ It works good with OpenSER Auth, Start, Stop without writing additional scripts. Lyubimkov Dmitriy --------------------------------------------------------------------- Date: Fri, 24 Nov 2006 12:06:48 +0200 From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> Subject: Re: [Users] Billing system with good integration with OpenSER / Documentations To: Nick De Cristofaro <nickdc(a)link2exchange.com> Cc: users(a)openser.org Message-ID: <4566C438.5020102(a)voice-system.ro> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, OpenSER is very good at generating accounting events (START, STOP). Starting from there you can get CDRs in any format suitable for some free billing systems out there. The CDRs can be generated via simple stored procedure or perl/php/... script. I haven't used any open source billing systems, but there are quite a lot. http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems Cheers, Daniel On 11/09/06 13:28, Nick De Cristofaro wrote: > Hey everyone, > > Good sessions yesterday at the OpenSER summit. I found the > presentations very insightful and I learned a lot of new things. I am > hoping to get some feedback as a good solution for a billing system > that we will need to implement over here in our company that can be > easily interacted through OpenSER. As I am pretty new to this I am > wondering what component pieces should be setup to do this right. > OpenSER as our main SIP router with database access and creating CDRs > or do we need a B2BUA with this? Can anyone recommend some good > starting point to look into or some publishings that would be a good > read to get more insight on building this right. > > Some of the presentations had some good insights on scaling, > redundencies and so forth and we would like to do this right from the > beginning and look at our cost platforms. > > Anyone ever used the Hiper biling system? I saw their exhibit > yesterday and it was pretty nice. > > Thanks > Nick > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > ------------------------------ _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 18, Issue 63 *************************************

3 3

[Users] Building Own DIDs
by openser＠comcast.net 28 Nov '06

28 Nov '06

Anyone have any ideas on how to build your own DIDs? I am currently using didx.net but they're not reliable and want to give it a try myself.

2 1

Re: [Users] NOTIFY message required fields ??
by Norman Brandinger 28 Nov '06

28 Nov '06

Resend to the list > Sip - Specific Event Notification > (http://www.ietf.org/rfc/rfc3265.txt) defines the NOTIFY Method. > Within the above RFC the Content-Length header is designated with a > "t" under the NOT (Notify) column (Section 7.1). > > The "t" designation is defined in the SIP > (http://www.ietf.org/rfc/rfc3261.txt) definition as follows: > > t: The header field SHOULD be sent, but clients/servers need to be > prepared to receive messages without that header field. > > If a stream-based protocol (such as TCP) is used as a > transport, then the header field MUST be sent. > > > Hope this helps > > Regards, > Norm > > Chris Robson wrote: >> >> Is the field "Content-length" a required field in a NOTIFY message? >> I see that it is part of almost every message from OpenSER except the >> NOTIFY message. This is causing some issues with Gaim SIMPLe >> interface which processes the header based on the header string >> "Content-Length". Note Gaim does not care about the value associated >> with the header string just the fact that the string "Content-Length" >> exists. Therefore, since OpenSER isnt sending the string, Gaim is >> core dumping. FYI on the core dump. Yes, Gaim still has a bug on >> how it is processing what it sees as unknown SIP messages which >> actually is causing the core dump, the real problem is the fact that >> Gaim is processing the header, expecting to see the field >> "Content-length". So before I report this to Gaim as a bug, I wanted >> to get insight into whether the field is a required field. >> >> Thanks...... >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >

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[Serusers] print destination uri
by Jose Silva 28 Nov '06

28 Nov '06

Hi, Is there anyway I can print the destination uri from the script? Thanks, Jose Silva

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[Serusers] how to deregister?
by jorain 28 Nov '06

28 Nov '06

Hi all, How to make the sip phone offline(no dial tone) when I stop the ser by command ser.init stop? Because now my sip phone will appear in the Online Table and dial tone on it although i stop my ser, until it reach the expire time. Thanks. Regards, jorain

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[Users] avp_printf error
by Gregoire 28 Nov '06

28 Nov '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello! I am trying to use AVP operation and I experiment some problems. I have a line in my configuration like this: avp_printf("i:45","$dd"); When I restart my OpenSER server, I got this error: /usr/sbin/openser[8023]: xl_parse_item: error - bad parameters /usr/sbin/openser[8023]: ERROR:avpops:fixup_printf_avp: unable to get pseudo-variable in param /usr/sbin/openser[8023]: ERROR: fix_actions: fixing failed (code=-2) at cfg line 240 And the line 240 is the line where I call the avp_prinf() function. I tried with others pseudo-variables but whatever the variable is, I get always the same problem. I am using openser-1.1.0-tls with debian package. Does someone have some help for me? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFa/9yI8gmGeMTr0sRAj2BAKCGmtXvzOi/hy3//l93fBDbyu6CbwCghiuc WgrGqq2Q6HaGed3IALZ0ess= =un7M -----END PGP SIGNATURE-----

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[Users] OpenSER 1.1.x and SEMS
by Scott Yagel 28 Nov '06

28 Nov '06

Hello, We have been using OpenSER quite successfully in the last few weeks to test various SIP applications. I am wanting to also use SEMS, and have been able to get a version of that running with OpenSER, but have problems with the voicemail part of SEMS. Apparently some slight dfferences between SER and OpenSER, such as avp.so, exist. Is there a document somewhere that specifies these differences? Thanks, ===================== Scott Yagel syagel(a)PacketCall.net

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[Users] how to configure openser to finish accounting?
by Jose Gil Navarrete 28 Nov '06

28 Nov '06

Hi My problem is to finish the accounting when some UAC is disconected and no message BYE is send, I can't use the mediaproxy or rtpproxy server. In a previous e-mail I read that is possible to finish the accounting using ping, How I can to configure openser with ping to finish the accounting? --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Users] Can't replace duplicate c= lines in SDP
by Jerome Martin 28 Nov '06

28 Nov '06

Hello everyone, I've already noticed a few posts on that topic, but none has been answered up to resolution of the problem. Here is the issue : I am receiving SDP in INVITEs from a Cisco AS5350 (IOS 12.4) gateway that contain two identical c= lines for the same media. When using force_rtp_proxy, only the last one is replaced by my proxy IP. If I use the mangler module on top of it, then both are replaced, but the one already replaced by force_rtp_proxy is now a concatenation of twice the proxy IP address. Is there a way to deal with this so that both lines are replaced ? Depending on which equipement gets the REQUEST next, either the first or the second line is used, and this leads to RTP proxy problems. Best Regards, Jerome Martin

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Re: [Serusers] Re: [Users] TM : retransmission timers
by Vaclav Kubart 28 Nov '06

28 Nov '06

Spam detection software, running on the system "rat.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Right, these tests were not objective. I don't know really objective tests. But I tried to do them. Please, does anybody has a suggestion what should I set up (config, compilation parameters) to get more objective tests? [...] Content analysis details: (6.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO -2.6 BAYES_00 BODY: Bayesian spam probability is 0 to 1% [score: 0.0000] 3.2 NO_DNS_FOR_FROM DNS: Envelope sender has no MX or A DNS records 2.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [62.245.93.50 listed in dnsbl.sorbs.net] 2.6 RCVD_IN_DSBL RBL: Received via a relay in list.dsbl.org [<http://dsbl.org/listing?62.245.93.50>] 1.9 RCVD_IN_NJABL_DUL RBL: NJABL: dialup sender did non-local SMTP [62.245.93.50 listed in combined.njabl.org] -1.3 AWL AWL: From: address is in the auto white-list

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[Serusers] Re: [Users] TM : retransmission timers
by Vaclav Kubart 28 Nov '06

28 Nov '06

Spam detection software, running on the system "rat.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: On Mon, Nov 27, 2006 at 11:16:01PM +0200, Daniel-Constantin Mierla wrote: > As I can see, you get better and better with openser, maybe you can get > some training so you will be able to configure and tune it properly to > fit your needs and get the appropriate results (googling will reveal > some doing trainings for openser). So which are right, these ones, the > previous ones or the next testing results? You tested something, which > (I suppose) you are very familiar with (ser), against something that you > do not know properly to configure. There are some internals that differ > a lot and may have quite a lot of performance impact. [...] Content analysis details: (6.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO -2.6 BAYES_00 BODY: Bayesian spam probability is 0 to 1% [score: 0.0000] 3.2 NO_DNS_FOR_FROM DNS: Envelope sender has no MX or A DNS records 2.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [62.245.93.50 listed in dnsbl.sorbs.net] 2.6 RCVD_IN_DSBL RBL: Received via a relay in list.dsbl.org [<http://dsbl.org/listing?62.245.93.50>] 1.9 RCVD_IN_NJABL_DUL RBL: NJABL: dialup sender did non-local SMTP [62.245.93.50 listed in combined.njabl.org] -1.3 AWL AWL: From: address is in the auto white-list

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[Users] benefits of ser and openser
by Rao Ramaratnamma 27 Nov '06

27 Nov '06

This was excellent comment, I am surprised people did not answer too much. I thought too that if we take ser codebase, openser documentation, all contributed modules, and ser new configuration, and openser release cycle, we will obtain best of best to our benefit. Are people confused by subject line and is this why there is so little comments for this important discussion? I changed the subject line. Hopefuly people who can assess how difficult taking best-of-best would be will answer for me? rr ----- Original Message ---- From: Andrei Pelinescu-Onciul <andrei(a)iptel.org> To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: Jiri Kuthan <jiri(a)iptel.org>; serusers(a)iptel.org; Kim Il <kim_il_s(a)yahoo.com>; users(a)openser.org Sent: Wednesday, November 22, 2006 6:59:04 PM Subject: Re: [Serusers] Re: Fw: [Users] TM : retransmission timers On Nov 22, 2006 at 12:39, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: [...] > > I'm being told that some other personal affiliated with > >the same company more or less copied-n-pasted TCP code from the allegibly > >discontinued SER to OpenSER. > > That's how open source works. I also copied lots of TLS extensions from > ser to openser, and even extended it. You can also copy my extensions > back to ser ... I would love to see it there :-) I'm sure Jiri has no problem with copying code from ser to openser. We will do the same with some openser modules and we already did with some fixes. The problem here was that it was claimed that ser was dead but in the same time new code commited in ser was added in openser (so the openser history writer was not very well intended). Speaking of copying/porting ser between ser and openser I have a few requests: 1. If someone sends a fix for code taken from ser, which was not modified in openser, please could you also bounce the message to the ser maintainer of that piece of code? If you don't to whom, just send it to me and I will make sure it reaches the right person. I'm asking this because my favourite pass-time is not to scan openser commitlogs for possible fixes to my code (e.g.:http://openser.cvs.sourceforge.net/openser/sip-server/fastlock.h?r1=1.…) and this will help me at a minimum effort for the openser developer part. Of course I think this is true not only for me, but also for other ser and openser developers and I (and I'm sure anybody else on the ser team) will return the favour. 2. Please give proper credits for the code you port from ser. I've saw several times things like: Foo sees something was fixed in ser or a new small feature was added and sends an email to the openser list (specifying that the change/patch comes form ser). The openser developers take the change, but the commit message says something like: fix for XyZ, credits go to Foo (ser is not mentioned at all). It would not only be polite, but it will also help us to filter more easily through the openser fixes. Comming back to the openser - ser performance / features discussion, I think the important part here is how we can improve both sers (and you should take the test results as a bug report and maybe ask yourself how did this go through testing, even old sers were faster then 4000 cps). If you think only of the users, then a combination of ser & openser will be the best version. While we disagree (probably) in many respects, I think there are at least a few clear advantages each version has. For example openser has very impressive documentation (at this time ser is very far away from this point of view) and lots of new modules. ser has also its share of new modules (though I think not so many) and a better/improved "infrastructure" (core, tm and lots of the "base" modules - we are actively improving them). ser code tends to also be more tested and stable (our release policies differ wildly). I've tried looking at openser core & tm commits and I haven't seen any significant changes (and this is not a flamewar attempt). Now the question is why doesn't openser take the core, tm and a big part of the modules from ser completely and concentrate then in adding new modules (which you seem to do anyway)? Does it make sense to re-invent the wheel ? Everybody (me too) suffers at least a little from the not-invented-here syndrome, but if you think of the users and at how much of your time it will save, it will make a lot of sense, even if some compromises would be necessary. I think the advantages will far outweigh the problems. Just think about it, we can concentrate each other on our favourite stuff and we can benefit much more easily from each other's work. Andrei _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

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[Users] siptrace and traced_user
by Mark Price 27 Nov '06

27 Nov '06

The sip_trace table has a field called traced_user. This is undocumented, and I am finding that this field is empty in the table. Is this currently unused, or is there something I need to turn on? Mark

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[Users] Radius accounting: what happens if radius server is down?
by kjcsb 27 Nov '06

27 Nov '06

I currently have (open)SER configured to write accounting records to a radius server. However there is a possibility that the Radius server may become unavailable. What is the recommended way to deal with this scenario? Is there a way to cache accounting records? Or is it best to write records to both flatfile and the radius server? Something like this: modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) Any suggestions would be appreciated. Regards Cameron

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[Users] loose_route() function
by Gregoire 27 Nov '06

27 Nov '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am wondering where I can find more information about the loose_route() function. On Wiki, I can see that it is RFC 3261 compliant and understand what it's the difference between loose routing and strict routing, but in my conf. file I have a test like: if(loose_route()){...} and I 'd like to know what kind of value this function can return or when can it return something else than true? Does someone have some information about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZv/iI8gmGeMTr0sRAgg8AKCA9CGb6hI9FEhdE8BMm5nlnXDKkACgjZBA NFrRWixi2BtEE6e2YrAfg9g= =ucXk -----END PGP SIGNATURE-----

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[Serusers] Contact in 200 OK response to REGISTER
by Stefan Sayer 27 Nov '06

27 Nov '06

Hi, registrar at iptel.org (0.10.99-iptelorg/tekelec-SOPv2) and ser 0.9.6 as well do not return a contact header in 200 response to REGISTER (see trace below), but in rfc (10.3) I read it's a MUST 8. The registrar returns a 200 (OK) response. The response MUST contain Contact header field values enumerating all current bindings. Each Contact value MUST feature an "expires" parameter indicating its expiration interval chosen by the registrar. The response SHOULD include a Date header field. Am I missing something (apart from the contact header) or is my register message not correct? Stefan U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.10.144:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. WWW-Authenticate: Digest realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372". Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4141 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . # U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. Authorization: Digest username="sayer", realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372", uri="sip:iptel.org", response="880ff39073b5eeb3cda68352b78de368", algorithm="MD5". . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.5.101:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4140 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . -- Stefan Sayer Media Services Development iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de

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