- sr-users - kamailio.org

[Serusers] Problem with TCP
by P Jayasree 05 Oct '06

05 Oct '06

Hi, I'm using ser proxy server the latest version 0.9.I'm running ser on ipv6 PC ,listening on ipv6 address whihc i configured in ser.cfg. Using some other clients which are running on ipv6 PC's can able to call from UDP-UDP properly.AS ser runs on both UDP and TCP ,the TCP-TCP calls also should go properly. But the error 'tcp_blocking_connect:(22) Invalid Argument' is coming when the next hop from the proxy is TCP. Kindly support for this problem. Regards, Jayasree The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail immediately and permanently delete the message and any attachments.

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Re: Re: [Users] tls. soket queues not flushed
by serega 05 Oct '06

05 Oct '06

Hello, I'm sorry for a big lag. I test system yesterday on other linux kernel (2.6) and obtain the same results. In this case i suppose, that this problem don't bases on system socket implementation. I tested openser on other numbers of tcp workers and obtain next results: 20 children: 2 sockets don't flushed (all had data on receive and send queues), cpu usage is above 48% on each of 2 openser processes, one of them invokes poll function in tls_server.c, other trying to write into tls socket. 1 child: 1 socket don't flushed (receive queue not empty, send queue empty), cpu usage is 90% on one process. 100 children: no problems, all work correctly 2 children: no problems. >Hello, > >On 09/29/06 16:06, serega wrote: >> Hi Klaus and all other who read this. >> >> It's not a problem of tcp socket - yesterday i rewrite gateway config for fowarding directed to extgateway requests to gateway itself. >> In tcp this problems not occured, but after i switched on tls, i see overfilled queues again. In that case i haven't extgateway, communications with >> remote sockets and etc. Did anybody test openser whith big payload? I think this problem occurs because in my case i have big traffic and several process that simultaniously use tls. Did anybody have the same problem? >> > >If you increase the number of the TLS/TCP workers do you get same >results? In case of TLS, there is more CPU used because of >encryption/decryption. How is the CPU usage in teh system while you are >testing? > >As Klaus said, the management of socket queue is done by OS. > >Cheers, >Daniel > >> >>> Hi Serega! >>> >>> I'm not sure - just a guess. openser uses "worker threads". This worker >>> threads get a SIP message from the listener thread, process them, and >>> then they send them. >>> >>> If for some reason the sending fails - e.g. the gateway sends via >>> TCP/TLS to extgateway and extgateway's receiving queue is full. Then >>> gateway's sending buffer gets full and the threads wait until they can >>> send. Thus, as the threads are all busy, also the receiving queue of >>> gateway gets full. I guess after some time the threads give up sending >>> and will read again. >>> >>> The question is: why does the sending buffer gets full? Where is gateway >>> sending the messages to? >>> >>> The sending queue must not get flushed. The queue is in the OS, not in >>> openser. As TCP guarantees that there is no loss/reordering the TCP >>> stack must not flush the queue. >>> >>> regards >>> klaus >>> >>> serega wrote: >>> >>>> Hi all. >>>> >>>> I have next problem while using openser with tls. System consists of >>>> sip statefull server (in next time, simply, server) and stateless sip >>>> gateway (gateway). Server used for connect to jabber server. Server >>>> contains rewrote jabber module logic and also use presence module. >>>> Gateway used to connect to other sip gateway (extgateway) via tls >>>> protocol. Gateway use 1.1 openser version and doesn’t contain changed >>>> code. When server together with gateway restarts and server in its >>>> database contains above 260 subscriptions (in watcherinfo table) i >>>> have error. Socket used by gateway to connect with extgateway >>>> contains in receive and sent queues a lot of data (above 50kByte on >>>> each sides). This data never flushed out. This happens because after >>>> restart sip server through gateway send notifications to extgateway. >>>> When I attached using gdb to process that send data, I saw that it >>>> was in infinity loop because tls library returns SSL_ERROR_WANT_WRITE >>>> and I think it’s correct because we have overfilled send queue. In >>>> this case I don’t interest why receive queue not empty (I think it >>>> happens because sending process have got block on socket). But I >>>> don’t understand why sent queue not flushed. I test this behavior >>>> using tcp – all was correct. Socket not closed by other side because >>>> this status can be unchanged above one day. Anybody can help me? Why >>>> this happens? >>>> >>>> I use openssl-0.9.8c and redhat os (Linux xdevel1 2.4.21-4.ELsmp #1 >>>> SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux). Computer >>>> have too Pentium 4 processors). >>>> >>>> >>>> Gateway.cfg (sip.qa.hbex.com – gateway address): >>>> >>>> # # $Id: router_qa.cfg,v 1.1 2006/08/02 18:14:34 ilya Exp $ # # >>>> simple quick-start config script # >>>> >>>> # ----------- global configuration parameters >>>> ------------------------ >>>> >>>> debug=9 # debug level (cmd line: -dddddddddd) >>>> log_facility=LOG_LOCAL0 fork=yes log_stderror=yes # (cmd line: -E) >>>> >>>> >>>> check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) >>>> rev_dns=no # (cmd. line: -R) children=20 >>>> fifo="/tmp/openser_fifo" server_signature=no >>>> >>>> fifo_db_url="mysql://root:@localhost/openser" >>>> >>>> listen = udp:sip.qa.hbex.com:5060 #incoming sip server address listen >>>> = tcp:sip.qa.hbex.com:5060 >>>> >>>> #tls address disable_tls = 0 listen = tls:sip.qa.hbex.com:5061 listen >>>> = udp:sip.qa.hbex.com:5061 >>>> >>>> tls_certificate = "/ home/inop/ex.com-cert.pem" tls_private_key = >>>> "/home/inop/ex.com-privkey.pem" tls_ca_list = >>>> "/home/inop/ex.com-calist.pem" tls_require_client_certificate=0 >>>> tls_verify_client=0 tls_verify_server=0 >>>> >>>> # ------------------ module loading >>>> ---------------------------------- loadmodule >>>> "/home/interop/openser/lib/openser/modules/rr.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/xlog.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/textops.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/maxfwd.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/sl.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/mysql.so" loadmodule >>>> "/home/interop/openser/lib/openser/modules/tm_unchanged.so" >>>> loadmodule "/home/interop/openser/lib/openser/modules/usrloc.so" >>>> loadmodule "/home/interop/openser/lib/openser/modules/registrar.so" >>>> >>>> # ----------------- setting module-specific parameters >>>> --------------- modparam("registrar", "default_expires", 120) >>>> modparam("registrar", "use_domain", 1) >>>> >>>> modparam("usrloc", "use_domain", 1) 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) >>>> >>>> # add value to ;lr param to make some broken UAs happy modparam("rr", >>>> "enable_full_lr", 1) modparam("maxfwd", "max_limit", 10) # >>>> ------------------------- request routing logic ------------------- >>>> # main routing logic route{ if (!mf_process_maxfwd_header("10")) { >>>> sl_send_reply("483","To Many Hops"); drop(); }; if (dst_port==5061) { >>>> if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) >>>> { # rewrite destination and forward to jabber (sip server) route(1); >>>> return; }; sl_reply_error(); return; } if (method=="REGISTER") { >>>> xlog("XXX: saving location msg=$mb\n"); if(!save_noreply("location")) >>>> { log("XXX: Error saving location!\n"); sl_reply_error(); } >>>> sl_send_reply("200","OK"); return; } if >>>> (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { >>>> #forward to self route(3); return; } ##forward to ext gateway >>>> route(2); } >>>> >>>> route[1]{ if (method!="MESSAGE" && method!="SUBSCRIBE" && >>>> method!="NOTIFY") { log("XXX: Request not forwarded to sip >>>> server!\n"); sl_send_reply("202","Accepted"); return; } >>>> t_on_reply("1"); record_route(); >>>> >>>> if(!lookup("location")) { if (method!="SUBSCRIBE") { log("XXX: only >>>> subscribes are processed for user that are not registered\n"); >>>> sl_reply_error(); return; } >>>> subst_uri('/(.*)@(.*)/\1#\2*jabber.qa.im.hb.com@jabber01.qa.im.hb.com/ig'); >>>> if(!t_relay()) { log("XXX: error forwarding jabber01...\n"); >>>> sl_reply_error(); return; } else { sl_send_reply("200","OK"); return; >>>> } } else { log("XXX: forwarding to the address of record...\n"); >>>> if(!t_relay()) { log("XXX: error forwarding to address of record >>>> \n"); sl_reply_error(); return; } else { sl_send_reply("200", "OK"); >>>> return; } } >>>> >>>> } >>>> >>>> # forwarding to external gateway route[2]{ log("XXX: rewriting >>>> headers\n"); >>>> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> >>>> if(!forward("tls:43.123.141.166:3000")) { log("XXX: Error forwarding >>>> to external gateway!\n"); sl_reply_error(); } else { >>>> sl_send_reply("200", "Accepted"); }; } >>>> >>>> # forwarding to SELF route[3]{ log("XXX: rewriting headers\n"); >>>> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> >>>> log("XXX: about to forward to self\n"); >>>> if(!forward("tcp:sip.qa.hbex.com:5061")) { log("XXX: Error forwarding >>>> to self!\n"); sl_reply_error(); } else { sl_send_reply("200", >>>> "Accepted"); }; } >>>> >>>> # this is executed for replies onreply_route[1]{ >>>> subst('/^(To:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> >>>> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >>>> route(2); } >>>> >>>> >>>> >>>> _______________________________________________ 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] OpenserAdmin: openser log
by Ronald Wiplinger 05 Oct '06

05 Oct '06

In the log file (OpenserAdmin) I see my registered phones with different addresses. I would like to know what it means: a. public IP with a port number sip:7111@59.124.230.151:65129 b. private IP with a port number sip:7112@192.168.10.75:5060 c. private IP with NO port number sip:7113@192.168.10.45 d. Eyebeam with public IP with port and rinstance (???) sip:7311@203.70.36.126:65321;rinstance=059afa5458d206a7 e. Gizmo phone, secondary number should be 7312. sip:17476063396@proxy01.sipphone.com:5060 d. and e. are on my desktop connected to a ADSL router, which obviously makes the NAT to 203.70.36.126, my current public IP on that line. b and c are also on a private network and connected through the same ADSL router as a. Why a has a public IP, while b and c not? Why have all a port, but c not ??? bye Ronald

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[Users] How to read log files?
by Ronald Wiplinger 04 Oct '06

04 Oct '06

I got two places to look: /var/log/openser and the output of ngrep. I would like to see only one case of one phone / phone call. Is there a tool to filter that out? Has anybody made a transfer from the ngrep to a mysql database? I expect that this would be easier to select single cases. Where are and how must I set-up openser to get CDRs ? bye Ronald Wiplinger

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[Users] uac_redirect
by Andreas Granig 04 Oct '06

04 Oct '06

Hi, Two questions about the uac_redirect module: 1. If I extract contacts from a 302 with two contacts having the same q-value by using get_redirects("1:1"), which one of the contacts is selected? The first one? The last one? 2. If I apply a filter rule like: modparam("uac_redirect", "default_filter", "deny") modparam("uac_redirect", "accept_filter", "sip:.+@foo.bar.com") and I get a 302 with a Contact, say, "sip:foo@some.other.tld", then the module logs this error: "ERROR:uac_redirect:get_redirects: get contact from shm_reply branch 0 failed". I'd say if zero contacts are extracted from branches due to filters, it's not really an error. Any suggestions for a better error handling? Andy

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[Users] check_route_param and rr_preset
by Helge Waastad 04 Oct '06

04 Oct '06

Hi, I was just wondering if the check_route_param("") works on record_route_preset Routes? It never hits when my Route: are, f.ex like xxx.xxx.xxx.xxx:<port>;nat=yes Is this true? br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

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[Serusers] re: How to change the To/From headers?
by lawrence k.y. lin 04 Oct '06

04 Oct '06

Hello, I looked at the FAQ in iptel.log and did a search on it for "how to change the To/From headers?" but did not find the answer. There is an item in FAQ for "Why not change the To/From headers?" in the following url "http://www.iptel.org/FAQ_To_From_change", but did mention how to change the To/From headers. Are there variables related to To/From headers that we can change ? Please advise, thanks, Larry _________________________________________________________________ Be seen and heard with Windows Live Messenger and Microsoft LifeCams http://clk.atdmt.com/MSN/go/msnnkwme0020000001msn/direct/01/?href=http://ww…

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[Serusers] blocking CID continuing using radius authentication
by Rosario Pingaro 04 Oct '06

04 Oct '06

I would like to use the ATAs features to block the CID for anonymous calls. At the same time I wouldn't like to break my accounting done by Radius Authentication. This is the ser.cfg about radius auth for international calls: if (!radius_is_user_in("From","int")) { sl_send_reply("403","Internazionali non abilitate"); break; I use to authenticate the "From" field so if I disable the CID on the ata I will receive an anonymous from field. What do I have to authenticate to mantain a working accounting? Do I have to authenticate against "Credentials"? What is the difference between Credentials and from headers? Regards Rosario

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[Users] Group creation with openserctl
by Antoine Fressancourt 04 Oct '06

04 Oct '06

Hello, I am somehow trying to add a user to a group, with the openserctl command : openserctl acl grant 5220 mitel I get a response : WARINIG: Invalid privilege: acl 'mitel' ignored I tryed to browse the web about that issue and about group creation without success, so I would be really pleased to hear from your advices. By the way, "bruteforcing" the addition of the user in the group table is working. Thank you, Antoine

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[Users] Mediaproxy
by Lucio Maciel 04 Oct '06

04 Oct '06

Hi How many simultaneous calls mediaproxy.py can handle with a good server (Dell poweredge 6850, dual XEON, lots of memory and bandwitch) ? Lúcio Fernando Maciel Telecom South America R. Ramiro D'Ávilla, 20 sala 306 Porto Alegre - RS - Brasil lucio(a)tesatelecom.com +55 (51) 3061-5569 +55 (51) 8442-1079

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Re: [Users] ERROR 1062 (23000) at line 1: Duplicate entry '' for key 'phplib_id'
by Bogdan-Andrei Iancu 04 Oct '06

04 Oct '06

maybe we should add for the future a detection of the md5 utility to avoid this kind of problems regards, bogdan Lindsey Hans Joseph Lao wrote: > it worked! :-) thanks a lot! > Bogdan-Andrei Iancu wrote: > >> you can run "MD5=md5 openserctl ...." or for a permanent change see >> in openserctl.base file the MD5 definition. >> >> regards, >> bogdan >> >> Lindsey Hans Joseph Lao wrote: >> >>> i have a utility called md5 installed in my freeBSD system. Can you >>> show me the way on how to integrate it with openserctl? >>> >>> Thanks! >>> >>> Bogdan-Andrei Iancu wrote: >>> >>>> from 1.0.x to 1.1.x there was a change in how the phplib_id is >>>> computed..now is done via md5sum to avoid any shell compatibility >>>> problems (as before). >>>> >>>> check if you have the md5 utility on your system (typically is >>>> md5sum) or check if it has a different name than "md5sum" >>>> >>>> regards, >>>> bogdan >>>> >>>> >>>> Lindsey Hans Joseph Lao wrote: >>>> >>>>> Indeed, it has an empty value in the table... i am unable to add >>>>> subscribers because the error shown below occurs. this happened >>>>> when i switched from openser1.0.1 no tls to openser 1.1.0-tls. >>>>> Here's what happens when I try to add a user to the database: >>>>> >>>>> md5sum: not found >>>>> md5sum: not found >>>>> md5sum: not found >>>>> >>>>> then it prompts for my mysql password. I don't think I have >>>>> installed those to binaries. However, when using openser1.0.1. I >>>>> couldn't encountering this error... >>>>> Probably a bug? >>>>> >>>>> Thanks! >>>>> >>>>> Bogdan-Andrei Iancu wrote: >>>>> >>>>>> ok- it must be the phplib_id column - can you check its value in >>>>>> the table? is must not be NULL or empty... >>>>>> >>>>>> I suspect your openserctl fails to compute the phplib_id and you >>>>>> get a constant empty string for all subscribers.. do you get any >>>>>> errors? do you have "md5sum" and "awk" installed on your machine? >>>>>> >>>>>> regards, >>>>>> bogdan >>>>>> >>>> >>> >> >

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[Users] ERROR 1062 (23000) at line 1: Duplicate entry '' for key 'phplib_id'
by Lindsey Hans Joseph Lao 04 Oct '06

04 Oct '06

Hello, I happen to encounter this error in Openser 1.1.0 with TLS: ERROR 1062 (23000) at line 1: Duplicate entry '' for key 'phplib_id' \E[37;31mERROR: introducing the new user '1050' to the database failed This happens when I try to add users to the mysql database using openserctl add username password email-address. Are there any workarounds for this? or is this a bug in the recent version? Thanks!

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Re: [Serusers]More than one domain
by Ricardo Carvalho 04 Oct '06

04 Oct '06

Hi Kamal, modparam("usrloc", "timer_interval", 30) #Number of seconds between two timer runs. The module uses timer to delete expired contacts, synchronize with database and other tasks, that need to be run periodically. In case of crash or restart contacts that are in memory only and haven't been flushed yet will get lost. If you want minimize the risk, use shorter timer interval. Default value is 60. That description can be seen in the README file located at ser_source_dir/modules/usrloc/. Regards, Ricardo. Kamal.Mann(a)t-systems.com wrote: > Hi Ricardo > It's working now. Thanks a lot. Will you please specify the meaning of > 'modparam("usrloc", "timer_interval", 30)'. > > Thanks in anticipation > Kamal Mann > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ricardo Carvalho > Sent: Wednesday, October 04, 2006 2:42 PM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers]More than one domain > > You need to edit ser.cfg with the following parameters: > > modparam("usrloc", "use_domain", 1) > modparam("registrar", "use_domain", 1) > modparam("usrloc", "timer_interval", 30) > > and you have to live blank the domain in the www_authorize and > www_challenge parameters so that it will work for every domain. In the > domain table you have to add every domain served too. Also add the > alias=every_domain_you_want_ser_to_listen_to in ser.cfg file. > > > Regards, > Ricardo. > > > > > > > > > Kamal.Mann(a)t-systems.com wrote: > >> Hi all >> >> I need to know the changes in configuration of ser if I want that a >> single server can serve more than one domain. >> >> Right now I am changing in following lines but it server only one >> domain at a time: >> >> >> >> /if (method=="REGISTER") {/ >> >> / / >> >> / if (!www_authorize("ims.de", "subscriber")) / >> >> / {/ >> >> / www_challenge("ims.de", "0");/ >> >> / break;/ >> >> / };/ >> >> / / >> >> / save("location");/ >> >> / break;/ >> >> / };/ >> >> >> >> Thanks in anticipation >> >> Kamal Mann >> >> >> > ------------------------------------------------------------------------ > >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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Re: [Users] rtpproxy has been disabled temorarily
by Bogdan-Andrei Iancu 04 Oct '06

04 Oct '06

Ronald, you need to set also the "-l" parameter (listening address). regards, bogdan Ronald Wiplinger wrote: > taipei:~ # rtpproxy -s udp:xxx.xxx.xxx.xxx:22222 > rtpproxy: explicit binding address has to be specified in UDP command > mode > > ??? but it is anyway defined: > > taipei:~ # grep 2222 /home/ronald/downloads/rtpproxy/rtpp_defines.h > #define CPORT "22222" > > What do I still miss? >

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[Serusers] How to keep track of currently active calls
by Ricardo Carvalho 04 Oct '06

04 Oct '06

Hi, As much as I know, active_sessions table exists to keep track of currently active web sessions, for use by serweb. Although, I don't have Serweb installed because I'm building my own web interface so I can't see those active sections looking at that table. Anyway, when I "select * from active_sessions", it doesn't retrieve nothing... I guess even with Serweb it wouldn't show anything because this table is always empty!! What may I be doing wrong? How may I enable active calls to be shown in that table? Regards, Ricardo.

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[Users] SSL: connect failed ..Please help
by Ferianto siregar 04 Oct '06

04 Oct '06

Dear all, Thank you very much for you time . All, I am sorry. May I ask you again, please..? I got error message in minisip command prompt when I tried using TLS (Transport Method = TLS and Ntwork Port = 5061).But, without TLS , I can make a call with minisip. (User`s registration success) The error message says : A. in client command prompt init 6/9: Creating MSip SIP stack init 7/9: Connecting GUI to SIP logic init 8.2/9: Starting TCP transport worker thread init 8.3/9: Starting TLS transport worker thread init 9/9: Registering Identities to registrar server Registering user didit(a)202.95.149.251 to proxy 202.95.149.251, requesting domain 202.95.149.251 SipMessageTransport: sendMessage: creating new socket Creating new SSL_CTX SSL: connect failed SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: creating new socket SSL: connect failed SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: creating new socket SSL: connect failed SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: creating new socket SSL: connect failed SipMessageTransport: sendMessage: exception thrown! B. in server terminal tls_tcpconn_init: Setting in ACCEPT mode (server) 11(5927) tcpconn_add: hashes: 835, 11 11(5927) handle_new_connect: new connection: 0x422d88f0 24 flags: 0002 11(5927) send2child: to tcp child 0 7(5919), 0x422d88f0 7(5919) received n=4 con=0x422d88f0, fd=20 7(5919) DBG: io_watch_add(0x80ed320, 20, 2, 0x422d88f0), fd_no=1 7(5919) tls_update_fd: New fd is 20 7(5919) tls_accept: Error in SSL: 7(5919) tls_error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number 7(5919) DBG: io_watch_del (0x80ed320, 20, 1, 0x10) fd_no=2 called 7(5919) releasing con 0x422d88f0, state -2, fd=20, id=11 7(5919) extra_data 0x422e8a08 11(5927) handle_tcp_child: reader response= 422d88f0, -2 from 0 11(5927) tcpconn_destroy: destroying connection 0x422d88f0, flags 0002 11(5927) tls_close: Closing SSL connection 11(5927) tls_update_fd: New fd is 24 11(5927) tls_shutdown: Shutdown successful 11(5927) tls_tcpconn_clean: Entered What`s wrong? How to solve the error "SSL3_GET_RECORD:wrong version number" and "SSL: connect failed"? Please forgive me for my lack of skill and many questions Please tell me if I have shown my openser.cfg file. Thanks Thank you for your attention and Have a nice day :) Regards, Ferianto Note: 1. I use Redhat 9 [root@localhost root]# rpm -qa|grep -i ssl perl-Crypt-SSLeay-0.45-7 openssl-devel-0.9.7a-2 openssl-perl-0.9.7a-2 openssl-0.9.7a-2 pyOpenSSL-0.5.1-8 mod_ssl-2.0.40-21 openssl096b-0.9.6b-3 docbook-style-dsssl-1.76-8 openssl096-0.9.6-15 [root@localhost root]# --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates. --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

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RE: [Serusers] Presence
by senthil.nachimuthu＠bt.com 04 Oct '06

04 Oct '06

Hi Javier, Thanks for your information. I need to make SEP to send/proxy the message to my presence server. SEP and presence server are physically different entities. Would be of greater help if you could let me know how can i specify the IP address of my presence server in SEP so that the presence specific messages reaching SEP gets forwarded to my presence server. Thanks and Regards, Senthil Kumar, ________________________________ From: Javier Ramirez [mailto:jvr_78@yahoo.com.ar] Sent: 25 September 2006 15:03 To: Nachimuthu,SK,Senthil,XSD4C C Subject: Re: [Serusers] Presence if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); }; handle_subscription("registrar"); }; if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; ----- Original Message ----- From: senthil.nachimuthu(a)bt.com To: serusers(a)lists.iptel.org Sent: Monday, September 25, 2006 8:28 AM Subject: [Serusers] Presence Hi, I want to configure SER <file://ser/about> 0.9.6 in such a way that It can proxy presence messages(SUBSCRIBE,PUBLISH) to presence server. Can you let me know how can I do that? Thanks and Regards, Senthil Kumar, Presence Capability, BT-Service Execution Platform, B62-MH,Orion Building,Adastral Park,Ipswich Work:+44 (0)1473 605 668 Mobile:+44 (0)7772287447 __________ NOD32 1.1768 (20060922) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers __________ NOD32 1.1768 (20060922) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

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[Users] rtpproxy has been disabled temorarily
by Ronald Wiplinger 04 Oct '06

04 Oct '06

If I use # -- nathelper -- modparam("nathelper", "rtpproxy_sock", "udp:59.124.210.41:22222") # default port 22222 #modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:59.124.210.41") # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- route(2); route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } I get in the log file: Oct 4 09:19:01 taipei Taipei-openser[4083]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4085]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4087]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4089]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4091]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4050]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4067]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4069]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily with: modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") this error does not appear. In neither way it work for me (ring but no audio) Can anybody help me please? bye Ronald Wiplinger

2 1

[Users] NAT: STUN, nathelper, rtpproxy, mediaproxy
by Ronald Wiplinger 04 Oct '06

04 Oct '06

We have STUN, nathelper with rtpproxy or mediaproxy, ... maybe even more. Which one is good for which situation? What are the differences? How can I cover each situation? Can I combine different methods? bye Ronald

2 1

[Users] SER with multiple asterisks
by Adi Simon 04 Oct '06

04 Oct '06

Hi, I have one SER and two asterisk boxes. I'm trying to duplicate each register message from the client so it will reach each of the asterisks. I tried the following in my openser.cfg: if ( (is_method("SUBSCRIBE")) || (is_method("REGISTER")) ) { seturi("sip:nobody@10.0.0.1:5060"); append_branch("sip:parallel@10.0.0.2:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } The first asterisk server registers the UA successfully. The second one gives this error: Oct 3 12:55:48 NOTICE[32364]: chan_sip.c:6395 check_auth: stale nonce received from 'sip:1234566@foo.bar' Transmitting (no NAT) to asterisk2:5060: I googled a bit and I understand I cannot fully duplicate a sip message since the nonce cannot be identical. Is there a way to recreate the sip message with another nonce? Is there any other solution to this scenario? Thanks, Adi.

1 1

[Serusers] SRV usage with SER
by Paul Antinori (pantinor) 04 Oct '06

04 Oct '06

Does SER support the SRV mechanism, sending a request to a URL with a SRV domain name? Also, if the first priority destination fails due to a timeout, will it failover to the second priority? Thanks for help, Paul

2 1

[Users] different between "silo" and "voice_silo" table
by cv aik 04 Oct '06

04 Oct '06

Hi, Thanx for the answer last time... I have add configuration of MSILO module in openser.cfg. The problem is I don't know how to test it (how to leave offline voice message between clients). If there is any application I have to use to save voice message, please let me know. And about the silo table, I've checked OPENSER database and I 've found that silo table is already there without I've to create it. But, when I configure my serweb, I found in my /var/www/html/serweb/config.php, there is another table named "voice_silo". There is no table named "voice_silo" in my OPENSER database. I'd like to know what's the different between "silo" and "voice_silo" (maybe in storing message (voice or text)). Thanx Regards, Aldi --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

2 1

[Serusers] authenticating forwarded calls
by Mark Price 04 Oct '06

04 Oct '06

Is it possible to set up SER with asterisk so that any INVITE that is forwarded from SER to asterisk does not need to be authenticated by asterisk? Thanks, Mark

2 1

[Users] Who want to take this challenge (20US$)
by Ronald Wiplinger 04 Oct '06

04 Oct '06

Hi, I want to get a success feeling, ... and with your help I would get my openser installation running with: a. mysql (done) b. nat traversal (not working yet) c. ENUM (not working yet) d. sipbroker (not started yet) e.g., *777xxxxx should go to Sipgate.de .... If you want to help me to set it up, please send me first an email and I am happy to reserve time for your set-up try. If you succeed I pay you US$ 20 to your PayPal account !!!! Environment: 1. Eyebeam is on my internal network 2. several video phones are available outside of my network (one on a public IP: Grandstream 3000 video phone) and others behind a nat 3. several ENUM numbers are available, which are going to my asterisk server (different machine, different domain name) 4. sipbroker can also reach our asterisk server 71xx IP range 192.168.10.x -------------- NAT 7111 (public IP) | | ---------------------------------------INTERNET -------------------------------------------- | | | extern: 203.70.36.126 | | second ADSL on 192.168.250.250 | public ip address | taipei.elmit.net (Openser server) other servers with 2 Ethernet | | IP range 192.168.250.x | | | | | --------------------------------------------------------------------------------------------- | | | | phones and desk top computers bye Ronald Wiplinger

1 0

[Users] ENUM dialing
by Ronald Wiplinger 03 Oct '06

03 Oct '06

I try to call an ENUM number. I use this settings for it: mpath="/usr/local/lib64/openser/modules/" loadmodule "enum.so" # -- enum params -- modparam("enum", "domain_suffix", "e164.arpa.") 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 >= 4096 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- log("I am checking for NAT now\n"); route(2); # ----------------------------------------------------------------- # if you want to make ENUM work with numbers starting with "00", # use the following to convert "00" it into a "+" # ----------------------------------------------------------------- log("I am checking for ENUM now\n"); if (uri=~"^sip:00[1-9][0-9]*@") { # tried before: taipei\.elmit\.net") { # strip leading "00" # (change example.net to your domainname or skip the stuff after the "@") strip(2); # (adjust, if your international prefix is something else than "00") prefix("+"); }; # check if request uri starts with an international phone # number (+X.), if yes, try to ENUM resolve in e164.arpa. # if no result, try in nrenum.net if (uri=~"sip:\+[0-9]+@") { # tried before: taipei\.elmit\.net") { # (change example.net to your domainname or skip the stuff after the "@") ####if ( !enum_query("e164.arpa.") ) { #### enum_query("nrenum.net."); #### }; if (enum_query()) { t_relay(); exit; } }; I tried to add some log statements, but I never see them in /var/log/messages either. Below is the ngrep output. It seems that it does not go anywhere in route() Any ideas how to track it down? bye Ronald Content-Type: application/sdp. User-Agent: eyeBeam release 1005g stamp 32576. Content-Length: 809. . v=0. o=- 2 2 IN IP4 192.168.250.108. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.250.108. t=0 0. m=audio 48702 RTP/AVP 107 100 106 6 0 105 18 3 5 101. a=alt:1 2 : O5/HQNzE iLXVkGqH 5.32.86.166 48702. a=alt:2 1 : ok7vH1EF jO6dCAlm 192.168.250.108 48702. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:E4797B0FC1A140648811BD63523C3B70. m=video 40488 RTP/AVP 125. a=alt:1 2 : urVBRqXw O9lI2kQp 5.32.86.166 40488. a=alt:2 1 : LCXVKz4X nxDzxq4V 192.168.250.108 40488. a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800. a=rtpmap:125 H264/90000. a=sendrecv. a=x-rtp-session-id:B8E2EE0EA3944D6DA47E5D98AABCD52B. # U 2006/09/28 16:57:11.567171 192.168.250.41:5060 -> 192.168.250.108:40430 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport=40430. To: "00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072. Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=11197 req_src_ip=192.168.250.108 req_src_port=40430 in_uri=sip:00886228357765@192.168.250.41 out_uri=sip:+886228357765@192.168.250.41 via_cnt==1". . # U 2006/09/28 16:57:11.569258 192.168.250.108:40430 -> 192.168.250.41:5060 ACK sip:00886228357765@192.168.250.41 SIP/2.0. Via: SIP/2.0/UDP 192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport. To: "00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072. Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk.. CSeq: 1 ACK. Content-Length: 0. .

2 3

[Users] Digest authentication
by Artur Hayne 03 Oct '06

03 Oct '06

Hi, Can I change the authentication protocol for any other or the Openser authenticate only with the digest? The Digest needs that the password in the DB stay in clear text and its not good. Thanks. --------------------------------- O Yahoo! está de cara nova. Venha conferir!

4 4

[Users] Who is the rtpproxy current maintainer ?
by Jeremie Le Hen 03 Oct '06

03 Oct '06

Hi there, sorry for bugging you with such a trifling question, but I tried to send a patch to the documented RTPProxy maintainer (Maxim Sobolev) twice (@PBXPress.com and @portaone.com) but I got a mail from MAILER-DAEMON each time, claiming the user doesn't exist. Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

3 5

[Serusers] Re: sems pthread issue
by jorain 03 Oct '06

03 Oct '06

Hi everyone, I'm doing some heavy tests on sems, which involves making lots of calls to ivr & isdngw modules. I noticed that the sems threads size (seen from process table or system monitor) keep increases with every call made. It wasn't release even when the call is already hangup, until it reach like 2GB, then I start getting pthread error code 12 whenever I tried calling. I have to restart sems to release the threads. I'm no good in C language and not sure how I can handle the pthread error. Can anyone give me some advice as how to utilize the memory usage for sems, so that it can keep on running like SER? Best Regards, Jorain

3 2

[Users] problem to create DB in openser cvs (1.2)
by Rafael J. Risco G.V. 03 Oct '06

03 Oct '06

Hi I've installed openser dev 1.2 to test new acc module in my lab but cant create DB in mysql server using ist openser_mysql, please see this error: ]# openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': 10.0.1.231 creating database openser ... /usr/local/sbin/openser_mysql.sh: line 262: method: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: cseq_no: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_code: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_reason: command not found real 0m0.000s user 0m0.000s sys 0m0.000s /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: method: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: cseq_no: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_code: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_reason: command not found real 0m0.000s user 0m0.000s sys 0m0.000s /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found ERROR 1064 (42000) at line 80: You have an error in your SQL syntax; check the manual that corresponds to your MySQL serverversion for the right syntax to use near '=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wh'at line 2

2 2

[Users] rewritehost failure route
by Cseke Tamas 03 Oct '06

03 Oct '06

Hello, I 've problem with rewritehost function in failure_route with openser 1.1.0 In core functions i don't know in which route can i use them, unlike functions in modules. I 'd like to make a failover system, In route block rewritehost is successfull but on failure route there is no effect i think, i pass the host to t_relay, so it is relayed to correct host, but in r-uri, there is incorrect host, and sip proxy reply with (SIP/2.0 400 Bad Request - 'Invalid IP Address) route{ ...... t_on_failure("1"); rewritehost("xxx.xxx.xxx.90"); t_relay(); ..... } failure_route[1] { t_on_failure("2"); append_branch(); rewritehost("xxx.xxx.xxx.91"); t_relay("xxx.xxx.xxx.91"); } failure_route[2] { # failover cisco (zajcek) append_branch(); rewritehost("xxx.xxx.xxx.31"); t_relay("xxx.xxx.xxx.31"); } Thanks any help, Tamas

2 1

[Users] SER / Load Balancing
by Thomas Deillon 03 Oct '06

03 Oct '06

Hi everybody, I saw this: http://www.iptel.org/faq/how_can_i_get_load_balancing_failover_with_ser. On asterisk, you can put and share on DB the dial plan and the sip definition but not the information relative to the REGISTER message (localisation). So, you cannot make perfect load-balanced servers. We have here a lot of clients and asterisk boxes in 3 towns with a very good connections between them. I wondered if it's possible with ser to put and share in the database the "REGISTER" information and so able any phone to connect to any server ? Thanks a lot for your help, Best Regards, Thomas

1 0

[Serusers] How to test basic multicast support on LAN?
by zain 03 Oct '06

03 Oct '06

Hi, I want to use basic multicast support in SER, but I don't know how to test the multicast function on LAN to see whether it's working in SER. The following are some descriptions about my version. OS: Debian kernel version: 2.6.8-2-686 SER version: 0.9.6 I add the following lines in ser.cfg. mcast_loopback=yes mcast_ttl=32 I use "/etc/ser start" to start SER. SER works good, but I don't know how to verify multicast on LAN. If the UA(User Agent) is kphone or SJphone. Thanks in advance. Justin

2 1

[Serusers] http digest authentication
by Holger Kinkelin 03 Oct '06

03 Oct '06

Hello list! I wonder how a UAC calculates the response to a 401 "Unauthorized" message from a server. I know that it works much like http digest authentication, but it's not exactely the same, since I do NOT have the Quality of Protection (qop) value. (Or am I wrong? My assumption is based on the fact that I don't see a qop-value in the REGISTER message my kphone sends after the 401) I've written some piece of code that is (in pseudo code) like that: HA1 = md5(username + ":" + realm + ":" + password) HA2 = md5(digestURI) response = md5(HA1 + ":" + nonce + ":" + recCounter + ":" + cNonce + ":" + qualOfProt + ":" + HA2) username = "1000" realm = "192.168.0.31" password = "1000" digestURI = "sip:192.168.0.31" <--- IS THIS CORRECT?? Or is it s.th like "REGISTER sip: ..."? nonce = "4520e111333a24d8c4f3d20c6171cc37dfa2be33" recCounter = "00000001" cNonce = "abcdefghi" qualOfProt IS MISSING! I know that my code is working for "normal" http digest authentication but not for "sip digest authentication". So what am I doing wrong? Thanks for replies in advance! Regards, Holger

2 1

[Serusers] error running radius client
by technos e 03 Oct '06

03 Oct '06

Hi All, I am trying to run a radius client using radiusclient -f digest localhost auth <secretkry> its giving error: error opening configuration file, can anyone pls help me ?? its very urgent.. thanks in advance. techno __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] Need help in adding certificate
by Ferianto siregar 03 Oct '06

03 Oct '06

Dear all, Thank you very much for your time. All, I read from tls.html (www.openser.org) that for making good certificate, we must add rootCA to calist file by using: cat add_cacert.pem >> calist.pem 1.When I try to run it, I got error message. It can not execute? Why? It says :"There is no command add_cacert.pem" Does that command just make new file named "calist.pem"? (There is no difference between cacert.pem) 2. I compile TLS with openssl.0.9.7a. The openssl-0.9.7.a packet, I got when installing Redhat. But, How can I know that TLS that I have build using openssl or ssl (for security) version? or How can we run openssl or ssl? I do hope anybody can help me. Please..Thank you very much. Regards, Ferianto --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

2 1

[Users] Problem with timer and t_forward_nonack
by Antoine Fressancourt 03 Oct '06

03 Oct '06

Hello, I have a problem with my openser.cfg file, and especially with the way I should deal with the fr_inv_timer. I can't get it to work in my openser.cfgfile, and every time the timer ends I get the following error: 6(21521) ERROR:tm:t_forward_nonack: no branch for forwarding 6(21521) ERROR:tm:w_t_relay: t_forward_nonack failed The error occurs in a route bloc : route[1] { xlog("L_INFO", "INFO: route1 with r-URI $ru\n"); xlog("L_INFO", "test 14 : if(isflagset(2)) (line 200)\n"); if(isflagset(2)){ xlog("L_INFO", "-----> TRUE ,if the flag is set to 2, it calls t_on_failure\n"); t_on_failure("1"); t_relay(); exit; } xlog("L_INFO", "EXECUTION test 15 : if (!t_relay()) (ligne 206)\n"); if (!t_relay()) { xlog("L_INFO", "---------> TRUE\n"); sl_reply_error(); }; exit; } I have a log message at the beginning of failure_route[1] that shows that it is not entered before the error occurs. Could you tell me what's wrong ? Thank you very much, Antoine

1 0

[Serusers] xlog
by Legos Engineering 03 Oct '06

03 Oct '06

HI, The log of xlog are generated in messages. So i would like to know, what i must do if i want that xlog send its information in an other file for only one request L_WARN(...) I want to use this request in order to do stats of connection of UA Thx for your help.

7 7

[Users] doubt in Loose route section
by raviprakash sunkara 03 Oct '06

03 Oct '06

Hello Users, When I working account part and Nat with OpenSER, The Transaction of route record entered into the Loose route Section. if(loose_route()) { acc_db_request(" ",""); t_relay(); exit; }; # call type transaction taking place here......... .............. ..................... ................... ...................... ..................... .................. All the Accounting and voice part are working ... but i'm doubt in loose route taking place all the call transcation.... -- 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] Ser_acct_authentication using ser
by rahman atikur 03 Oct '06

03 Oct '06

Dear all: I am new in ser. I install ser 0.9.6. And make a call using halaphone.. Now i want to use radius server for authenticate. I already install Radius server freeradius-0.9.0 and radiusclient-ng-0.5.3 .. Now the problem is auth_radius module is not installed... >From the scratch ... how i solve the problem. Regards John __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] How to leave offline-voice message?
by cv aik 02 Oct '06

02 Oct '06

Hi all, I'd like to ask some questions about MSILO module.. I have setup server and it works well, then I add msilo configuration into openser.cfg For addition, my clients are using MS Windows. I used 2 clients to test MSILO feature. I try to call the offline client, but my x-lite display that user not found. Please tell me what's the problem. Thanx Regards, Aldi --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

2 1

[Serusers] SER connecting to SIP VoIP providers
by Voipers Portugal 02 Oct '06

02 Oct '06

Dear all, I am am using SER to provide SIP service to my clients, but I would like to allow them to make outbound calls. This is where a SIP VoIP provider comes in. As far as I know, I have 2 possibilities: - using IP authentication - using client authentication As for the first one, I only have to forward the INVITE messages, right? What I would like to know, is how to use "user authentication". In this case, it would be transparent to the VoIP provider since it should be treated as a regular SIP user agent. Can anyone help me out? Jose Simoes

4 5

[Users] how to handle "488 Not Acceptable Here"
by Richard Bennett 02 Oct '06

02 Oct '06

hi, When calling from Sipura to softphone (either SJphone or the sip client in my e61 nokia phone), with the Sipura set to Prefered codec: g723 Use prefered codec only: no and the softphone only supporting g711, I get this reply from the softphone: U xxxx:11474 -> xxxxx:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP xxxx;branch=z9hG4bK23e.248c6cd1.1,SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. To: <sip:1@test.com>;tag=ujk6mpqhbhhc6kj68irr. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Warning: 304 192.168.1.3 Media type not available. Content-Length: 0. the invite was: U xxxx:5060 -> xxxx:11474 INVITE sip:1@test.co SIP/2.0. Record-Route: <sip:xxxx;lr=on;ftag=d0404120874d710eo0>. Via: SIP/2.0/UDP xxxx:branch=z9hG4bK23e.248c6cd1.1. Via: SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. To: <sip:1@test.com>. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Max-Forwards: 69. Contact: sipura line1 <sip:2@xxx:10001>. Expires: 240. User-Agent: Sipura/SPA2002-3.1.5. Content-Length: 419. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 36143 36143 IN IP4 xxxx. s=-. c=IN IP4 xxxxxx. t=0 0. m=audio 62238 RTP/AVP 4 0 2 8 18 96 97 98 100 101. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. If I set prefered codec to g711 on the Sipura it works normally. What is the best way to handle this on the sip proxy? Thanks, Richard

2 2

[Serusers] Accounting problem with radius server
by Caxton 02 Oct '06

02 Oct '06

Dear all: I setup SER and use radius server to accounting. However, the record after I test are not complete. A(Caller) <------> B(Callee) Suppose A is caller and B is callee. The record will be only one and correct if A hang up first. Unfortunately, if B hang up faster than A, the record will be divide into two records like table1(for example.). Table 1. |-----------------------------------------------------------------------------------------------------------------| |AccSessionID AccUniqueId ...<skip>... Start Time Stop Time ...<skip>... | |-----------------------------------------------------------------------------------------------------------------| |AHandUpFirst 001 2006-10-02 2006-10-02 | | 12:40:03 12:45:32 | |-----------------------------------------------------------------------------------------------------------------| |BHandUpFirst 002 2006-10-02 0000-00-00 | | 12:58:40 00:00:00 | |-----------------------------------------------------------------------------------------------------------------| |BHandUpFirst 003 2006-10-02 2006-10-02 | | 13:05:03 13:05:03 | |-----------------------------------------------------------------------------------------------------------------| How can I solve this problem? Please give me a hand. Test User Agent: Window messenger 5.1 and X-lite Regards, Caxton

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[Users] OT: searching for a softclient with SDP feature
by Thomas Balsfulland 02 Oct '06

02 Oct '06

hello group, i am searching for a sip-softclient (windows or linux - opensource, freeware preferred), which is able to send an INVITE completely without any SDP information (no message body). does someone of you know something like this? regards thomas -- thomas balsfulland tbals(a)ctrl-c.de zwischen mut und dummheit liegt nur ein grat der sich durch das vorbereitetsein unterscheidet

2 1

[Serusers] SER for inbound calls
by Voipers Portugal 02 Oct '06

02 Oct '06

Dear all, Another question that I have to do is: Imagine that I have a connection to a VoIP provider using a username and a password (so that SER register in the VoIP provider as a normal user). Once it is registered, do you think it possible to receive calls? That is, make it ring in one or all of the SIP users configured in SER? Any help would be appreciated. Jose Simoes

3 2

[Serusers] DNS SRV support ?
by Kostas Marneris 02 Oct '06

02 Oct '06

Hello, I run ser.0.9.6 and I can use either rewritehostport("1.2.3.4:5060"); or rewritehostport("pstngw.foo.com:5060"); with t_relay() command on INVITE/PSTN handler in order to route calls to a Cisco PSTN GW. This works fine. Is "DNS SRV queries" supported on ser.0.9.6 and is there a specific conf ? thanx Kostas

6 7

[Serusers] problem
by vijay tiwari 02 Oct '06

02 Oct '06

i have install the ser and it installed but i have some problem 1. first i am not able to run the ser command 2. i want to make the scripts for ser and mediaproxy but its show me one error for running this command i am in /etc/init.d when i am running this command chkconfig --add ser error reading information on service ser: No such file or directory please help me out with regards vijay --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

[Users] complete openser.cfg available?
by Ronald Wiplinger 02 Oct '06

02 Oct '06

I would like to see beside the sample and the "advices" to add to the cfg file a complete cfg file. Is such a more complese sample available? I am sure most of the users will use more than just the basics, and most of them will use the same modules and set-up. As new user, I face often the problem where to put the snip set from the documentation. I think it would be better to comment out all not necessary parts for the basic, or mention the spot to put in other configuration parts (anyway only the route() part needs that) bye Ronald

3 2

[Serusers] UAC multiple Registration
by Abdul Qadir 02 Oct '06

02 Oct '06

Hi, Is it possible in SER to implement login concept like yahoo messenger, when you login from another pc your prevois login is signedoff by server. i.e unregister a previously registerd client when same client tries to register from another location. Thanks in advance, Abdul Qadir --------------------------------- Get your email and more, right on the new Yahoo.com

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[Users] Duplicating SIP Messages
by Adi Simon 01 Oct '06

01 Oct '06

Hi, I'm trying to figure out if it is possible to duplicate SIP Messages. In a setup of a SER proxy and two asterisk boxes I want to send all the Register messages to both of the asterisks, I tried doing the following in the openser.cfg: if (method=="REGISTER") { route(2); } else { route(1); }; route[2] { rewritehostport("10.0.0.1:5060"); if (!t_relay()) { sl_reply_error(); }; rewritehostport("10.0.0.2:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } But I keep getting : 0(22183) ERROR: t_newtran: transaction already in process 0xb6136318 0(22183) DEBUG:sl:sl_reply_error: error text is I'm terribly sorry, server error occurred (1/SL) Is it possible at all? Thanks, Adi.

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[Users] Please help for tasting TLS with minisip
by Ferianto siregar 01 Oct '06

01 Oct '06

Dear Jeevan, How are you? I do hope your are fine. I would like to say thanks to you for your help before. Thank you. I am so sorry for disturb you. Sorry Jeevan, I still have problem with configuring minisip like I asked you a week ago. I still can not make a call with minisip. There is always error message. Can I ask you again, Please? Here is my problem: 1. I run ./gen_rooCA.sh and running this script, I got rootCA directory. This directory contains: a.cacert.pem b.index c.serial d.private directory which contains cakey.pem e. cert directory which contain 01.pem 2. I run ./gen_usercert.sh user and running this script,I got user directory. This directory contains: a.user-calist.pem b.user-cert.pem c.user-privkey.pem d.user-cert_req.pem 3. When running minisip (suppported TLS), I done these tasks: a.In "Preferences settings", I use TLS port in "enable TLS Local Port" and "Transport Method" is TLS. My Questions: 1. In "Personal Settings", what file should I put? I mean, in menu "choose a certificate", what should I browse to fill this? 2. In " choosen a private key", what should I browse to fill this? 3. In "CA database", what sholud I browse to fill this? Because when I fill "chosee a certificate" with "cacert.pem", the "choosen a private key" with "cakey.pem" ,and the " CA databese" with "user-cert.pem","cacert.pem","user-calist.pem","user-privkey.pem", I got the error message: The error message said that the "SSL : connect failed" Here is : SipMessageTransport: sendMessage: creating new socket IP4Address(string): sip.pcr.ac.id (202.95.149.251) Creating new SSL_CTX Enter PEM pass phrase:openser Unknown command: o Packets will be displayed to the screen Unknown command: e Unknown command: n Unknown command: s Unknown command: e SSL: connect failed SipMessageTransport: sendMessage: exception thrown! Registering all identities to their registrar: Registration request sent (username = aldi(a)pcr.ac.id) SipMessageTransport: sendMessage: creating new socket IP4Address(string): sip.pcr.ac.id (202.95.149.251) What should I do, Jeevan? I have tried it many times, but I still can not make a call with minisip. I realize that my skill is too weak. Would you mind guide me again to solve this problem? Please... Thank you very much.. I am sorry for waste your time, Thanks. Regards :) Ferianto Note: 1. I can not run openssl x509 -in cakey.pem -out cakey.crt. It says that it can not converted. TRUSTED CERTIFICATE 2. I copy all the file ("cacert.crt",cakey.pem","user-calist.pem",etc) from my linux server to windows. Because windows can not download with the extension .pem. Is it correct? --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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[Users] [openser] Enabling TLS Support
by John Shellhouse 30 Sep '06

30 Sep '06

Hi, I am about to enable TLS support although my current openser.cfg system works fine and I do not particularly want to damage it, but I would like to incorporate TLS. I was planning to save my current openser.cfg file and just add the necessary TLS info and wonder if this can be done? 1) I understand that I can use both TLS(5061) and TCP(5060), so therefore particular portions in my current openser.cfg should still work, is this a proper assumption? 2) How would this affect users that are using port TCP(5061) ? 3) So when I remake with TLS support how will this affect the current system? 4) Will the remake write over all files stored in /usr/local/etc/openser and other openser directories? 5) Will the remake create new database tables, basiecally would I be able to keep the current database structure I currently have without reloading openser_mysql.sh? 6) In a nutshell, how will the remake negatively affect my current openser system that I should be aware of? I apologize for the lengthy amount of questions but I am just not sure in these areas. Thanks in advance. John

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Re: Re: [Users] tls. soket queues not flushed
by serega 30 Sep '06

30 Sep '06

Hi Klaus and all other who read this. It's not a problem of tcp socket - yesterday i rewrite gateway config for fowarding directed to extgateway requests to gateway itself. In tcp this problems not occured, but after i switched on tls, i see overfilled queues again. In that case i haven't extgateway, communications with remote sockets and etc. Did anybody test openser whith big payload? I think this problem occurs because in my case i have big traffic and several process that simultaniously use tls. Did anybody have the same problem? >Hi Serega! > >I'm not sure - just a guess. openser uses "worker threads". This worker >threads get a SIP message from the listener thread, process them, and >then they send them. > >If for some reason the sending fails - e.g. the gateway sends via >TCP/TLS to extgateway and extgateway's receiving queue is full. Then >gateway's sending buffer gets full and the threads wait until they can >send. Thus, as the threads are all busy, also the receiving queue of >gateway gets full. I guess after some time the threads give up sending >and will read again. > >The question is: why does the sending buffer gets full? Where is gateway >sending the messages to? > >The sending queue must not get flushed. The queue is in the OS, not in >openser. As TCP guarantees that there is no loss/reordering the TCP >stack must not flush the queue. > >regards >klaus > >serega wrote: >> Hi all. >> >> I have next problem while using openser with tls. System consists of >> sip statefull server (in next time, simply, server) and stateless sip >> gateway (gateway). Server used for connect to jabber server. Server >> contains rewrote jabber module logic and also use presence module. >> Gateway used to connect to other sip gateway (extgateway) via tls >> protocol. Gateway use 1.1 openser version and doesn’t contain changed >> code. When server together with gateway restarts and server in its >> database contains above 260 subscriptions (in watcherinfo table) i >> have error. Socket used by gateway to connect with extgateway >> contains in receive and sent queues a lot of data (above 50kByte on >> each sides). This data never flushed out. This happens because after >> restart sip server through gateway send notifications to extgateway. >> When I attached using gdb to process that send data, I saw that it >> was in infinity loop because tls library returns SSL_ERROR_WANT_WRITE >> and I think it’s correct because we have overfilled send queue. In >> this case I don’t interest why receive queue not empty (I think it >> happens because sending process have got block on socket). But I >> don’t understand why sent queue not flushed. I test this behavior >> using tcp – all was correct. Socket not closed by other side because >> this status can be unchanged above one day. Anybody can help me? Why >> this happens? >> >> I use openssl-0.9.8c and redhat os (Linux xdevel1 2.4.21-4.ELsmp #1 >> SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux). Computer >> have too Pentium 4 processors). >> >> >> Gateway.cfg (sip.qa.hbex.com – gateway address): >> >> # # $Id: router_qa.cfg,v 1.1 2006/08/02 18:14:34 ilya Exp $ # # >> simple quick-start config script # >> >> # ----------- global configuration parameters >> ------------------------ >> >> debug=9 # debug level (cmd line: -dddddddddd) >> log_facility=LOG_LOCAL0 fork=yes log_stderror=yes # (cmd line: -E) >> >> >> check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) >> rev_dns=no # (cmd. line: -R) children=20 >> fifo="/tmp/openser_fifo" server_signature=no >> >> fifo_db_url="mysql://root:@localhost/openser" >> >> listen = udp:sip.qa.hbex.com:5060 #incoming sip server address listen >> = tcp:sip.qa.hbex.com:5060 >> >> #tls address disable_tls = 0 listen = tls:sip.qa.hbex.com:5061 listen >> = udp:sip.qa.hbex.com:5061 >> >> tls_certificate = "/ home/inop/ex.com-cert.pem" tls_private_key = >> "/home/inop/ex.com-privkey.pem" tls_ca_list = >> "/home/inop/ex.com-calist.pem" tls_require_client_certificate=0 >> tls_verify_client=0 tls_verify_server=0 >> >> # ------------------ module loading >> ---------------------------------- loadmodule >> "/home/interop/openser/lib/openser/modules/rr.so" loadmodule >> "/home/interop/openser/lib/openser/modules/xlog.so" loadmodule >> "/home/interop/openser/lib/openser/modules/textops.so" loadmodule >> "/home/interop/openser/lib/openser/modules/maxfwd.so" loadmodule >> "/home/interop/openser/lib/openser/modules/sl.so" loadmodule >> "/home/interop/openser/lib/openser/modules/mysql.so" loadmodule >> "/home/interop/openser/lib/openser/modules/tm_unchanged.so" >> loadmodule "/home/interop/openser/lib/openser/modules/usrloc.so" >> loadmodule "/home/interop/openser/lib/openser/modules/registrar.so" >> >> # ----------------- setting module-specific parameters >> --------------- modparam("registrar", "default_expires", 120) >> modparam("registrar", "use_domain", 1) >> >> modparam("usrloc", "use_domain", 1) 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) >> >> # add value to ;lr param to make some broken UAs happy modparam("rr", >> "enable_full_lr", 1) modparam("maxfwd", "max_limit", 10) # >> ------------------------- request routing logic ------------------- >> # main routing logic route{ if (!mf_process_maxfwd_header("10")) { >> sl_send_reply("483","To Many Hops"); drop(); }; if (dst_port==5061) { >> if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) >> { # rewrite destination and forward to jabber (sip server) route(1); >> return; }; sl_reply_error(); return; } if (method=="REGISTER") { >> xlog("XXX: saving location msg=$mb\n"); if(!save_noreply("location")) >> { log("XXX: Error saving location!\n"); sl_reply_error(); } >> sl_send_reply("200","OK"); return; } if >> (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { >> #forward to self route(3); return; } ##forward to ext gateway >> route(2); } >> >> route[1]{ if (method!="MESSAGE" && method!="SUBSCRIBE" && >> method!="NOTIFY") { log("XXX: Request not forwarded to sip >> server!\n"); sl_send_reply("202","Accepted"); return; } >> t_on_reply("1"); record_route(); >> >> if(!lookup("location")) { if (method!="SUBSCRIBE") { log("XXX: only >> subscribes are processed for user that are not registered\n"); >> sl_reply_error(); return; } >> subst_uri('/(.*)@(.*)/\1#\2*jabber.qa.im.hb.com@jabber01.qa.im.hb.com/ig'); >> if(!t_relay()) { log("XXX: error forwarding jabber01...\n"); >> sl_reply_error(); return; } else { sl_send_reply("200","OK"); return; >> } } else { log("XXX: forwarding to the address of record...\n"); >> if(!t_relay()) { log("XXX: error forwarding to address of record >> \n"); sl_reply_error(); return; } else { sl_send_reply("200", "OK"); >> return; } } >> >> } >> >> # forwarding to external gateway route[2]{ log("XXX: rewriting >> headers\n"); >> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> >> if(!forward("tls:43.123.141.166:3000")) { log("XXX: Error forwarding >> to external gateway!\n"); sl_reply_error(); } else { >> sl_send_reply("200", "Accepted"); }; } >> >> # forwarding to SELF route[3]{ log("XXX: rewriting headers\n"); >> subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> >> log("XXX: about to forward to self\n"); >> if(!forward("tcp:sip.qa.hbex.com:5061")) { log("XXX: Error forwarding >> to self!\n"); sl_reply_error(); } else { sl_send_reply("200", >> "Accepted"); }; } >> >> # this is executed for replies onreply_route[1]{ >> subst('/^(To:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> >> subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); >> route(2); } >> >> >> >> _______________________________________________ Users mailing list >> Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Users] Is there a way to give a dial tone to the caller?
by Ronald Wiplinger 30 Sep '06

30 Sep '06

When I am placing a call I will never hear a dial tone. Is there a way to add dial tone? bye Ronald

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[Users] Please help for creating CA root certificate nad client certificate
by Ferianto siregar 30 Sep '06

30 Sep '06

Dear all, Thank you very much for this chance. Thank you for your time to read this message. All, I have read document about tls from openser.org In that document, I read that to create CA root and client certifacate, we just must run script ./gen_rootCA.sh and ./gen_usercert.sh where is in tls/tools. It will produce rootCA directory and user directory. Then, I put the rootCA directory and user directory into my public webserver in order the client can download it. But, when I try to download the certificate from the server (using Redhat 9.0), to the client computer (using Windows XP), why I can not download it? Do, I have to convert the extension form .pem to .crt? How can I convert it? Please help..Please.. Regards, Ferianto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Best platform for SER --->please suggest some thing
by ravi reddy 30 Sep '06

30 Sep '06

Hello guys, I want to install SER on a SERVER so please suggest me some servers (prefererably with configuration like HP, or some thing) because basically i dont have any knowledge on server types . so, any comments and suggestions are appreciated . Thank You. Regards, Ravi.

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[Serusers] SER and Routing Help Question
by Mario Roeber 30 Sep '06

30 Sep '06

Hello, maybe someone can help me with the routing. here the schema. Firewall with 3 Network interfaces 2x interfaces intern 1x interface extern the default provider is sipgate.de. ser shoult route all registration default to sipgate and listen on 2 intern interfaces. ser shoult handel all intern calls and route direct intern. user auth i'dont need or radius i'dont need. all inbount calls from extern shoult route automatic to right sip- phone. current i use the default configuration with 2 options more. listen 212.21.68.62 listen 212.21.69.110 maybe someone can help me to with the routing thanks a lot for your time, if you can help me. regards mario roeber

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[Users] [openser] Displaying Caller-ID
by Tracy Lofton 29 Sep '06

29 Sep '06

Hi Everybody, I was wondering if I would be able to display the CallerID in the display of the callee's phone rather than it displaying anonymous, maybe it can be done if I used something like this from the tm module: if(search("From:[0-9]{10}@.*"){ append_hf("remote-party-id:<sip:2134445555 at mydomain.com>") ) or would i use openser auth module with append_rpid_hf() in some form or fashion? If so, how can I do this? Just need a little input, guidance and assistance and to know if it actually can be done. If not this method, then what can I use to display the callID of the calling party to the caller display. Some providers are known to block the call if a certain amount of digits are not displayed. Thanks in advance. Tracy _________________________________________________________________ Add fun gadgets and colorful themes to express yourself on Windows Live Spaces http://clk.atdmt.com/MSN/go/msnnkwsp0070000001msn/direct/01/?href=http://ww…

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Re: [Serusers] how to add mysql.so
by vijay tiwari 29 Sep '06

29 Sep '06

thanks when i install mysql and i type this command ./configure --prefix=/usr/local/mysql --enable-large-files --enable-shared=yes it show me some checking and show me this error checking "LinuxThreads"... "Not found" configure: error: This is a linux system and Linuxthreads was not found. On linux Linuxthreads should be used. Please install Linuxthreads (or a new glibc) and try again. See the Installation chapter in the Reference Manual for more information. and i am not able to install mysql sip <sip(a)arcdiv.com> wrote: Why not? That's a little vague. Can't add them because you don't have it? Can't add them because of permissions problems? Can't add them because the directory doesn't exist? Can't add them because you don't know the unix copy command? A little more specific information goes a long way. N. On Fri, 29 Sep 2006 07:30:18 -0700 (PDT), vijay tiwari wrote > > hi all > > > i am not able to add the mysql in that locations /usr/local/lib/ser/modules > > > thanks > vijay > --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster. Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

1 0

[Users] thought exercise
by Mark Kent 29 Sep '06

29 Sep '06

I wonder if anyone on this list is open to a thought exercise? I'm trying to put together a cogent message about this, but in the meantime: I've got a situation where my openser box seems to route stuff to itself and it loops around and around until I get the classic "Too Many Hops" error. This seems to happen most often on BYE, although I've seen an ACK do it too. Here is a small ngrep snippet, after PHONE has called OPENSER-ONE who statelessly relays to OPENSER-TWO who t_relays to PSTNGW, and then the pstn user hangs-up: U 23:03:53.254382 PSTNGW:5060 -> OPENSER-TWO:5060 BYE sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone SIP/2.0 Call-ID: callid@PHONE From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674 To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag Content-Length: 0 CSeq: 1 BYE Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch Route: <sip:JoeUser@OPENSER-ONE:5060> Max-Forwards: 70 # U 23:03:53.255013 OPENSER-TWO:5060 -> OPENSER-ONE:5060 BYE sip:JoeUser@OPENSER-ONE:5060 SIP/2.0 Record-Route: <sip:JoeUser@OPENSER-TWO;lr=on;ftag=26674> Call-ID: callid@PHONE From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674 To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag Content-Length: 0 CSeq: 1 BYE Via: SIP/2.0/UDP OPENSER-TWO;branch=otherbranch.0 Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch Max-Forwards: 69 Route: <sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone> Right here OPENSER-ONE freaks out, and the delay causes OPENSER-TWO to re-transmit: # U 23:03:53.611227 OPENSER-TWO:5060 -> OPENSER-ONE:5060 BYE sip:JoeUser@OPENSER-ONE:5060 SIP/2.0 Record-Route: <sip:JoeUser@OPENSER-TWO;lr=on;ftag=26674> Call-ID: callid@PHONE From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674 To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag Content-Length: 0 CSeq: 1 BYE Via: SIP/2.0/UDP OPENSER-TWO;branch=otherbranch.0 Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch Max-Forwards: 69 Route: <sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone> OK, so who on this list is good enough to be able to suggest plausible theories about what my openser.cfg flaw is? If you don't want to speculate openly on the list then please send me a direct email. Thanks, -mark

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[Users] e2e ACK matching problem
by JF 29 Sep '06

29 Sep '06

Hi all, I'm facing a cenario where an INVITE transaction goes through one (Open)SER more than one time, and the ACK to a 480 Temporarily Unavailable is wrongly being matched as an e2e ACK. I dug through tm code and found this comment in t_lookup.c: /* here we do an exercise which will be removed from future code * versions: we try to match end-2-end ACKs if they appear at our * server. This allows some applications bound to TM via callbacks * to correlate the e2e ACKs with transaction context, e.g., for * purpose of accounting. We think it is a bad place here, among * other things because it is not reliable. If a transaction loops * via SER the ACK can't be matched to proper INVITE transaction * (it is a separate transactino with its own branch ID) and it * matches all transaction instances in the loop dialog-wise. * Eventually, regardless to which transaction in the loop the * ACK belongs, only the first one will match. */ Does anyone have a solution to this problem? Thanks, JF

1 0

[Serusers] Registration
by senthil.nachimuthu＠bt.com 29 Sep '06

29 Sep '06

> Hi, > I want to configure SER as a proxy server. I would like SER to > proxy presence messages and I have done that with the help of > record_route and t_relay_to_udp. > I would like to have SER forwarding the register message to another > server to authenticate and after successful authentication SER store > the location to route the requests. I observed SER is not allowing URI > getting registered. > > I tried the following but did not work.. > > if(!uri==myself) { > > if( method=="REGISTER") { > record_route(); > save("location"); > t_relay_to_udp("132.146.109.243","5060"); > break; > }; > }; > > > Can anyone let me know how can I enable other domain URI passing > through SER to reach other server which authenticates them? > > > Thanks and Regards, > Senthil Kumar, >

1 0

[Serusers] how to add mysql.so
by vijay tiwari 29 Sep '06

29 Sep '06

hi all i am not able to add the mysql in that locations /usr/local/lib/ser/modules thanks vijay --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.

1 0

[Users] video rtp
by Ronald Wiplinger 29 Sep '06

29 Sep '06

Can anybody explain me this, please. How is the rtp stream with video? Which ports are usually used for that? Following thoughts I have how it could be: 1. video and audio are two indipendent different rtp streams 2. audio starts with the lower free udp numbers, video with the higher ones. Only that could give me an explanation, why audio is ok in our test, but no video, and only if we want to call to a NAT, but worked from it. Where can I get more info? bye Ronald

1 0

[Serusers] Can I configure SER to send Radius Accounting packets?
by a.kolesnikov 29 Sep '06

29 Sep '06

Hello, everyone. Can I configure SER to send Radius Accounting packets? С уважением, Колесников Андрей Netris телефон: +7 495 950 55 25 ext. 1303 <http://www.netris.ru> http://www.netris.ru

1 0

[Serusers] Busy Call Fwd..
by Natambu Obleton 29 Sep '06

29 Sep '06

I have my SER server setup to forward calls when busy and I am using polycomm 601 registered to SER. When the Polycomm is on the phone and another phone call its, they don't get a busy signal, but show up on line2 for answering. I think this is a stateful vs. stateless problem, but how can I set it up so that when the user is on the phone it rings busy? Really just limit the number of simultaneous calls. Thanks. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

2 1

[Serusers] Re: SER and SPA 942
by Alex Fler 29 Sep '06

29 Sep '06

Hello All, Sorry for stupid question, I have ser server configured, and it works fine for my needs, I never had register my UA's directly to SER. When I registered I SPA 942 phone directly to ser and put call on hold, my phone display the message "Trying to hold" for a few seconds, and after that call getting dropped. Does anybody know why? Thank you for help. Regards Alex Fler

2 1

[Users] cdrtool/mediaproxy/openser
by Richard Bennett 29 Sep '06

29 Sep '06

Hi, Does anyone have an example openser.cfg file for using openser with Mediaproxy and CDRTool? I have Mediaproxy/openser working with the onsip getting started example, but this only passes calls that are behind nat to Mediaproxy. What is the best way to setup openser so that all calls pass through mediaproxy, and the corrected CDRs* can be written with accounting? * as mediaproxy disconnects zombie-calls, and corrects the accounting to the time the last rtp packets were detected. Thanks, Richard

1 0

[Users] RTPProxy manual
by Jeremie Le Hen 29 Sep '06

29 Sep '06

Hi, sorry if this is a bit off-topic, but I am looking for a document as near as a manual page as possible. I remember to have found such a document a few weeks ago, but I can't find it any more. It basically explained what did the command-line options, what is the bridge mode, etc. Pointers are welcome. Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

1 0

[Users] mediaproxy working, but not if asterisk is involved
by Arne Van Theemsche 29 Sep '06

29 Sep '06

hi my users subscribe with openser, en asterisk is used as connectivity to pstn i am now installing a mediaproxy, for all users, so every call goes via a mediaproxy. I'm doing this as follows (relevant statements only) in route #I installed the t_on_reply here to be sure that every reply gets parsed, but normally in the INVITE section should be enough? t_on_reply("1"); if (method==INVITE) { use_media_proxy(); } onreply_route[1] { log(-3,"reply received"); if (!search("^Content-Length:[ ]*0")) { log(-3,"using mediaproxy"); use_media_proxy(); }; } the weird is, for all local users, this works fine, but as soon as asterisk is involved, the reply doesn't get triggered (not seeing the "reply received" either, only when disconnecting the call). The call get's established fine, asterisk is sending media to the mediaproxy, but the SDP towards the calling phone is not modified (since the onreply isn't triggered) am I missing something here? thanks Arne

2 4

[Users] Cancel question/problem
by Andreas M. 29 Sep '06

29 Sep '06

Hello, i have following setup: Cisco AS5300 -- Openser --- Client Now if there is an incoming call from the CiscoGW to the client and is canceled, but the "cancel" message does not arrive at the client (because any reason), the phone doesn´t stop ringing. Cisco --- OSER --- Client Cancel -> <- Canceling Cancel -> thats it. It seems that no one is missing any OK or ACKS! It happens only sometimes, but who is responsible to handle this behavior ? The Cisco Gateway or the Proxy ? Using OSER 1.0.1 regards, Andreas M.

2 1

[Users] tls. soket queues not flushed
by serega 29 Sep '06

29 Sep '06

Hi all. I have next problem while using openser with tls. System consists of sip statefull server (in next time, simply, server) and stateless sip gateway (gateway). Server used for connect to jabber server. Server contains rewrote jabber module logic and also use presence module. Gateway used to connect to other sip gateway (extgateway) via tls protocol. Gateway use 1.1 openser version and doesn’t contain changed code. When server together with gateway restarts and server in its database contains above 260 subscriptions (in watcherinfo table) i have error. Socket used by gateway to connect with extgateway contains in receive and sent queues a lot of data (above 50kByte on each sides). This data never flushed out. This happens because after restart sip server through gateway send notifications to extgateway. When I attached using gdb to process that send data, I saw that it was in infinity loop because tls library returns SSL_ERROR_WANT_WRITE and I think it’s correct because we have overfilled send queue. In this case I don’t interest why receive queue not empty (I think it happens because sending process have got block on socket). But I don’t understand why sent queue not flushed. I test this behavior using tcp – all was correct. Socket not closed by other side because this status can be unchanged above one day. Anybody can help me? Why this happens? I use openssl-0.9.8c and redhat os (Linux xdevel1 2.4.21-4.ELsmp #1 SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux). Computer have too Pentium 4 processors). Gateway.cfg (sip.qa.hbex.com – gateway address): # # $Id: router_qa.cfg,v 1.1 2006/08/02 18:14:34 ilya Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) log_facility=LOG_LOCAL0 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=20 fifo="/tmp/openser_fifo" server_signature=no fifo_db_url="mysql://root:@localhost/openser" listen = udp:sip.qa.hbex.com:5060 #incoming sip server address listen = tcp:sip.qa.hbex.com:5060 #tls address disable_tls = 0 listen = tls:sip.qa.hbex.com:5061 listen = udp:sip.qa.hbex.com:5061 tls_certificate = "/ home/inop/ex.com-cert.pem" tls_private_key = "/home/inop/ex.com-privkey.pem" tls_ca_list = "/home/inop/ex.com-calist.pem" tls_require_client_certificate=0 tls_verify_client=0 tls_verify_server=0 # ------------------ module loading ---------------------------------- loadmodule "/home/interop/openser/lib/openser/modules/rr.so" loadmodule "/home/interop/openser/lib/openser/modules/xlog.so" loadmodule "/home/interop/openser/lib/openser/modules/textops.so" loadmodule "/home/interop/openser/lib/openser/modules/maxfwd.so" loadmodule "/home/interop/openser/lib/openser/modules/sl.so" loadmodule "/home/interop/openser/lib/openser/modules/mysql.so" loadmodule "/home/interop/openser/lib/openser/modules/tm_unchanged.so" loadmodule "/home/interop/openser/lib/openser/modules/usrloc.so" loadmodule "/home/interop/openser/lib/openser/modules/registrar.so" # ----------------- setting module-specific parameters --------------- modparam("registrar", "default_expires", 120) modparam("registrar", "use_domain", 1) modparam("usrloc", "use_domain", 1) 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) # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("maxfwd", "max_limit", 10) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","To Many Hops"); drop(); }; if (dst_port==5061) { if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { # rewrite destination and forward to jabber (sip server) route(1); return; }; sl_reply_error(); return; } if (method=="REGISTER") { xlog("XXX: saving location msg=$mb\n"); if(!save_noreply("location")) { log("XXX: Error saving location!\n"); sl_reply_error(); } sl_send_reply("200","OK"); return; } if (search("^To:.*@.*cmnicator.com") || search("^To:.*@.*cihb.com")) { #forward to self route(3); return; } ##forward to ext gateway route(2); } route[1]{ if (method!="MESSAGE" && method!="SUBSCRIBE" && method!="NOTIFY") { log("XXX: Request not forwarded to sip server!\n"); sl_send_reply("202","Accepted"); return; } t_on_reply("1"); record_route(); if(!lookup("location")) { if (method!="SUBSCRIBE") { log("XXX: only subscribes are processed for user that are not registered\n"); sl_reply_error(); return; } subst_uri('/(.*)@(.*)/\1#\2*jabber.qa.im.hb.com@jabber01.qa.im.hb.com/ig'); if(!t_relay()) { log("XXX: error forwarding jabber01...\n"); sl_reply_error(); return; } else { sl_send_reply("200","OK"); return; } } else { log("XXX: forwarding to the address of record...\n"); if(!t_relay()) { log("XXX: error forwarding to address of record \n"); sl_reply_error(); return; } else { sl_send_reply("200", "OK"); return; } } } # forwarding to external gateway route[2]{ log("XXX: rewriting headers\n"); subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); if(!forward("tls:43.123.141.166:3000")) { log("XXX: Error forwarding to external gateway!\n"); sl_reply_error(); } else { sl_send_reply("200", "Accepted"); }; } # forwarding to SELF route[3]{ log("XXX: rewriting headers\n"); subst('/^(From:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); log("XXX: about to forward to self\n"); if(!forward("tcp:sip.qa.hbex.com:5061")) { log("XXX: Error forwarding to self!\n"); sl_reply_error(); } else { sl_send_reply("200", "Accepted"); }; } # this is executed for replies onreply_route[1]{ subst('/^(To:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); subst('/^(Contact:[^@#]*)#([^@]*)\*jabber.qa.im.hb.com@jabber...qa.im.hb.com(.*)/\1@\2\3/ig'); route(2); }

2 1

[Users] TLS : Please help for TLS concept
by Ferianto siregar 29 Sep '06

29 Sep '06

Dear all, Thank you very much for all your time to read this message. All, I now have problem with TLS security concept. I have tried to read the tutorial (tls.html) from www.openser.org. I have read it many times. But, I still confused to undestand it. Would you mind help me, Please? Here are my questions: 1. How can we evidence that the communication that using TLS is secure communication? (Real proof) 2. When using TLS, how can we know that the communication has been eavesdropped? 3. Is TLS just use port 5061? If yes? How can TLS can protect the communication from sniffer who knows TLS using port 5061? 4. What kind of type encryption that TLS use? 5. What sofware that we can use to test for making a call in opeser with TLS support? I have tried minisip, but I always get error message..Please... I realize that my undestanding in reading TLS tutorial is weak. I do hope, anybody can help me, Please... Thank you very much, Regards, Ferianto --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

2 1

[Serusers] SER MySQL Support
by Holger Kinkelin 29 Sep '06

29 Sep '06

Hello everybody! I want to use the mySQL-support for SER but I've got a problem: When I start SER I get the following error DrScheme@dell:/usr/local/etc/ser> /usr/local/sbin/ser 0(7443) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory 0(7443) parse error (18,13-14): failed to load module ERROR: bad config file (1 errors) When I get this right, the libmysqlclient.so-file is missing. I looked for the file and indeed, it's not on the box. I don't know where to get that file from and where to put it, when I have it. Could please someone tell me? Incase you need to know what I did up to now: - I installed mySQL (4.1.13-3) using YaST on my Suse 10.0 box. The mySQL server is running happily - I installed SER 0.9.6 from a tar.gz package, run the script for mySQL-Tables and I use the following config-file # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen=192.168.0.31 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" # mysql-support fifo_db_url="mysql://ser:heslo@localhost/ser" # --- # mysql-support loadmodule "/usr/local/lib/ser/modules/mysql.so" # --- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # mysql-support loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # --- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # mysql-support modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # --- 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()) { route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ # mysql support if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; # --- if (!save("location")) { sl_reply_error(); }; } Regards, Holger

4 7

[Users] What to do when the ip address of the phone changes?
by Ronald Wiplinger 29 Sep '06

29 Sep '06

When a phone registers it has an expire time of 3600 sec. The user just turned off the phone and plugged it into another network. The server does not let register anymore, since it is already registered. Need to wait 1300 seconds for next register. Is there a way (openserctl xxxxxx) to release the phone ? I don't want to remove the phone from the database!!! bye Ronald

2 1

Re: [Users] thought exercise
by Mark Kent 29 Sep '06

29 Sep '06

>> Probably OPENSER-ONE does not recognize the BYE as targeted to itself Why would a BYE be targeted at a middle-man? I acknowledge that I don't check for this in my BYE handler :-) I did add this as a debug aid: if (is_uri_host_local()) { xlog("L_ERR", "relay to myself? $rm $ru $si $fu\n"); } in my stateful relay handler: route[1] { if (is_uri_host_local()) { xlog("L_ERR", "relay to myself? $rm $ru $si $fu\n"); } t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); } } but, obviously without understanding, I changed that to: route[1] { if (is_uri_host_local()) { xlog("L_ERR", "relay to myself? $rm $ru $si $fu\n"); sl_send_reply("200", "OK"); } else { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); } } } and that seems to snuff this out, but I also have the feeling that I'm just covering up for a flaw elsewhere. Thanks, -mark

2 3

[Users] Problem with module dispatcher
by Jose Gil Navarrete 29 Sep '06

29 Sep '06

Hi all I have implemented the load balance in mode statefull, I obtained the next messages: phone1 dispatcher sipserver phone2 --------INVITE--------> ----INVITE-----------> ----INVITE--------> <----100 TRYING---- <---100 TRYING---- <---100 TRYING---- <----180 RINGING-- <---180 RINGING--- <---180 RINGING--- <----200 OK---------- <----- 200 OK-------- <---200 OK-------- -------- ACK --------> --------- ACK --------> You can see that the sipserver never sends ACK message to video_phone2. Is this correct? Then I try the next, in the sipserver: if (method =="ACK"){ if (!t_relay()) { sl_reply_error(); }; return; } But the sipserver doesn't send the message ACK at the phone2? Does some body know what it's wrong? --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Users] How to get debug print out?
by Steve Xu 29 Sep '06

29 Sep '06

Hi folks, I setup "debug=7" in openser.cfg, however I still cannot get any detail debug print out in /var/log/messages. Here is how I launch the openser: openserctl start There is a logic errors in openser.cfg and I try to trace down the call step by step, how can I do it? Thanks for your time to read my question, I am really appreciated on it. Steve

3 3

[Serusers] just start SER
by vijay tiwari 28 Sep '06

28 Sep '06

hi all today i have install ser on my Linux machine Linux bishthost.bisht 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/Linux and its install everthing is ok and i found two file in /usr/local/etc/ser dictionary.ser ser.cfg please tell me can i start change in ser.cfg or take backup of this file and start working on mail ser.cfg and how can i used hello word ser.cfg file just start copy paste what can i do please guide me, which is best for me thanks vijay --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.

1 0

[Users] [openser] Displaying Caller-ID
by Tracy Lofton 28 Sep '06

28 Sep '06

Hi, This is probably a simply question for some of you, but is there a way to get openser to display the Caller-ID (telephone number) to the calling party's phone display screen, rather than displaying Annonymous? If so, how would I be able to do this? Thanks, Tracy _________________________________________________________________ Try the new Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&FORM=WL…

1 0

[Users] "BUG: io_watch_del" logs
by Federico Giannici 28 Sep '06

28 Sep '06

I have noticed that the following errors began to appear in the logs when OpenSER (1.1.0 under OpenBSD) is closed: Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 32, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 33, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 34, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 35, not in [0, 0) Sep 27 17:53:07 eowyn OpenSER[29102]: BUG: io_watch_del: invalid fd 36, not in [0, 0) It says "BUG", so is it serious? Bye. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 3

[Users] Can I avoid password openser.cfg ?
by Ronald Wiplinger 28 Sep '06

28 Sep '06

How can I avoid details in the openser.cfg? modparam("usrloc|auth_db","db_url","mysql://openser:openserrw@localhost/openser") bye Ronald

3 6

[Users] New Rules: no line wrapping
by Mark Kent 28 Sep '06

28 Sep '06

I'ld like to make a plea to everyone on this list: Please make sure that your mailer doesn't fold long lines in your outbound email. There are certain things that should be sent verbatim (although possibly "anonymized") for example, ngrep output and openser.cfg snippets. I hope I don't send mangled email... let me know if I do, but please make sure it's not your mail reader that is doing the mangling. Thanks, -mark

1 0

[Users] CPL module
by Nicolas Palix 28 Sep '06

28 Sep '06

Hi, I have installed openser and its CPL module and I use CPLEd to write and upload service. I have some problem with the timeout attribut of proxy XML node. It's seem to be not taken into account and the "no answer" node is interpreted about 2 min later. I set the timeout in the CPL script at 5 sec. After re-reading the CPL module documentation and check the cpl_proxy.h code, I try to set the "timer_avp" param without success. Can I have some information on how AVP and timer works ? What I should configured to have CPL enable installation of OpenSER ? Best regards -- Nicolas Palix http://phoenix.labri.fr/people/palix/

2 4

Re [serusers] Cannot Access Serweb
by Tracy Lofton 28 Sep '06

28 Sep '06

Hi All, I followed the intall directions completely and installed both runkit and classkit. I am using php 5.1.1 and when I go to http://myserver/serweb/html/admin/index.php, I get the following message: Function aggregate_methods() doesn't exists. Try install Classkit extension. http://pecl.php.net/package/classkit I can see the module sitting in the diretory as noted below: /usr/lib/php/modules/classkit.so php.ini file ; Directory in which the loadable extensions (modules) reside. extension_dir = "/usr/lib/php/modules" extension=classkit.so extension=runkit.so [runkit] runkit.internal_override=1 Does anyone have any idea of why the classkit module fails to load? Thanks for any help Tracy _________________________________________________________________ Get today's hot entertainment gossip http://movies.msn.com/movies/hotgossip

1 0

[Users] dbtext script problem
by Lindsey Hans Joseph Lao 28 Sep '06

28 Sep '06

Hello, I have a problem with sc.dbtext script. I understand that it is used to create text file for dbtext to store information such as location, address, etc. I downloaded from Openser CVS the sc.dbtext script and replaced my current openserctl file with it. Unfortunately, it still failed to run: Here's the error: /usr/local/sbin/openserctl: 463: Syntax error: Bad substitution Does anyone have an idea on how to resolve this? Thank you. Hans

2 4

[Users] dbtext openser
by Lindsey Hans Joseph Lao 28 Sep '06

28 Sep '06

Hello, I'm planning to use DBTEXT instead of mysql. I found a forum somewhere in the Internet that a script, serctl dbtext [dbtext path] I believe. Is this supported in OpenSER 1.1.0? Because when I tried to invoke this particular command, nothing happens. Thanks!

3 3

Re: [Users] Caller ID & Time of Day
by John Peters 28 Sep '06

28 Sep '06

Try configuring this on the endpoint. For instance, on Sipura ATAs you might set the following parameters if the location were using Eastern Standard Time: Time_Zone "GMT-05:00" ; Primary_NTP_Server "time-b.nist.gov" ; We just rolled out service, and a lot of users are complaining about the > time displayed on their phones being incorrect. Exactly 2 hours behind. > > The switch which we receive all of our inbound calls from is 2 hours > behind > us. Is it possible to adjust the time in the Caller ID so that their > phones > will display properly? >

2 1

[Users] trusted IP with accounting
by ram 28 Sep '06

28 Sep '06

Hi all iam trying to deploy trusted IP and Accounting so iam bit confused can some one clarify me how can i achive this PSTN--OpenerSER(myserver)------------------Customer(Asterisk or ser or any predictive Dialers)---their clients 1. how can i give access to the customer to call using his IP, and he can creat internal Number of users 2. how can i account to his IP address billing in Acc 3. how can i forward to DID from myserver to his server blindly 4. how can i bill that DID billing also, for incoming. any suggestiong will be great Ram

2 7

Re: [Users] AVPOPS error
by Bogdan-Andrei Iancu 28 Sep '06

28 Sep '06

ok -super :) bogdan Jose Silva wrote: > Thanks!!! your solution worked out fine. > > Thank you very mutch. > Jose Silva > > 2006/9/28, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>: > >> maybe your query is about 1024 char len, bot exceeding it all the >> time.... >> >> try to increase the buffer size - see modules/avpops/avpops_impl.c line >> 66 for >> #define AVP_PRINTBUF_SIZE 1024 >> >> note you need to recompile and re-install. >> >> regards, >> bogdan >> >> >> Jose Silva wrote: >> >> > Yes it's a very long query! >> > But the query works for the first messages, but after a while it fails >> > to work. >> > How can I correct this? >> > >> > thanks, >> > Jose Silva >> > >> > >> > 2006/9/28, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>: >> > >> >> Hi Jose, >> >> >> >> maybe you are trying to form a very long query...by default, the used >> >> buffer to build the query has a length of 1024 chars... >> >> is this the case?? >> >> >> >> regards, >> >> bogdan >> >> >> >> Jose Silva wrote: >> >> >> >> > Hi I'm using avpops module to do some DB queries, everything >> goes fine >> >> > and all the queries are executed correctly for the first >> messages, but >> >> > after receiving allot of messages what I get on my log for one >> of the >> >> > queries (just one of the queries, the other ones are executed >> >> > correctly) is the following. >> >> > >> >> > (...) >> >> > xl_printf: no more space for text [175] >> >> > 3(4820) xl_printf: buffer overflow -- increase the buffer size... >> >> > 3(4820) avpops:ops_dbquery_avps: error - cannot print the query >> >> > (...) >> >> > >> >> > How can I solve this problem? >> >> > >> >> > Thanks in advance, >> >> > Jose Silva >> >> > >> >> > _______________________________________________ >> >> > Users mailing list >> >> > Users(a)openser.org >> >> > http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >> >> >> >> >> >>

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[Users] Mysql error
by Nick De Cristofaro 28 Sep '06

28 Sep '06

Hey Im just trying to add users after doing the basic install. And I get this? : "ERROR 1062 (23000) at line 1: Duplicate entry '' for key 2 -e \E[37;31mERROR: introducing the new user 'test' to the database failed" Nick

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[Users] AVPOPS error
by Jose Silva 28 Sep '06

28 Sep '06

Hi I'm using avpops module to do some DB queries, everything goes fine and all the queries are executed correctly for the first messages, but after receiving allot of messages what I get on my log for one of the queries (just one of the queries, the other ones are executed correctly) is the following. (...) xl_printf: no more space for text [175] 3(4820) xl_printf: buffer overflow -- increase the buffer size... 3(4820) avpops:ops_dbquery_avps: error - cannot print the query (...) How can I solve this problem? Thanks in advance, Jose Silva

2 3

[Users] CPL module
by Nicolas Palix 28 Sep '06

28 Sep '06

Hi, I have installed openser and its CPL module and I use CPLEd to write and upload service. I have some problem with the timeout attribut of proxy XML node. It's seem to be not taken into account and the "no answer" node is interpreted about 2 min later. I set the timeout in the CPL script at 5 sec. After re-reading the CPL module documentation and check the cpl_proxy.h code, I try to set the "timer_avp" param without success. Can I have some information on how AVP and timer works ? What I should configured to have CPL enable installation of OpenSER ? Best regards -- Nicolas Palix http://phoenix.labri.fr/people/palix/

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[Users] rtpproxy still not working for me
by Ronald Wiplinger 28 Sep '06

28 Sep '06

Status: openser compiled, including mysql openser working with mysql (registering) openser server has two ethernet ports with one public and one private IP rtpproxy is running without any parameter 7311 is eyebeam on local desktop 7312 is gizmo on the same local desktop 7111 is a grandstream remote, behind a nat Openser adminstrator shows all three phones as registered with following sip addresses: sip:7111@192.168.10.91 sip:7311@192.168.250.108:42800;rinstance=f17a00051f63a625 sip:17476063396@proxy01.sipphone.com:5060 <=== that should be 7312(a)192.168.250.108 Internal calls: call from 7312 to 7311 works. call from 7311 to 7312 ends up at a voicemail box (most likely gizmo problem) call from 7311 to 17476063396(a)proxy01.sipphone.com ends up at the voicemail box External calls: call from 7311 (local phone) 7111 (remote phone behind NAT) does not work, see the ngrep info below. Looking at the output I feel that the nathelper is not working. As I understand the documents, the address should be replaced with an public IP address, but it uses my servers internal ethernet port (192.168.250.41) 5.32.86.166 in alt1 is the IP address of my Hamachi network. I have no idea, why this one shows up here. # U 2006/09/28 08:56:06.482374 192.168.250.108:42800 -> 192.168.250.41:5060 INVITE sip:7111@192.168.250.41 SIP/2.0. Via: SIP/2.0/UDP 192.168.250.108:42800;branch=z9hG4bK-d87543-a177451df022e84f-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:7311@192.168.250.108:42800>. To: "7111"<sip:7111@192.168.250.41>. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=13766900. Call-ID: ZGIzNjlhYmIwODBmOTAzMDJiMDU1Zjc0ZTZhNDM0MzE.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 1005g stamp 32576. Content-Length: 809. . v=0. o=- 0 2 IN IP4 192.168.250.108. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.250.108. t=0 0. m=audio 59382 RTP/AVP 107 100 106 6 0 105 18 3 5 101. a=alt:1 2 : eE24SyNT p1rs0KkA 5.32.86.166 59382. a=alt:2 1 : eKP8Z+Ou +E2ejbdN 192.168.250.108 59382. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:B8DA69C497FC40B2AACE87E3E86D914A. m=video 46168 RTP/AVP 125. a=alt:1 2 : Ac/XN/Nt wQExOeCB 5.32.86.166 46168. a=alt:2 1 : ZKv1sx6w JY5CEQdN 192.168.250.108 46168. a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800. a=rtpmap:125 H264/90000. a=sendrecv. a=x-rtp-session-id:AD24C300DC234CCFA115E5F0B36CDEAB. # U 2006/09/28 08:56:06.483471 192.168.250.41:5060 -> 192.168.250.108:42800 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.250.108:42800;branch=z9hG4bK-d87543-a177451df022e84f-1--d87543-;rport=42800. To: "7111"<sip:7111@192.168.250.41>. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=13766900. Call-ID: ZGIzNjlhYmIwODBmOTAzMDJiMDU1Zjc0ZTZhNDM0MzE.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=8787 req_src_ip=192.168.250.108 req_src_port=42800 in_uri=sip:7111@192.168.250.41 out_uri=sip:7111@192.168.10.91 via_cnt==1". . # U 2006/09/28 08:56:30.829647 192.168.250.108:42800 -> 192.168.250.41:5060 . . # U 2006/09/28 08:56:36.004956 192.168.250.41:5060 -> 192.168.250.108:42800 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 192.168.250.108:42800;branch=z9hG4bK-d87543-a177451df022e84f-1--d87543-;rport=42800. To: "7111"<sip:7111@192.168.250.41>;tag=fa997f81440371de71ab448ebdb9af56-1877. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=13766900. Call-ID: ZGIzNjlhYmIwODBmOTAzMDJiMDU1Zjc0ZTZhNDM0MzE.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=8791 req_src_ip=192.168.250.108 req_src_port=42800 in_uri=sip:7111@192.168.250.41 out_uri=sip:7111@192.168.10.91 via_cnt==0". . # /var/log/messages shows: Sep 28 09:00:27 taipei /usr/local/sbin/openser[8787]: ERROR: force_rtp_proxy2: no available proxies Sep 28 09:00:40 taipei /usr/local/sbin/openser[8787]: ERROR: unforce_rtp_proxy: no available proxies Sep 28 09:00:40 taipei /usr/local/sbin/openser[8787]: ERROR: unforce_rtp_proxy: no available proxies Where is my mistake? bye Ronald

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[Users] proxy authorization required sent to the wrong port
by Benko 28 Sep '06

28 Sep '06

Hello! I've tried to modify my setup to use rtpproxy, following the guide posted at http://siprouter.onsip.org/doc/gettingstarted/, most of it works fine, but i have a small but annoying problem: When i receive a invite from a UA, i verify the identity with proxy_authorize. However, the proxy-authenticate-request is not sent to the originating sip-port of the UA but to 5060 for some reason. In my old setup this worked flawless, the proxy authorize request is sent to the correct external port of the phone. To me it seems there is no difference in this part of the route-plan, so how does this happen? I have attached both config-files and a ngrep of the requests(*.old is the working setup, *.rtpproxy is the new setup that doesn't work), maybe someone can help me with this? Thanks in advance! Regards Christian

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Re: [Users] OPENSER 1.1.0 equivalent version for SER
by Bogdan-Andrei Iancu 28 Sep '06

28 Sep '06

Hi Bill, Bill Zhang wrote: > Hi, > > I just sew the new face of www.iptel.org <http://www.iptel.org/> and > see they start to move things forward now. Got some questions about > SER versus OPENSER: > > 1. What is the OPENSER 1.1.0 equivalent version for SER? 0.10.x or > 0.9.x? Is there a lot of difference? > OpenSER 1.1.0 is the latest stable release. AFAIK, last stable release for SER is 0.9.x for more than one year ago.... ..yes there are a lot of differences... > 1. SER has a SEMS server, does OPENSER has something equivalent? > SEMS can be also used with OpenSER > > 1. When will OPENSER have the new Presence module? I know it’s > soon, but how soon? > in a matter of weeks - are you interested in testing it? Regards, Bogdan

1 0

[Users] Openser Administrator misses /var/log/openser
by Ronald Wiplinger 28 Sep '06

28 Sep '06

Openser Administrator says: No such file or directory - /var/log/openser - I agree with that! Where do I set it? tail /var/log/messages Sep 27 19:59:41 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 19:59:41 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7211> Sep 27 19:59:44 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 19:59:47 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7211> Sep 27 19:59:50 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 19:59:59 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7113> Sep 27 19:59:59 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7211> Sep 27 20:00:02 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7111> Sep 27 20:00:03 taipei /usr/local/sbin/openser[13042]: ERROR:registrar:update_contacts: invalid cseq for aor <7113> Sep 27 20:00:07 taipei /usr/local/sbin/openser[13044]: ERROR:registrar:update_contacts: invalid cseq for aor <7113> BTW, what do these error messages mean? bye Ronald

2 1

[Users] t_on_failure does not work for me
by Jeremie Le Hen 28 Sep '06

28 Sep '06

Hi list, my configuration file contains the following block: % if (is_method("INVITE")){ % force_rtp_proxy(""); % t_on_failure("1"); % append_hf("P-hint: RTP session ports reserved by rtpproxy\r\n"); % }; If I dump the SIP dialog on the network, I can indeed see the "P-hint" header. Then, the configuration file contains: % failure_route[1] { % if (!(status == "407")) { % unforce_rtp_proxy(); % append_hf("P-hint: RTP session dropped\r\n"); % } else { % append_hf("P-hint: Proxy authentication required, don't drop RTP session\r\n"); % } % } Unfortunately, when a 407 reply code comes back from the registrar, the outgoing SIP packet doesn't have any "P-hint" header. I didn't manage to test the other cases, unfortunately. Did I miss something in the configuration file ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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RE: [Serusers] Implementing call fwd
by Kamal.Mann＠t-systems.com 28 Sep '06

28 Sep '06

Hi all Please help out ________________________________ From: Mann, Kamal Sent: Tuesday, September 26, 2006 5:46 PM To: 'serusers(a)lists.iptel.org' Subject: RE: [Serusers] Implementing call fwd Hi all I am implementing Blind call forwarding with 0.9.6 SER version. But with current configuration it's not working. Please help on same. I am enclosing small part of usr_prefrences table & my ser.cfg.

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Re: [Serusers] Implementing call fwd
by Greger V. Teigre 28 Sep '06

28 Sep '06

Please copy the list. It depends on what you want to achieve. Blind forwarding is to me the following: - Check if this is a local user (myself) - Check if this is an alias (lookup("aliases")) - Check if the AOR is registered (lookup("location")) - Check if the user has blind call forwarding - If no, relay to looked up location - If yes, rewrite ruri. (Then what you do depends on your setup complexity, either you forward to PSTN, so just rewrite to GW and relay, or you allow recursive forwarding. Then you need to either relay to yourself or call a route that runs the above described process. Or simpler, you can do as you do by duplicating the lookups, now with the new ruri. Then you allow only one step of forwarding, not recursive.) g-) Kamal.Mann(a)t-systems.com wrote: > > I think so, because we need to check is r-uri (to which call get > forwarded) is registered or not. > > I am newbie if anything wrong please suggests me. > > > > Thanks in anticipation > > Kamal Mann > > ** ** > > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Wednesday, September 27, 2006 8:19 PM > *To:* Mann, Kamal > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Implementing call fwd > > > > You do lookup("aliases") and lookup("location") AFTER you have changed > the uri through avp_pushto from usr_preferences. > Is this really what you want? > g-) > > Kamal.Mann(a)t-systems.com <mailto:Kamal.Mann@t-systems.com> wrote: > > > Hi All > I want to setup Blind call forwarding. Like if a uri - mann(a)iptel.org <mailto:mann@iptel.org> > is not registered with SER but this user had already configured in ser > database table usr_prefrences that his call should be forwarde to > another uri kamal(a)iptel.org <mailto:kamal@iptel.org>. For that my usr_prefrence table is like > > > >> USR_PREFRENCES table >> >> +------+----------+-----------+-----------+---------------------+------ >> > + > >> ---------------+ >> >> | uuid | username | domain | attribute | value | type >> > | > >> modified | >> >> +------+----------+-----------+-----------+---------------------+------ >> > + > >> ----------------+ >> >> | | bhaskar | debian | callfwd | kamal(a)iptel.org <mailto:kamal@iptel.org> >> | 0 | 20060926165852 | >> >> | | mann | iptel.org | callfwd | sip:kamal@iptel.org <mailto:sip:kamal@iptel.org> | 0 >> > | > >> 20060926150443 | >> >> | | prateek | iptel.org | callfwd | kamal(a)iptel.org <mailto:kamal@iptel.org> | 0 >> > | > >> 20060926170701 | >> >> +------+----------+-----------+-----------+---------------------+------ >> > + > >> ----------------+ >> > > > Is this entry is correct?? Is there any flaw in my ser.cfg?? > Any help would be appreciated. > Regards > Kamal Mann > -----Original Message----- > From: Steve Blair [mailto:blairs@isc.upenn.edu] > Sent: Wednesday, September 27, 2006 2:26 PM > To: Mann, Kamal > Cc: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > Subject: Re: [Serusers] Implementing call fwd > > > > Does your config file relay REFER messages? It needs to in order to > perform the transfer. > > Kamal.Mann(a)t-systems.com <mailto:Kamal.Mann@t-systems.com> wrote: > > >> Hi all >> >> Please help out >> >> >> > ------------------------------------------------------------------------ > >> *From:* Mann, Kamal >> *Sent:* Tuesday, September 26, 2006 5:46 PM >> *To:* 'serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org>' >> *Subject:* RE: [Serusers] Implementing call fwd >> >> Hi all >> >> I am implementing Blind call forwarding with 0.9.6 SER version. But >> with current configuration it's not working. Please help on same. >> >> I am enclosing small part of usr_prefrences table & my ser.cfg. >> >> ----------------------------------------------------------------------- >> > - > >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Users] Mediaproxy / proxydispatcher: couldn't retrieve DNS SRV record
by kjcsb 28 Sep '06

28 Sep '06

I am getting the following error in the log file: Sep 28 06:24:25 beta proxydispatcher[24129]: error: couldn't retrieve DNS SRV record '_mediaproxy._tcp.domain.co.nz': (111, 'Connection refused') However I think the DNS records are OK. dig SRV _mediaproxy._tcp.domain.co.nz ; <<>> DiG 9.2.3 <<>> SRV _mediaproxy._tcp.domain.co.nz ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 63894 ;; flags: qr rd ra; QUERY: 1, ANSWER: 3, AUTHORITY: 2, ADDITIONAL: 3 ;; QUESTION SECTION: ;_mediaproxy._tcp.domain.co.nz. IN SRV ;; ANSWER SECTION: _mediaproxy._tcp.domain.co.nz. 86100 IN SRV 0 100 25060 c-mp-03.domain.co.nz. _mediaproxy._tcp.domain.co.nz. 86100 IN SRV 10 50 25060 c-mp-01.domain.co.nz. _mediaproxy._tcp.domain.co.nz. 86100 IN SRV 10 50 25060 c-mp-02.domain.co.nz. ;; AUTHORITY SECTION: domain.co.nz. 258929 IN NS ns2.widge.net. domain.co.nz. 258929 IN NS ns1.widge.net. ;; ADDITIONAL SECTION: c-mp-03.domain.co.nz. 86129 IN A 60.234.nnn.nnn ns1.widge.net. 170576 IN A 82.165.160.211 ns2.widge.net. 170576 IN A 82.165.163.115 ;; Query time: 83 msec ;; SERVER: 64.62.196.100#53(64.62.196.100) ;; WHEN: Thu Sep 28 06:28:07 2006 ;; MSG SIZE rcvd: 276 On the target server (c-mp-03.domain.co.nz) there's something getting through to to mediaproxy as shown in the log file: Sep 28 06:26:21 domain01 mediaproxy[16128]: command status Sep 28 06:26:21 domain01 mediaproxy[16128]: command execution time: 0.54 ms Sep 28 06:26:41 domain01 mediaproxy[16128]: command status Sep 28 06:26:41 domain01 mediaproxy[16128]: command execution time: 0.55 ms Could anyone advise how to resolve this issue? Regards Cameron

2 1

[Serusers] unhandled SIP state transition
by jorain 28 Sep '06

28 Sep '06

Hi all, When i call from pstn to sip phone, and hang up it before someone pick up the sip phone, then my sems will show this error and immediately end the sems. Error: changeSIPState(IsdnCall.cpp:339):unhandled SIP state transition(inbound) from (DISCONNECTED|DISCONNECTED) to (CONNECTED|DISCONNECTED) What could be the problem? Any help is appreciated. Thanks Regards, jorain

1 0

[Users] OPENSER 1.1.0 equivalent version for SER
by Bill Zhang 27 Sep '06

27 Sep '06

Hi, I just sew the new face of www.iptel.org <http://www.iptel.org/> and see they start to move things forward now. Got some questions about SER versus OPENSER: 1. What is the OPENSER 1.1.0 equivalent version for SER? 0.10.x or 0.9.x? Is there a lot of difference? 2. SER has a SEMS server, does OPENSER has something equivalent? 3. When will OPENSER have the new Presence module? I know it's soon, but how soon? Thanks in advance. Best Regards, Bill

1 0

[Users] Re: Problem handling multiple calls in FreeBSD 6.1-RELEASE
by Mark Kent 27 Sep '06

27 Sep '06

> >The problem is easy to reproduce: just reducing the children parameter > >to 1, and openser will not handle a second call (obviously that doesn't > >happen in Debian, using the same configuration for openser). I just tested this on FreeBSD 6.1-RELEASE, using openser 1.1.0, and I could place two simultaneous calls without any issues or delays. As you may know, freebsd has a few different thread libraries. I made sure all my compiles, for everything in and around openser, was compiled with -pthread on the command line. -mark

1 0

Re: [Users] OpenserAdmin Can't Login
by John Shellhouse 27 Sep '06

27 Sep '06

Hi, First, let me thank you for your assistance and time with my problem. But I am not sure what UR configuration of openser you are speaking of. in the config/database.yml I have indicated this info: development: adapter: mysql database: openser username: openser password: openserrw host: localhost and did the same for production in database.yml. There was not much to do. what other config could it be. How did you do your UR configuration of openser? Thanks, jwshel >From: "raviprakash sunkara" <sunkara.raviprakash.feb14(a)gmail.com> >To: "John Shellhouse" <jwshel(a)hotmail.com> >Subject: Re: [Users] OpenserAdmin Can't Login >Date: Wed, 27 Sep 2006 18:27:54 +0530 > >Username is openser >passwors is openserrw > >My Guess is that Ur configuaration of openser admin is not Clear > > > >On 9/27/06, John Shellhouse <jwshel(a)hotmail.com> wrote: >> >>Hi, >> >>I do not understand what you are describing, can you be more specific? I >>did try to login at: >>http://localhost:3000/login/login, but password and username not accepted. >> >>jwshel >> >> >> >From: "raviprakash sunkara" <sunkara.raviprakash.feb14(a)gmail.com> >> >To: "John Shellhouse" <jwshel(a)hotmail.com> >> >Subject: Re: [Users] OpenserAdmin Can't Login >> >Date: Wed, 27 Sep 2006 14:51:12 +0530 >> > >> >Hello , >> >U can't configured the openser admin 0.3 the Database.dl in that :P >> > >> >Try this ............. >> >http://localhost:3000/login/login >> > >> > >> >On 9/27/06, John Shellhouse <jwshel(a)hotmail.com> wrote: >> >> >> >>Hi, >> >> >> >>Second request, I cannot login to openser Admin. Followed directions, >> >>added >> >>ruby rails, gems, etc. started ./script/server entered default >> >>username:password of openser:openserrw. >> >> >> >>This is the only result: >> >>All login in attempts are logged. Unauthorized access is forbidden. >> >> >> >>Need help by someone, anyone even will be appreciated. >> >> >> >>thanks >> >>jswhel >> >> >> >> >> >> >From: "John Shellhouse" <jwshel(a)hotmail.com> >> >> >To: users(a)openser.org >> >> >Subject: [Users] [openser] OpenserAdmin Login >> >> >Date: Sun, 24 Sep 2006 05:57:01 -0700 >> >> > >> >> >Hi, >> >> > >> >> >I am cannot login to the initial Webpage for OpenserAdmin after >>starting >> >> >./script/server. I put in the default username:password of >> >> >openser:openserrw but I am not allowed access. Any suggestions >>anyone? >> >> > >> >> >Thanks, >> >> > >> >> >jswhel >> >> > >> >> > >> >> > >> >> >_______________________________________________ >> >> >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 >> >> >> > >> > >> > >> >-- >> >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 >> >> >> >>_______________________________________________ >>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

1 0

[Users] Caller ID & Time of Day
by Kelvin Williams 27 Sep '06

27 Sep '06

We just rolled out service, and a lot of users are complaining about the time displayed on their phones being incorrect. Exactly 2 hours behind. The switch which we receive all of our inbound calls from is 2 hours behind us. Is it possible to adjust the time in the Caller ID so that their phones will display properly? Kw

1 0

Re: [Users] mysql.so missing
by Ronald Wiplinger 27 Sep '06

27 Sep '06

Largo wrote: > Hi Ronald > > That means U didint compile the > mysql module ! > > Fix it like this : > > - compiling all the packets : > > *make all* > > - make with a prefix : > > *make prefix=/usr/local/ install* > > > - now install the excluded packets, as example here mysql for mysql > database or pgsql for postgresql database support: > > *make modules include_modules=”mysql”* > > *make install include_modules=”mysql”* > Was the same error at the time I compiled it, ... I had to do first: "unset CPU" This made and installed the module mysql.so > ** > > **** > > **** > > - copying the init script to the “/etc/init.d” does not lead to a > result. The init script already in /usr/local/sbin to the init.d > directory as shown in the following line: > > *ln -s /usr/local/sbin/openser /etc/init.d/openser* > Are you sure? shouldn't it be openserctl? I tried ln -s /usr/local/sbin/openserctl /etc/rc.d/openser (SuSE 10.1 has all init files in /etc/rc.d) # /etc/rc.d/openser start Starting OpenSER : ERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed Where have I lost the pid file? BTW, Largo, you have a nice email address !!! bye Ronald

1 2

[Serusers] Customize SERWeb
by Legos Engineering 27 Sep '06

27 Sep '06

Dear all, I find a way to add a new column in accounting page. So in this way, I think that I can create a new table with my price/prefix and then make a calcul base with the duration of call. But maybe someone knows if this already exist. If not, what's the best way to do something like that ? Help will appreciated. Thanks. Best regards

1 0

[Users] Problem handling multiple calls in FreeBSD 6.1-RELEASE
by Hernán A. Pérez Masci 27 Sep '06

27 Sep '06

Hi there, We've been running openser for a long time on Debian Sarge boxes without problems, but recently we installed it (version 1.0.1) on a FreeBSD 6.1-RELEASE box, and started having problems when testing it even with very low load conditions. The problem is easy to reproduce: just reducing the children parameter to 1, and openser will not handle a second call (obviously that doesn't happen in Debian, using the same configuration for openser). Is this a known issue? I haven't found any message related to it. Regards, Hernan.

2 2

[Users] mysql.so missing
by Ronald Wiplinger 27 Sep '06

27 Sep '06

taipei:/usr/local/etc/openser # openser ERROR: bad config file (1 errors) /var/log/messages shows: Sep 27 19:23:16 taipei openser: ERROR: load_module: could not open module </usr/local/lib64/openser/modules/mysql.so>: /usr/local/lib64/openser/modules/mysql.so: cannot open shared object file: No such file or directory Sep 27 19:23:16 taipei openser: parse error (39,13-14): failed to load module mysql.so exists at /usr/lib64/php5/extensions/mysql.so However, a symlink does not work for that either. I use SuSE 10.1 on a Semperon (Athlon) CPU. I have installed mysql-devel. Besides this, I am confused about the openser.cfg file, which does not match the SER - Getting Started. Is there a better document available? What do I need to install to get the system running with phones behind NAT ? The document suggests many different things (Nathelper, STUN, rtpproxy and mediaproxy). Which one do I need? Is it either rtpproxy or mediaproxy? Is it in addition to STUN or Nathelper???? I also do not really know what I need. My server is on a public IP, with a second Ethernet card for the internal LAN. Where the user has his phone I do not know yet. Maybe on a symetric NAT or asymetric NAT, maybe on a public IP. For all it should work. bye Ronald (obviously confused)

2 1

[Users] openser working with pre-paid and post-paid?
by moacir FERRARI NETO 27 Sep '06

27 Sep '06

Hello guys... It would like that to openser it worked with service of pre-paid end post-paid... I need a "how to" install this service. I saw many diagrams working with B2BUA and I still with many doubts about it. thanks, Moacir. _______________________________________________________ Você quer respostas para suas perguntas? Ou você sabe muito e quer compartilhar seu conhecimento? Experimente o Yahoo! Respostas ! http://br.answers.yahoo.com/

1 0

[Serusers] t_on_retransmission route
by Hendrik Scholz 27 Sep '06

27 Sep '06

Hi! We were poking the idea around of having an optional route that would be invoked upon a retransmission. The fr timers all fire way too late if you'd have to implement some sort of fallback if your peering partner or PSTN gateway died. I magine a routing mechanism much like t_on_failure() / failure_route() to allow something like this: - set transmission timer via AVP / t_set_fr() like - t_on_retransmission(1); - t_relay_to_udp() and then you'd have something like retransmission_route[1] { # select new gateway t_relay_to_udp() } The retransmission route should be invoked if no provisional (i.e. 100 Trying) nor final response was received within the given interval. I'm not sure about how to handle TCP connections, though. A simple TCP ACK might be enough to assume the other side is up and happy to answer our request. This would allow for redirecting traffic without having to wait for a final response (408 timeout). The timer value would be fr_timer - fr_inv_timer starting at the same time as fr_timer itself. Last but not least I'm not sure about the impact of having quite a lot of timers firing after a couple of msecs. If anybody thinks this would make sense I'd give it a try ;) Cheers, Hendrik -- Hendrik Scholz - <hscholz(a)raisdorf.net> - http://www.wormulon.net/ drag me, drop me - treat me like an object

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Re: [Users] OpenserAdmin Can't Login
by John Shellhouse 27 Sep '06

27 Sep '06

Hi, I do not understand what you are describing, can you be more specific? I did try to login at: http://localhost:3000/login/login, but password and username not accepted. jwshel >From: "raviprakash sunkara" <sunkara.raviprakash.feb14(a)gmail.com> >To: "John Shellhouse" <jwshel(a)hotmail.com> >Subject: Re: [Users] OpenserAdmin Can't Login >Date: Wed, 27 Sep 2006 14:51:12 +0530 > >Hello , >U can't configured the openser admin 0.3 the Database.dl in that :P > >Try this ............. >http://localhost:3000/login/login > > >On 9/27/06, John Shellhouse <jwshel(a)hotmail.com> wrote: >> >>Hi, >> >>Second request, I cannot login to openser Admin. Followed directions, >>added >>ruby rails, gems, etc. started ./script/server entered default >>username:password of openser:openserrw. >> >>This is the only result: >>All login in attempts are logged. Unauthorized access is forbidden. >> >>Need help by someone, anyone even will be appreciated. >> >>thanks >>jswhel >> >> >> >From: "John Shellhouse" <jwshel(a)hotmail.com> >> >To: users(a)openser.org >> >Subject: [Users] [openser] OpenserAdmin Login >> >Date: Sun, 24 Sep 2006 05:57:01 -0700 >> > >> >Hi, >> > >> >I am cannot login to the initial Webpage for OpenserAdmin after starting >> >./script/server. I put in the default username:password of >> >openser:openserrw but I am not allowed access. Any suggestions anyone? >> > >> >Thanks, >> > >> >jswhel >> > >> > >> > >> >_______________________________________________ >> >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 >> > > > >-- >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] Ser vovida B2BUA
by prabhakar g 27 Sep '06

27 Sep '06

I'm looking for Ser vovida B2BUA & Radius howto, related documentation. Can anybody help me. -- Prabhakar.g Network Engineer Hyderabad-34,India

3 2

[Serusers] Voice recorder?
by Tobias Lindgren 27 Sep '06

27 Sep '06

Hi all, I'm looking for a good voice recorder software (OpenSource pref.) that can record a conversation in a phonecall. Please let me know if you have tested one. Br, Tobias

1 0

[Users] mediaproxy signIn IP address
by Klaus Darilion 27 Sep '06

27 Sep '06

Hi! It looks like there is some restriction of the IP address allowed to signIn to mediaproxy. Can this be configured or is this hardcoded? thanks klaus

1 0

Re: [Serusers] Nathelper Ping OPTIONS
by Greger V. Teigre 27 Sep '06

27 Sep '06

:-D If she really is beautiful, I would take whatever she could give... However, this one doesn't want to talk to you. You need to find some tricks. g-) olivier wrote: > The beautifullest girl of the world can't give you what she doesn't > have ;) > > Thanks anyway, > > Olivier > > Greger V. Teigre a écrit : >> Yes, I saw that. Last time I tried, it worked. I couldn't see what >> possibly is wrong from the output and still can't. Sorry. >> g-) >> >> olivier.taylor wrote: >>> Hi Greger, >>> >>> I have tried to apply the patch to 0.9.6, see the result below :( >>> >>> I have downloaded the last 0.9.6. >>> >>> Olivier >>> >>> Greger V. Teigre a écrit : >>>> The patch should work for 0.9.6. AFAIK, 0.10.x already has the >>>> OPTIONS support. >>>> g-) >>>> >>>> olivier.taylor wrote: >>>>> Hello, >>>>> >>>>> just trying to enable the ping functionality in nathelper to send >>>>> SIP message (i.e. OPTIONS) instead of 4-byte UDP packets >>>>> on SER 0.9.6 and 0.10.0 >>>>> >>>>> as shown below, patch doesn't apply >>>>> Is there another possibility to have this feature? >>>>> >>>>> Olivier >>>>> >>>>> >>>>> patch -p1 < nathelper_ping.patch >>>>> Hmm... Looks like a unified diff to me... >>>>> The text leading up to this was: >>>>> -------------------------- >>>>> |diff -Naur -x CVS nathelper/nathelper.c nathelper_ping/nathelper.c >>>>> |--- nathelper/nathelper.c 2006-08-16 08:45:48.000000000 +0200 >>>>> |+++ nathelper_ping/nathelper.c 2006-08-10 14:54:14.000000000 +0200 >>>>> -------------------------- >>>>> Patching file nathelper.c using Plan A... >>>>> Hunk #1 failed at 118. >>>>> Hunk #2 failed at 172. >>>>> Hunk #3 failed at 193. >>>>> Hunk #4 failed at 212. >>>>> Hunk #5 failed at 239. >>>>> Hunk #6 failed at 257. >>>>> Hunk #7 failed at 267. >>>>> Hunk #8 failed at 276. >>>>> Hunk #9 failed at 320. >>>>> Hunk #10 failed at 367. >>>>> Hunk #11 failed at 1543. >>>>> 11 out of 11 hunks failed--saving rejects to nathelper.c.rej >>>>> Hmm... The next patch looks like a unified diff to me... >>>>> T >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> Serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> >>>> >>>> >>> >>> >> >> > >

1 0

[Serusers] Carrier-grade framework for SER
by Andreas Granig 27 Sep '06

27 Sep '06

Hi guys, I'd like to propose another possibility for a highly-available and scalable system design based on SER: The problems I've encountered for scalable systems are: - Distribution of the user location and alias location among the nodes (user location is based on registrations, alias location comes from web interfaces and is used for call forwarding). - Reloading up to date location tables after breakdown and recovery of a node So I'm just thinking loud about the following provisioning system: - Write a client which fulfills the this demands: - Receive one or more locations from SER via a SER module or from a web application and distribute them to other known clients. Take care of retransmissions if a client isn't reachable or reports a temporary failure. - Receive one or more locations from other clients and write them into the SER FIFO. If writing into the FIFO fails, try to write directly into the database (location-table, alias-table etc.). Report a temporary failure if this also fails. Maybe a centralized server should be used which receives the locations from the clients and distributes them to other clients, so that the nodes just know about the server and nothing about other nodes. This would make integration of new nodes easy. On the other hand, it's another single point of failure, so a decentralized solution should be considered. But that would mean that you've to inform every node about the existence of a new node. The protocol used between the nodes should be simple and fast. So I think SOAP drops out here. Maybe XMLRPC or ICE (http://www.zeroc.com/ice.html) could be used. One might think now why not just use replication on SIP layer, but t_replicate only supports one peer and you've no possibility to get locations on a node while it's down. Replication of other location tables like the alias-table is also not possible. I'd be willing to release these parts as GPL for creating an open framework for carrier-grade SER integration, so any feedback, improvements or flames are highly welcome. Cheers, Andy

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[Users] auth ACK and BYE?
by Mark Kent 27 Sep '06

27 Sep '06

I just noticed that my SIP phone sends authentication credentials for ACK and BYE in a session where the INVITE required authentication credentials. I recognize this as a good thing... but it also surprised me. Currently, I don't handle it in my openser.cfg... I neither ask for the credentials nor do I strip them off before passing the message down the line... other than for INVITE. I'm going to correct this in my config, and I thought it worth mentioning for others just in case others also hadn't considered it. -mark

2 1

[Serusers] digit matching
by Richard C. Thompson 27 Sep '06

27 Sep '06

Hi All I hope someone can point me in the right direction or just give me a hint. I have the following route in ser.cfg and it works great. All I would like to do is change it so it matches on any number that starts with 002 regardless of length not just 12 more digits or a fixed length. if(uri=~"sip:(002............)@") I tried if(uri=~"sip:(002*)@") but it didn't work TIA Rick

4 3

[Users] Beer in oslo tomorrow
by Atle Samuelsen 27 Sep '06

27 Sep '06

This is just a beer reminder.. Remember beer in Oslo tomorrow. :) Topic: ser 0.10 _A

2 1

[Serusers] Callfwd example not routing
by Murray Hooper 27 Sep '06

27 Sep '06

I am trying to get a call forwarding to a PSTN gateway. I see the SIP call come in but appears I am missing something to get the routing/call forward to be recognized. Below is SIP log. Sep 26 13:04:52 localhost ./ser[7102]: SIP Request: Sep 26 13:04:52 localhost ./ser[7102]: method: <INVITE> Sep 26 13:04:52 localhost ./ser[7102]: uri: <sip:5199571788@216.183.86.133> Sep 26 13:04:52 localhost ./ser[7102]: version: <SIP/2.0> Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=1 Sep 26 13:04:52 localhost ./ser[7102]: Found param type 232, <branch> = <z9hG4bK0a45e9d0>; state=6 Sep 26 13:04:52 localhost ./ser[7102]: Found param type 235, <rport> = <n/a>; state=17 Sep 26 13:04:52 localhost ./ser[7102]: end of header reached, state=5 Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: Via found, flags=1 Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: this is the first via Sep 26 13:04:52 localhost ./ser[7102]: After parse_msg... Sep 26 13:04:52 localhost ./ser[7102]: preparing to run routing scripts... Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=128 Sep 26 13:04:52 localhost ./ser[7102]: end of header reached, state=9 Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: get_hdr_field: <To> [33]; uri=[sip:5199571788@216.183.86.133] Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: to body [<sip:5199571788@216.183.86.133> ] Sep 26 13:04:52 localhost ./ser[7102]: get_hdr_field: cseq <CSeq>: <102> <INVITE> Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: get_hdr_body : content_length=218 Sep 26 13:04:52 localhost ./ser[7102]: found end of header Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: is_maxfwd_present: max_forwards header not found! Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=64 Sep 26 13:04:52 localhost ./ser[7102]: check_via_address(216.143.130.61, 216.143.130.61, 0) Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=33554432 Sep 26 13:04:52 localhost ./ser[7102]: DEBUG: add_param: tag=as35382f6d Sep 26 13:04:52 localhost ./ser[7102]: end of header reached, state=29 Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=256 Sep 26 13:04:52 localhost ./ser[7102]: find_first_route: No Route headers found Sep 26 13:04:52 localhost ./ser[7102]: loose_route: There is no Route HF Sep 26 13:04:52 localhost ./ser[7102]: is_local(): Realm '216.183.86.133' is not local Sep 26 13:04:52 localhost ./ser[7102]: is_local(): Realm '216.143.130.61' is not local Sep 26 13:04:52 localhost ./ser[7102]: parse_headers: flags=-1 Sep 26 13:04:52 localhost ./ser[7102]: check_via_address(216.143.130.61, 216.143.130.61, 0) Sep 26 13:04:52 localhost ./ser[7102]: DEBUG:destroy_avp_list: destroying list (nil) Sep 26 13:04:52 localhost ./ser[7102]: receive_msg: cleaning up Sep 26 13:04:52 localhost ./ser[7098]: SIP Request: Sep 26 13:04:52 localhost ./ser[7098]: method: <ACK> Sep 26 13:04:52 localhost ./ser[7098]: uri: <sip:5199571788@216.183.86.133> Sep 26 13:04:52 localhost ./ser[7098]: version: <SIP/2.0> Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: flags=1 Sep 26 13:04:52 localhost ./ser[7098]: Found param type 232, <branch> = <z9hG4bK0a45e9d0>; state=6 Sep 26 13:04:52 localhost ./ser[7098]: Found param type 235, <rport> = <n/a>; state=17 Sep 26 13:04:52 localhost ./ser[7098]: end of header reached, state=5 Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: Via found, flags=1 Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: this is the first via Sep 26 13:04:52 localhost ./ser[7098]: After parse_msg... Sep 26 13:04:52 localhost ./ser[7098]: preparing to run routing scripts... Sep 26 13:04:52 localhost ./ser[7098]: parse_headers: flags=4 Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: add_param: tag=7770e7624e56f146f4f19d50775e8c7e.230c Sep 26 13:04:52 localhost ./ser[7098]: end of header reached, state=29 Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: get_hdr_field: <To> [75]; uri=[sip:5199571788@216.183.86.133] Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: to body [<sip:5199571788@216.183.86.133>] Sep 26 13:04:52 localhost ./ser[7098]: DEBUG: sl_filter_ACK : local ACK found -> dropping it! Sep 26 13:04:52 localhost ./ser[7098]: DEBUG:destroy_avp_list: destroying list (nil) Sep 26 13:04:52 localhost ./ser[7098]: receive_msg: cleaning up

2 1

[Serusers] Nathelper Ping OPTIONS
by olivier.taylor 27 Sep '06

27 Sep '06

Hello, just trying to enable the ping functionality in nathelper to send SIP message (i.e. OPTIONS) instead of 4-byte UDP packets on SER 0.9.6 and 0.10.0 as shown below, patch doesn't apply Is there another possibility to have this feature? Olivier patch -p1 < nathelper_ping.patch Hmm... Looks like a unified diff to me... The text leading up to this was: -------------------------- |diff -Naur -x CVS nathelper/nathelper.c nathelper_ping/nathelper.c |--- nathelper/nathelper.c 2006-08-16 08:45:48.000000000 +0200 |+++ nathelper_ping/nathelper.c 2006-08-10 14:54:14.000000000 +0200 -------------------------- Patching file nathelper.c using Plan A... Hunk #1 failed at 118. Hunk #2 failed at 172. Hunk #3 failed at 193. Hunk #4 failed at 212. Hunk #5 failed at 239. Hunk #6 failed at 257. Hunk #7 failed at 267. Hunk #8 failed at 276. Hunk #9 failed at 320. Hunk #10 failed at 367. Hunk #11 failed at 1543. 11 out of 11 hunks failed--saving rejects to nathelper.c.rej Hmm... The next patch looks like a unified diff to me... T

2 3

[Serusers] How to restrict only one call per user at the same time
by Ricardo Carvalho 27 Sep '06

27 Sep '06

Hi all, In an environment where the same user can be registered at the same time in multiple phones, how may I restrict users to only be able to make one call simultaneously? I mean... only make other call after hanging up the last one, never simultaneously made from different phones. How may I implement that? Regards, Ricardo.

2 1

[Serusers] Re: Ser Radius Howto
by Alex Fler 27 Sep '06

27 Sep '06

Hello All, I'm looking for Ser Radius howto, or any other related documentation. For some reason docs were removed from iptel website. Regards Alex Fler

2 1

[Serusers] Call Forwarding Using PSTN Gateway
by Murray Hooper 27 Sep '06

27 Sep '06

Newbie to SER. I am a little confused about how much I have to set up in databases to get an incoming SIP call to call forward to a PSTN number through SER. I have setup the username, attribute and value for my number, "callfwd", sip:1234567890@gateway.com in usr_preferences and the username in the subscriber file , added gateway ip to trusted, but call is immediately dropped when it arrives at SER. I have been through the getting started and have had success in modifying the values in the scripts, but don't see what data must be added into MySQL. Getting pointed in the right direction would be greatly appreciated Thanx murray

2 1

[Serusers] Difference between subscription handed by rls and pa
by Quan Huo Sheng 27 Sep '06

27 Sep '06

Hi; I am using ser-0.10.99-dev35-pa-4.1 to talk with SIP client(UA) and XCAP server for handling subscription and publish. Anyone can give me some comments for: 1. Difference between subscription handled by rls (handle_rls_subscription) and pa (handle_subscription). 2. Difference between table ser.subscriber and ser.rls_subscription. Thanks Smile [This e-mail is confidential and may be priviledged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---

2 1

[Serusers] rls, xcap and remote storage
by Quan Huo Sheng 27 Sep '06

27 Sep '06

Hi; I have a softphone (ie. Eyebeam) that talks with SER. The question is whether Eyebeam can upload or download the contact list/buddy list file in xml format from XCAP server via RLS modules of SER. Please advise. Regards Smile [This e-mail is confidential and may be priviledged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---

2 1

[Users] want a sample openser.cfg can forward calls to Asterisk (PSTN)
by Steve Xu 27 Sep '06

27 Sep '06

Hi folks, I have Asterisk as a PSTN termination GW, Openser will handle the user registration. Openser users has name format 7xxxxxxxx ( all digistes), which can call each other peer to peer ( it is a basic openser function). I want to forward the call ( 1xxxxxxxxxx or 011xxxxxxxxxxxxx ) to Asterisk when openser user try to terminate US or International calls. I tried to use the setting like this: if(is_method("INVITE") && !has_totag() && uri=~"sip:011[0-9]+@.*") { route(4); exit; }; route[4] { strip(3); # route to Asterisk Media Server rewritehostport("222.222.222.79:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } OpenSER keep looking the 011xxxxxxxxx from location and got 408 always , it does not forward the call to Asterisk. I read a doc about how to share acc and voice meail with Asterisk, will it have to use failure_route to accomplish it ? I am new to Openser and dont know much about the routing logic. Anybody could point me to the right way? A sample cfg will really appreciated. Steve

1 0

[Users] [openser] OpenserAdmin Login
by John Shellhouse 27 Sep '06

27 Sep '06

Hi, I am cannot login to the initial Webpage for OpenserAdmin after starting ./script/server. I put in the default username:password of openser:openserrw but I am not allowed access. Any suggestions anyone? Thanks, jswhel

1 1

FW: [Serusers] digit matching
by Richard C. Thompson 26 Sep '06

26 Sep '06

To All I answered it myself if(uri=~"sip:002[0-9]*@") Thanks Rick _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard C. Thompson Sent: Tuesday, September 26, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] digit matching Hi All I hope someone can point me in the right direction or just give me a hint. I have the following route in ser.cfg and it works great. All I would like to do is change it so it matches on any number that starts with 002 regardless of length not just 12 more digits or a fixed length. if(uri=~"sip:(002............)@") I tried if(uri=~"sip:(002*)@") but it didn't work TIA Rick

1 0

[Users] avp_write with $ruri and /username /domain
by Papadopoulos Georgios 26 Sep '06

26 Sep '06

hello, Trying to migrate from 1.0.0 to 1.1.0 I came across a problem that has already been discussed in the list (of course if I had first searched the archives I would have saved myself from a couple of days of debbuging). Here are the two discussions: http://openser.org/pipermail/users/2006-August/006246.html http://openser.org/pipermail/users/2006-August/006284.html I am just sending this again because I think it looks like a bug to me. In the avpops documentation for 1.0.0 the example says avp_write("$ruri/username","$email"); In 1.1.0 it has changed to avp_write("$ru/username","$avp(email)"); Neither of those two will produce the username in avp email with openser 1.1.0. The same holds for avp_write("$ru/domain","$avp(whatever)"); avp_write("$ru/domain","$avp(whatever)"); The solution is to use: avp_write("$rU","$avp(whatever)"); avp_write("$rd","$avp(whatever)"); If the /username, /domain are not supposed to work, then shouldn't they be removed from the document? And shouldn't they cause some syntax error? thanks a lot George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] how to make SER send Digest-Username(115) attribute
by a.kolesnikov 26 Sep '06

26 Sep '06

Is it possible to make SER to send Digest-Username(115) attribute defined in RFC 4590 instead or in addition to Sip-Uri-Users(208) attribute? С уважением, Колесников Андрей Netris <http://www.netris.ru> http://www.netris.ru

2 1

[Serusers] Help needed to connect to a PSTN Gateway
by Ralf Eisele 26 Sep '06

26 Sep '06

Hi, I want connect my SER-Server to a public PSTN Gateway. The sample config found in 'SER - Getting Started' doesn't work. Someone out there who can help me? Mit besten Gruessen Ralf Eisele --------- eXtension World Wide Connections GmbH Partner der abbex-group.ag Finninger Strasse 56 Fon: +49 731 9216333 D-89231 Neu-Ulm Fax: +49 731 9216335 Germany Mobile: +49 179 2991333 http://www.extension.net - info(a)extension.net PGP Fingerprint: C29E 9EE9 6A36 8EDA 7146 1613 0BA8 9080 5B4A 36A3

1 0

[Users] gen_CA.sh
by Gregoire 26 Sep '06

26 Sep '06

Hi everyone! I am running openser 1.1.0-tls installed with debian package. I am trying to configure correctly TLS, and generate certificate, but in tutorials I found, people talk about a script located in /etc/openser/tls/tools named "gen_rootCA.sh", but I don't have the directory "tools" in /etc/openser/tls. Does someone have any idea about this script? Another thing is when I tried to send a mail to a user with the command: openserctl mail <username> Openser makes errors like: Write email to mail: now ... mail: You must specify direct recipients with -s, -c, or -b ERROR: sending message failed I had to change the line 1777 in /usr/sbin/openserctl in send_email $2 instead of send_email $1 to make it work. Greg

2 1

[Users] NEW SIP-to-XMPP (Google Talk/Jabber) gateway
by Daniel-Constantin Mierla 26 Sep '06

26 Sep '06

Hello, thanks to Andrea Spirea, a new SIP-to-XMPP gateway was included in OpenSER. There is a new module, XMPP, which enables SIP users to chat with any XMPP user (Google Talk, Jabber) user in the world. The design permits flexibility in configuration and deployments, the new module can at alone as a XMPP server or can work in tandem with a standalone XMPP server (for more complex communication scenarios like TLS ...). Also, the module can be plugged in your main SIP server or can be deployed alone, enabling in a easy manner geographic distribution and redundancy. There is no need of mapping SIP addresses to XMPP addresses, no need to store translation profiles, everything is done in the fly. The conversation will survive to restarts, the session being recovered form the messages. As a result, there is no need of a persistent storage, the footprint is very small, embedding the gateway in small devices should be straightforward. The news is posted here: http://www.openser.org/index.php?option=com_content&task=view&id=56&Itemid=9 Module's documentation: http://www.openser.org/docs/modules/1.2.x/xmpp.html Feedback will be very appreciated, the next steps are to enhance the standalone-xmpp-server version of the module with other features of a xmpp server and to add support for presence and voice. Testing and reports are always welcome. Cheers, Daniel

2 2

[Serusers] store in AVP something searched with a regular expression
by Ricardo Carvalho 26 Sep '06

26 Sep '06

How may I store in AVP something searched with a regular expression from SIP messages? Regards, Ricardo

1 0

[Users] Loose route...again
by Helge Waastad 26 Sep '06

26 Sep '06

Hi, I've just a quick question regarding the loose_route function We had a discussion earlier this year when the answer for loose_route was: "by default, loose_route() function detects the pre-loaded route: if there is only one Route hdr pointing to the server, it will be consumed and loose_route() will return false. If more than one Route hdr is present, the routing will be done as usual - function will consume the Route and set the destination to the next Route - true is returned." I just wonder about the "returning false" since I always see loose_route return true...is this due to the lr tag? eks: My yyy.yyy.yyy.yyy receives a BYE message BYE sip:67512391@10.46.11.132 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK84fb.7836a4d5.0 Via: SIP/2.0/UDP 10.46.11.101:5060;rport=5060;branch=z9hG4bK7cae83600f656a72 Route: <sip:yyy.yyy.yyy.yyy:5060;lr> >From the explaiantion I would guess if (loose_route()){ do-something} should not do-something??? I hope Im not totally out of my head right now br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 1

[Users] pdt problem
by Francesco Castellano 26 Sep '06

26 Sep '06

Dear list members, I've a problem trying to get pdt module working. These are the log I get with debug=9. pdt_print_tree: [sip.mydomain.com] pdt_print_node: [2] [proxy01.sipphone.com] pdt_print_node: [2393] [other.host.at.domain.com] pdt_print_node: [51] [my.host.at.domain.com] PDT:pdt_sync_cache: ... PDT:pdt_sync_cache: sync op[1]=1... prefix alredy allocated pdt_add_to_tree internal error! PDT:pdt_sync_cache: Error to insert into tree PDT:prefix2domain: cannot update the cache I'm working with openser 1.1.0-notls, and pdt.c version "v 1.8 2006/07/03 15:26:17 miconda Exp". MySQL description of pdt table is: +---------+--------------+------+-----+---------+-------+ | Field | Type | Null | Key | Default | Extra | +---------+--------------+------+-----+---------+-------+ | sdomain | varchar(255) | NO | PRI | NULL | | | prefix | varchar(32) | NO | PRI | NULL | | | domain | varchar(255) | NO | | NULL | | +---------+--------------+------+-----+---------+-------+ but the association sdomain,prefix,domain seems to be correct, as the pdt_print log lines report. Strange behaviour also occurs when I try to list from fifo the table every two times once I get: host:~# openserctl fifo pdt_list database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd pdt_list 400 pdt_fifo_list - domain not found FIFO command was: :pdt_list:openser_receiver_25901 while the second time openserctl hangs until I kill it with ctrl+c. Any suggestion will be helpful, thanks Francesco Castellano

2 1

[Users] db-script not working
by Gerd Feiner 26 Sep '06

26 Sep '06

hi list, i have strange problems with the openser_mysql.sh script: # openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': sip.de creating database openser ... ERROR 1133 (42000) at line 7: Can't find any matching row in the user table Install SERWEB tables ?(y/n): for some reason it does not do anything besides creating a database (with no tables in it). how am i supposed to create the database instead? -- mit freundlichen Grüßen / with best regards Gerd Feiner Leitung IP-Dienste Kabelfernsehen München ServiCenter GmbH & Co. KG Abt. Mediaport Medienallee 24 85774 Unterföhring T: +49 89 95083 243 F: +49 89 95083 132 E: g.feiner(a)cablesurf.de W: www.cablesurf.de

2 1

[Users] fix_nated_contact not rewriting contact?
by kjcsb 26 Sep '06

26 Sep '06

I have the following code in openser.cfg if (client_nat_test("3")) { #IP address in contact is private IP #Via is different to address client has contacted us from #If so then setflag7 to indicate that sender is behind NAT xlog("client nat test 3 returned true for r-uri <$ru> in route 3 so will setflag7"); setflag(7); xlog("contact before update <$ct>"); xlog("via before update <$hdr(via)>"); force_rport(); #Rewrite Contact: header to contain Sip client's public IP:port fix_nated_contact(); xlog("contact is now <$ct>"); xlog("via is now <$hdr(via)>"); }; However the value returned by $ct is the same before and after fix_nated_contact(). Is that because I'm doing something wrong, or because I've misunderstood fix_nated_contact or because the $ct variable doesn't get updated by fix_nated_contact? If the latter, how can I find out the result of fix_nated_contact? The same applies to force_rport and the via details. Any advice appreciated. Regards Cameron

2 1

[Users] undefined symbol: mem_lock
by Ovidiu Sas 26 Sep '06

26 Sep '06

Hi, I'm trying to build openser 1.1.0 (notls) for arm for Linksys NSLU - OpenSLUG OS. The cross compilation went ok, but when I try to run the program I encounter the following error: root@LKG7BE78E # openser -c 0(4517) ERROR: load_module: could not open module </usr/lib/openser/modules/sl.so>: /usr/lib/openser/modules/sl.so: undefined symbol: mem_lock 0(4517) parse error (41,13-14): failed to load module Here are the compile flags: NOTE: make -e MAKEFLAGS= bin-target=/usr/sbin doc-target=/usr/share/doc/openser lib-target=/usr/lib/openser/ modules-target=/usr/lib/openser/modules/ cfg-target=/etc/openser/ prefix=/usr cfg-prefix=/ ARCH=armeb LD=ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb all bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex Compiling action.c ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -isystem/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/include -Os -fomit-frame-pointer -frename-registers -fno-unroll-loops -fno-inline-functions -fweb -funit-at-a-time -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"armeb"' -DOS='"linux"' -DCOMPILER='"ccache 3.4.4"' -D__CPU_armeb -D__OS_linux -D__SMP_no -DCFG_DIR='"/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DUSE_SYSV_SEM -DHAVE_SIGIO_RT -DHAVE_SELECT -c action.c -o action.o ... Linking openser ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -L/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-rpath-link,/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-O1 action.o core_stats.o crc.o daemonize.o data_lump.o data_lump_rpl.o dprint.o dset.o error.o fifo_server.o flags.o forward.o io_wait.o ip_addr.o items.o items_extra.o main.o md5.o md5utils.o modparam.o msg_translator.o pass_fd.o proxy.o qvalue.o re.o receive.o resolve.o route.o route_struct.o script_cb.o serialize.o socket_info.o sr_module.o statistics.o tcp_main.o tcp_read.o timer.o tsend.o udp_server.o unixsock_server.o usr_avp.o mem/f_malloc.o mem/mem.o mem/memtest.o mem/q_malloc.o mem/shm_mem.o mem/vq_malloc.o parser/hf.o parser/msg_parser.o parser/parse_allow.o parser/parse_content.o parser/parse_cseq.o parser/parse_disposition.o parser/parse_diversion.o parser/parse_event.o parser/parse_expires.o parser/parse_fline.o parser/parse_from.o parser/parse_hname2.o parser/parse_hostport.o parser/parse_methods.o parser/parse_nameaddr.o parser/parse_param.o parser/parse_refer_to.o parser/parse_rpid.o parser/parse_rr.o parser/parse_sipifmatch.o parser/parse_sst.o parser/parse_supported.o parser/parse_to.o parser/parse_uri.o parser/parse_via.o parser/parser_f.o parser/digest/digest.o parser/digest/digest_parser.o parser/digest/param_parser.o parser/contact/contact.o parser/contact/parse_contact.o db/db.o db/db_fifo.o db/db_id.o db/db_pool.o lex.yy.o cfg.tab.o -ldl -lresolv -o openser ... make[1]: Entering directory `/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/work/openser-1.1.0-notls-r0/openser-1.1.0-notls/mod ules/sl' Compiling sl.c ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -fPIC -DPIC -isystem/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/include -Os -fomit-frame-pointer -frename-registers -fno-unroll-loops -fno-inline-functions -fweb -funit-at-a-time -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"armeb"' -DOS='"linux"' -DCOMPILER='"ccache 3.4.4"' -D__CPU_armeb -D__OS_linux -D__SMP_no -DCFG_DIR='"/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DUSE_SYSV_SEM -DHAVE_SIGIO_RT -DHAVE_SELECT -c sl.c -o sl.o ... Linking sl.so ccache armeb-linux-gcc -march=armv5te -mtune=xscale -mthumb-interwork -mno-thumb -shared -L/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-rpath-link,/home/admin/slug-3.10/releases/slugos-3.10-beta/openslug-nslu2.tmp/staging/armeb-linux/lib -Wl,-O1 sl.o sl_cb.o sl_funcs.o -o sl.so Here's the output of the nm -D for openser: root@LKG7BE78E:/usr/sbin# nm -D /usr/sbin/openser U _IO_getc 00066f38 R _IO_stdin_used w _Jv_RegisterClasses U __assert_fail U __ctype_b_loc w __deregister_frame_info U __dn_expand U __errno_location U __fxstat w __gmon_start__ 00089000 A __init_array_end 00089000 A __init_array_start U __libc_current_sigrtmax U __libc_current_sigrtmin U __libc_start_main U __lxstat w __register_frame_info U __res_init U __res_search U __res_state U __xstat U abort U accept U alarm U atoi U bind U chdir U chmod U chown U chroot U clearerr U close U closelog U connect U ctime U difftime U dlclose U dlerror U dlopen U dlsym U exit U fclose U fcntl U fdopen U ferror U fflush U fgets U fileno U fopen U fork U fprintf U fputc U fputs U fread U free U freopen U fscanf U fwrite U getcwd U geteuid U getgrnam U gethostbyaddr U gethostbyname U gethostbyname2 U getopt U getpgid U getpid U getprotobyname U getpwnam U getrlimit U getsockname U getsockopt U inet_pton U ioctl U isatty U kill U listen U malloc U memchr U memcmp U memcpy U memmove U memset U mkfifo U mktime U mmap U munmap U open U openlog 0008b444 B optarg 0008b434 B opterr 0008b448 B optopt U pathconf U pause U perror U poll U printf U putchar U puts U rand U read U realloc U recv U recvfrom U recvmsg U regcomp U regexec U regfree U select U semctl U semget U semop U send U sendmsg U sendto U setbuf U seteuid U setgid U setrlimit U setsid U setsockopt U setuid U sigaddset U sigdelset U sigemptyset U sigismember U signal U sigprocmask U sigtimedwait U sleep U snprintf U socket U socketpair U srand 0008b43c B stderr 0008b440 B stdin 0008b438 B stdout U strcasecmp U strchr U strcmp U strcpy U strdup U strerror U strlen U strncasecmp U strncmp U strncpy U strtol U syslog U system U time U uname U unlink U vfprintf U vsnprintf U wait U waitpid U writev root@LKG7BE78E:/usr/sbin# nm -D /usr/lib/openser/modules/sl.so U MDStringArray 0000b260 A _DYNAMIC w _Jv_RegisterClasses 0000b60c A __bss_end__ 0000b570 A __bss_start 0000b570 A __bss_start__ w __cxa_finalize 0000b454 D __data_start w __deregister_frame_info 0000b60c A __end__ U __errno_location w __gmon_start__ w __register_frame_info 0000b60c A _bss_end__ 0000b570 A _edata 0000b60c A _end 00002b68 T _fini 000011e4 T _init U add_lump_rpl U build_res_buf_from_sip_req U crcitt_string_array U debug 000018b0 T destroy_slcb_lists U dprint U err2reason_phrase 0000b460 D exports U fm_free U fm_malloc U get_send_socket U get_ticks U log_facility U log_stderr U mem_block U mem_lock U memcmp U memcpy U memset U mhomed 0000b4e8 D mod_stats 0000b560 D module_flags 0000b564 D module_version 0000b5e4 B modules U parse_headers 0000b5e0 B pgid_file 0000b5f0 B pid_file U prev_ser_error U print_dset 0000b600 B rcv_acks U register_script_cb 00001900 T register_slcb U reply_to_via 000019e0 T run_sl_callbacks U semop 0000b5f4 B sent_err_rpls 0000b5ec B sent_rpls U shm_block 0000b45c D sl_enable_stats 00002868 T sl_filter_ACK 000026a8 T sl_reply_error 00001f58 T sl_send_reply 00001ef0 T sl_shutdown 00001d3c T sl_startup 0000b58c B slcb_hl U stat_lock U strerror U strlen U syslog U tcp_disable 0000b608 B tcp_listen U tcp_send 0000b5d8 B tx_1xx_rpls 0000b5d4 B tx_2xx_rpls 0000b5dc B tx_3xx_rpls 0000b5f8 B tx_4xx_rpls 0000b5e8 B tx_5xx_rpls 0000b5fc B tx_6xx_rpls 0000b604 B udp_listen U udp_send U update_sock_struct_from_via root@LKG7BE78E:/usr/sbin# A similar problem was encounter and fixed for OpenBSD .... Any help would be greatly appreciated. -ovi

2 1

[Serusers] Prepaid and SERWeb
by Legos Engineering 26 Sep '06

26 Sep '06

Hi, I find something about prepaid system for my SERWeb. In fact, I want to print the price of communication in the accounting page for my final customers. I'm thinking about develop myself a db with price by minute and do the calcul on demand in accounting page. But if something already exist, can you tell me more about it ? Thanks

1 0

[Users] What causes loose_route() to return false ?
by Jeremie Le Hen 26 Sep '06

26 Sep '06

Hi list, in numerous example configuration files, I saw the following statement (where "route(1)" is the final routing block): % if (loose_route()) { % route(1); % } % % ... % % route(1); It works for me, but I am just wondering what causes the loose_route() function to return a failure ? In this case, it seems number of actions used to be performed before calling the final message routing block. Naturally, I checked the rr module documentation, but the descriptions is very small and only refers to RFC 3161. I have read some parts of this RFC, thus I know what loose routing means (basically, it uses always the same URI all along the proxy string, removing IP addresses in the Route header gradually, contrariwise to strict routing which puts the real URI at the end of the Route header and use the next proxy as URI). In spite of this, I cannot figure why this function should fail. I'm pretty confident this is to help for the scripting logic, and I was supplied with some explanations. Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 1

[Users] Call gets always busy!
by Ronald Wiplinger 26 Sep '06

26 Sep '06

I have set-up two Grandstream phones, one is inside my LAN and one is outside. Below is the ngrep output. Can you please explain me both (openserctl ul show and ngrep) and where the problem my be? 7111 calls 7211, both are registerd as: ...Record(0x2b68516ca938)... domain: 'location' aor : '7111' ~~~Contact(0x2b68516ca9b0)~~~ domain : 'location' aor : '7111' Contact : 'sip:7111@192.168.250.115' Expires : 404 q : Call-ID : '748276c266577824(a)192.168.250.115' CSeq : 10157 User-Agent: 'Grandstream GXV3000 1.0.0.24' received : '' Path : '' State : CS_NEW Flags : 0 Sock : 192.168.250.41:5060 (0x610fc0) Methods : 7807 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0x2b68516d0950)... domain: 'location' aor : '7211' ~~~Contact(0x2b68516cb7d0)~~~ domain : 'location' aor : '7211' Contact : 'sip:7211@192.168.87.103' Expires : 2382 q : Call-ID : '8d96c35345b355d7(a)192.168.87.103' CSeq : 10001 User-Agent: 'Grandstream GXV3000 1.0.0.24' received : '' Path : '' State : CS_NEW Flags : 0 Sock : 59.124.210.41:5060 (0x610ef0) Methods : 7807 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... # U 2006/09/26 10:49:27.574906 192.168.250.115:5060 -> 192.168.250.41:5060 INVITE sip:7211@192.168.250.41 SIP/2.0. Via: SIP/2.0/UDP 192.168.250.115;branch=z9hG4bKf777932528b49881. From: <sip:7111@192.168.250.41>;tag=d71217459a54cde3. To: <sip:7211@192.168.250.41>. Contact: <sip:7111@192.168.250.115>. Supported: replaces, timer. Call-ID: 76740d0122a46445(a)192.168.250.115. CSeq: 15245 INVITE. User-Agent: Grandstream GXV3000 1.0.0.24. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 438. . v=0. o=7111 8000 8000 IN IP4 192.168.250.115. s=SIP Call. c=IN IP4 192.168.250.115. t=0 0. m=audio 15004 RTP/AVP 0 3 18 4 8. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=ptime:20. m=video 15006 RTP/AVP 99. a=sendrecv. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4C/ID=. a=framerate:15. # U 2006/09/26 10:49:27.587708 192.168.250.41:5060 -> 192.168.250.115:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.250.115;branch=z9hG4bKf777932528b49881. From: <sip:7111@192.168.250.41>;tag=d71217459a54cde3. To: <sip:7211@192.168.250.41>. Call-ID: 76740d0122a46445(a)192.168.250.115. CSeq: 15245 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=31770 req_src_ip=192.168.250.115 req_src_port=5060 in_uri=sip:7211@192.168.250.41 out_uri=sip:7211@192.168.87.103 via_cnt==1". . # U 2006/09/26 10:49:56.862266 192.168.250.41:5060 -> 192.168.250.115:5060 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 192.168.250.115;branch=z9hG4bKf777932528b49881. From: <sip:7111@192.168.250.41>;tag=d71217459a54cde3. To: <sip:7211@192.168.250.41>;tag=fa997f81440371de71ab448ebdb9af56-37d0. Call-ID: 76740d0122a46445(a)192.168.250.115. CSeq: 15245 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=31771 req_src_ip=192.168.250.115 req_src_port=5060 in_uri=sip:7211@192.168.250.41 out_uri=sip:7211@192.168.87.103 via_cnt==0". . # U 2006/09/26 10:49:56.865291 192.168.250.115:5060 -> 192.168.250.41:5060 ACK sip:7211@192.168.250.41 SIP/2.0. Via: SIP/2.0/UDP 192.168.250.115;branch=z9hG4bKf777932528b49881. From: <sip:7111@192.168.250.41>;tag=d71217459a54cde3. To: <sip:7211@192.168.250.41>;tag=fa997f81440371de71ab448ebdb9af56-37d0. Contact: <sip:7111@192.168.250.115>. Call-ID: 76740d0122a46445(a)192.168.250.115. CSeq: 15245 ACK. User-Agent: Grandstream GXV3000 1.0.0.24. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Length: 0. . #

1 0

[Serusers] retrieve data from database
by jorain 26 Sep '06

26 Sep '06

Hi all, I want to pull data from mysql table inside ser.cfg. can we declare variable within route()? which module will be used? Is it exec module? Thanks. regards, jorain

1 0

[Serusers] Problem with ATA 186
by Teran Jesus 25 Sep '06

25 Sep '06

Hello, i have a problem with the ATA 186 in the SER 0.9.6, when i generate a call from my ata, when the ATA received the OK from the SER, the ATA don't generates the ACK correctly, this missing the user in the ACK, something like this, ACK sip:XXX.XXX.XXX.XXX SIP/2.0, i believe the sip poroxy is missing something in the OK, because i have tested this in the SER version 0.8.12 and this don't have the same problem. Do you know how can i resolv the problem?

2 1

[Serusers] Does Ser support Multi Call-Legs accounting?
by Ricardo Carvalho 25 Sep '06

25 Sep '06

Hi all, As seen in the following URL for OpenSer, I wonder if Ser supports Multi Call-Legs accounting, (feature that OpenSer already implements): http://openser.org/docs/modules/1.1.x/acc.html#MULTI-CALL-LEGS I have my system just already in production using Ser, but only now I realised that this functionality maybe lacking in Ser!... I tried to use the directive: modparam("acc", "multi_leg_enabled", 1) in my ser.cfg file, but doing so, Ser doesn't start! Is there any other way to enable in Ser this feature? How? Regards, Ricardo.

3 3

Re: [Serusers] Performance
by Jiri Kuthan 25 Sep '06

25 Sep '06

At 23:59 25/09/2006, Weiter Leiter wrote: >SER goes past a couple of thousands CPS into simple proxying setups. >However, in most common setups, the storage backends prove to add quite a drag. Glad to second that opinion. Well -- it is easy to explain, for one SIP transaction one has modulo six database transactions. SIP is about text processing. Database is about I/O. So what are you going to optimize first? :-) -jiri >There isn't (yet?) a benchmarking SER config. >Hence the question: how did you test? > >On 9/25/06, Victoria Cortez <<mailto:ser_list@conexiongroup.com>ser_list(a)conexiongroup.com> wrote: >Hi all, > > I've been making some performance tests with Ser on a huge and powerfull hardware. I couldn't reach 1000 CPS. Did anybody reached or past this amount of traffic with Ser? > > Thanks in advance, > >Víctor > > > >_______________________________________________ >Serusers mailing list ><mailto:Serusers@lists.iptel.org>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 -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Commerical Support SER IP->PSTN Gateway
by Murray Hooper 25 Sep '06

25 Sep '06

Hi Folks I am looking for anyone that is familiar with getting SER to automatically call forward all calls coming in via IP to call forward to PSTN Gateway with given telno. I am trying to get this up quickly to prove viability of concept.

3 2

[Serusers] Writing in acc table information from SIP messages
by Ricardo Carvalho 25 Sep '06

25 Sep '06

Hi, How may I edit ser.cfg so that every time Ser receives an 302 reply, it writes in acc table the "SIP contact address" of the forwarded URI address that comes in that message? I think that can be done with avp_write that searches that contact URI address in SIP messages and stores it in a variable, and than using avp_db_store to write the contents of that variable to the database, isn't it? But how can I do that? Any help? Regards, Ricardo.

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[Users] NEW: ACC module - new multi-leg format
by Bogdan-Andrei Iancu 25 Sep '06

25 Sep '06

Hi everybody, with the new ACC changes, I introduced a new format for multi-leg accounting. The overall behaviour is the same, but it's a more flexible approach - instead of having a fix logging frame (source and destination), the script writer has the liberty to define whatever and how many fields to be logged for each leg. All this is controlled via a single module parameter - the "multi_leg_info". See: http://openser.org/docs/modules/1.2.x/acc.html#AEN248 Regards, Bogdan

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[Users] Re: Users Digest, Vol 16, Issue 53
by Mark Kent 25 Sep '06

25 Sep '06

Andrew And wrote: > How is DNS resolution If I have some Openser destination and I use > t_relay()? and Klaus replied: > according to RFC3263. Depending on the URI (numerical/domain, with or > without port, with or without transport protocol) it us NAPTR+SRV+AAAA/A and my question (with a GeorgeBush squint) is: "What?" Suppose I get this on sip.example.com: INVITE 14085551212(a)sip.example.com What do I need to do in openser to find out if I should t_relay that to a VoIP destination, because 14085551212 is known in the e164 hierarchy (both e164.arpa and e164.org), versus just shipping it off to my pstn gateway? Thanks, -mark

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[Serusers] how to get ser 0.10
by olivier.taylor 25 Sep '06

25 Sep '06

how to get ser 0.10 sorry for the noob question :( Olivier

2 1

[Serusers] Presence
by senthil.nachimuthu＠bt.com 25 Sep '06

25 Sep '06

Hi, I want to configure SER <ser/about> 0.9.6 in such a way that It can proxy presence messages(SUBSCRIBE,PUBLISH) to presence server. Can you let me know how can I do that? Thanks and Regards, Senthil Kumar, Presence Capability, BT-Service Execution Platform, B62-MH,Orion Building,Adastral Park,Ipswich Work:+44 (0)1473 605 668 Mobile:+44 (0)7772287447

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[Serusers] USE XLOG
by Legos Engineering 25 Sep '06

25 Sep '06

Hello, I am a student and i do a workplacement. So i must work on the VOIP, but ihave no experience in. I would like know thank to a log, when an user connect/disconnect his user ugent. I know that i must use xlog, but i'm lost for many reasons. The first : Idon't know variables i must use to do what i want. The second : I don't know where i must write the line in ser.cfg to have this log. The third : Where is the log generated by xlog, and if there isn't one, how i can create this log. The fourth : If i must write something in xlog.so, how open it. Because when i open it with VI or NANO there is only encrypted data. THX to help me. If you can answer on my mail : sylvain.nigen(a)legos.fr, because it's easier ton know if i have an answer than to search the answer on SER USER. THANKS

2 1

[Serusers] ipv6 support
by P Jayasree 25 Sep '06

25 Sep '06

hi, I;ve installed ser 0.9.x version on linux ipv6 enabled PC.Now the ser is running.But it says it is listening on ipv4 address on udp and tcp.How do i know whether it is listening on ipv6 ?do i need to configure for anything ,for it to listen on ipv6 adress also! Kindly help in this issue. Regards, Jayasree The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail immediately and permanently delete the message and any attachments.

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[Users] NAT: minimise media proxying whilst maximising usability
by kjcsb 25 Sep '06

25 Sep '06

I am trying to reach a suitable trade-off between minimising the amount of proxied media and minimising one-way/no audio for all clients including NATed ones. When a UA that has been configured to use STUN sends a register message to SER the behaviour differs depending on the device. Some, such as a Sipura device, send the private address in the Via and the public address in the Contact: U 60.234.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-64d76b15..From: <sip:6435488773@domain.com >;tag=4cdc1208fa650916o0..To: <sip:6435488773@domain.com>..Call-ID: 1143f50-45857c6e@192.168.0.14..CSeq: 1 REGISTER..Max-Forwards: 70..Contact: <sip:6435488773@60.234.nnn.nnn:5060>;expires=60..User-Agent: Sipura/SPA3000-2.0.13(GWg)..Content-Length: 0..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura.... Others, such as a Grandstream device, send the public address in both the Via and Contact: U 60.234.nnn.nnn:32483 -> 147.202.nnn.nnn:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 60.234.nnn.nnn:32483;branch=z9hG4bK113cfc09c6acd189..From: "User" <sip: 5551234(a)domain.com>;tag=dede17c9405d4e0e..To: <sip:5551234@domain.com>..Contact: <sip:5551234@60.234.nnn.nnn:32483>.. Call-ID: fc200867c9a03268@192.168.0.10..CSeq: 100 REGISTER..Expires: 60..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... I am using the opsip.org configuration file together with mediaproxy. In the case of the Sipura, flag(6) is set in route[2] since client_nat_test("7") evaluates to true and so any calls with this device will be proxied. From ready various posts on the topic I understand this is a very safe configuration since it will deal with virtually every scenario. However it is also very expensive since it will proxy audio in situations where it is not really required. For example a call between two Sipura devices will be proxied when, due to STUN, it is unecessary in most cases. As I understand it the cases when it would be necessary are: - both UAs behind the same NAT with the SER proxy outside the NAT and router doesn't support hairpinning - devices behind symmetric NATs On the other hand, flag(6) will not be set with the Grandstream and so audio will not be proxied when this device calls another device that is either not behind NAT or is another NATed Grandstream. This is great in most cases but audio problems could arise in the cases described above. What I am therefore thinking is that by default I will not proxy media for devices where a public IP address has been determined. But I will create an avp that specifies by user whether to enable the standard onsip.org config. Also there could even be a dial prefix (e.g. *666) that turns proxying on. route[2] #Do this by default if (!search("^Contact: [ ]*\*") && client_nat_test("1") { etc #Do this if avp set or prefix dialled if (!search("^Contact: [ ]*\*") && client_nat_test("7") { etc I would appreciate any comments on the above. Have I understood the issues correctly? Will this type of configuration work? What are the disadvantages? Regards Cameron

4 4

[Serusers] Problem in testing MSILO module. (Please response..Really need help!!)
by Meidiana Ten 25 Sep '06

25 Sep '06

Hi there, Good afternoon.. I'd like to ask some questions depending on my last project problems.. maybe I had ever asked it in this forum, but I quite don't understand the solution you guys have told me. Maybe it's because I don't give the details about my OS and many more, and I'm a newbie too. But I really thanx for the reply on my last question. I'm using Red Hat 9 Linux to build my SIP Server. I use SER 0.9.7 as the server. The problem I found here is to test the MSILO module which I add to the ser.cfg Below is the description of my ser.cfg: ------------------------------------------------------- debug=9 fork=no log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") alias="pcr.ac.id" 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(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ------------------------------------------------------- I don't know whether my configuration below is right or still wrong..I've test it using 2 clients. I made one of them online, and the other off. I try to call the offline one, but it said user not found. In my thought, if I've activated MSILO module, it will automatically ask the online client if he wants to store offline voice message or not... Please tell me what's wrong with the configuration. In addition, the server can be use to make a call between online clients. Second question is about my serweb. I have seen my config.php in /var/www/html/serweb directory. It contains as shown below: ------------------------------------------------------- /* Unless you used brute-force to change SER table names */ $this->table_subscriber="subscriber"; $this->table_pending="pending"; $this->table_grp="grp"; $this->table_aliases="aliases"; $this->table_location="location"; $this->table_missed_calls="missed_calls"; $this->table_accounting="acc"; $this->table_phonebook="phonebook"; $this->table_event="event"; $this->table_netgeo_cache="netgeo_cache"; $this->table_ser_mon="server_monitoring"; $this->table_ser_mon_agg="server_monitoring_agg"; $this->table_message_silo="silo"; / below is the problem / $this->table_voice_silo="voice_silo"; ------------------------------------------------------- And below is my SER tables: +-----------------------+ | Tables_in_ser | +-----------------------+ | acc | | active_sessions | | admin_privileges | | aliases | | calls_forwarding | | config | | domain | | event | | grp | | location | | missed_calls | | pending | | phonebook | | preferences_types | | reserved | | server_monitoring | | server_monitoring_agg | | silo | | speed_dial | | subscriber | | trusted | | uri | | usr_preferences | | version | +-----------------------+ 24 rows in set (0.00 sec) ------------------------------------------------------- You can see that there is no table named "voice_silo" in my SER tables. It cause error on one of my web page named message_store.php. It requires voice_silo table. I'd like to ask you what is the different between table named "silo" and "voice_silo"? Can both of them store offline voice message? If I have to add "voice_silo" in my SER table, please give me the description of entities needed to build voice_silo table. Last, I'd like to thank you guys for the willingness to read my questions and problems. Of course I'm expecting the answers too. Thanx again. Regards, Meidiana Ten __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Fwd: [Users] mediaproxy working, but not if asterisk is involved
by Arne Van Theemsche 25 Sep '06

25 Sep '06

I had forgotten to cc the list ---------- Forwarded message ---------- From: Arne Van Theemsche <arnevt(a)gmail.com> Date: 22-sep-2006 9:06 Subject: Re: [Users] mediaproxy working, but not if asterisk is involved To: daniel(a)voice-system.ro the problem is that I don't even see the "reply received"... So for some reason the asterisk reply isn't passed through to the onreply_route. My theory is that asterisk doesn't respect the reply parameters somewhere, but it isn't clear to me where arne 2006/9/22, Daniel-Constantin Mierla <daniel(a)voice-system.ro>: > > Do you get "using mediaproxy" message in the logs? If not, that the > search() matches, I cannot sot right now what is wrong with the > expression. But you can move t_on_reply("1") into if*method=="INVITE") > statement and replace the search condition with if (status =~ > "(183)|(2[0-9][0-9])"). > > See: > http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy > > Cheers, > Daniel > > > On 09/21/06 21:46, Arne Van Theemsche wrote: > > below is the transaction of the failed mediaproxy invite. I allready > > could tell that replies go through openser, but I don't see the reason > > why ser doesn't see them as replies (and use the mediaproxy function). > > > > as you can see, the invite from <ip client> to <ip asterisk> (through > > <ip OPENSER>, which is also ip of mediaproxy) goes in one direction > > good (the ip in the SDP is changed from <ip client> to <ip openser>, > > but the return path en the OK (with it's SDP) is not changed > > > > I did a tcpdump with a call between 2 clients, where the proxy works, > > and the only difference I see is that in the reply of asterisk, there > > is no rinstance field in the contact header > > > > thanks > > arne > > > > U <ip client>:5060 -> <ip OPENSER>:5060 > > INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..From: "arne" > > <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: > > "701"< sip:701@sipgat > > e.evonet.be <http://e.evonet.be>>..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..Via: SIP/2.0/UDP <ip > > client>:5060;rport;branch=z9hG4bK-7a70a-1d > > e331c2-69dc..Max-Forwards: 70..Supported: > > replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, > > OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP1 > > 0 SP v1.0.1 (Build 3) 3.0.5.1..Allow-Events: talk, hold, > > conference..Contact: "arne" <sip:1002@<ip > > client>:5060;transport=UDP>..Session-Expires: 1800..Content- > > Type: application/sdp..Content-Length: 246....v=0..o=rtp/1 501514 > > 501514 IN IP4 <ip client>..s=-..c=IN IP4 <ip client>..t=0 0..m=audio > > 50000 RTP/AVP 18 0 8.. > > a=fmtp:18 annexb=yes..a=ptime:40..a=SilenceSupp:on..a=rtpmap:18 > > g729/8000..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv.. > > # > > > > U <ip OPENSER>:5060 -> <ip asterisk>:5060 > > INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..Record-Route: > > <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: > > "arne" < sip:1002@si > > pgate.evonet.be > > <http://pgate.evonet.be>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: > > "701"<sip:701@<sip domain>>..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> client>..C > > Seq: 1 INVITE..Via: SIP/2.0/UDP <ip OPENSER>;branch=0..Via: > > SIP/2.0/UDP <ip > > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Max-Forwards: > > 69..Supp > > orted: replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, > > NOTIFY, CANCEL, OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP10 SP > > v1.0.1 (Build 3) 3.0.5.1..Allo > > w-Events: talk, hold, conference..Contact: "arne" <sip:1002@<ip > > client>:5060;transport=UDP>..Session-Expires: 1800..Content-Type: > > application/sdp..Content-Leng > > th: 246....v=0..o=rtp/1 501514 501514 IN IP4 <ip client>..s=-..c=IN > > IP4 <ip OPENSER>..t=0 0..m=audio 60106 RTP/AVP 18 0 8..a=fmtp:18 > > annexb=yes..a=ptime:40..a > > =SilenceSupp:on..a=rtpmap:18 g729/8000..a=rtpmap:0 > > pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv.. > > # > > > > U <ip asterisk>:5060 -> <ip OPENSER>:5060 > > SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip > > OPENSER>;branch=0;received=<ip OPENSER>..Via: SIP/2.0/UDP <ip > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2- > > 69dc..From: "arne" <sip:1002@<sip > > domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip > > domain>>..Call-ID: 1064dc44-514a90c3-13c4-7a70 > > a-1de331be-529@<ip <mailto:a-1de331be-529@%3Cip> client>..CSeq: 1 > > INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, > > BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@ > > <ip asterisk>>..Content-Length: 0.... > > # > > > > U <ip OPENSER>:5060 -> <ip client>:5060 > > SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..From: > > "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c > > 4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..User-Agent: Asteri > > sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > > NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Length: 0.... > > # > > > > U <ip asterisk>:5060 -> <ip OPENSER>:5060 > > SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip OPENSER>;branch=0;received=<ip > > OPENSER>..Via: SIP/2.0/UDP <ip > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc > > ..Record-Route: <sip:<ip > > OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: "arne" > > <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f > > ..To: "701"<sip:701@<sip domain>>;tag=as60ebd3fc..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX > > ..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > > NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: > > application/sdp..Content-Length: 188....v= > > 0..o=root 26276 26276 IN IP4 <ip asterisk>..s=session..c=IN IP4 <ip > > asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 > > PCMU/8000..a=rtpmap:8 PCMA/8000..a= > > silenceSupp:off - - - -.. > > # > > > > U <ip OPENSER>:5060 -> <ip client>:5060 > > SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip > > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Record-Route: > > <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70 > > a-1de331c0-5e4f>..From: "arne" <sip:1002@<sip > > domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip > > domain>>;tag=as60ebd3fc..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, > > CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NO > > TIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: > > application/sdp..Content-Length: 188....v=0..o=root 26276 26276 IN IP4 > > <ip asterisk>..s=session..c=IN IP4 > > <ip asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 > > PCMU/8000..a=rtpmap:8 PCMA/8000..a=silenceSupp:off - - - -.. > > # > > > > > > > > > > > > 2006/9/21, Daniel-Constantin Mierla <daniel(a)voice-system.ro > > <mailto:daniel@voice-system.ro>>: > > > > Hello, > > > > watch the network traffic with ngrep on your sip server. You can > > see the > > call flow which may help to identify the issue. You can paste it > > to the > > list and someone may give you hints. > > > > Cheers, > > Daniel > > > > > > On 09/21/06 12:28, Arne Van Theemsche wrote: > > > hi > > > > > > my users subscribe with openser, en asterisk is used as > connectivity > > > to pstn > > > > > > i am now installing a mediaproxy, for all users, so every call > goes > > > via a mediaproxy. > > > > > > I'm doing this as follows (relevant statements only) > > > > > > in route > > > > > > #I installed the t_on_reply here to be sure that every > reply > > > gets parsed, but normally in the INVITE section should be enough? > > > t_on_reply("1"); > > > > > > if (method==INVITE) { > > > use_media_proxy(); > > > } > > > > > > > > > onreply_route[1] { > > > log(-3,"reply received"); > > > if (!search("^Content-Length:[ ]*0")) { > > > log(-3,"using mediaproxy"); > > > use_media_proxy(); > > > }; > > > } > > > > > > > > > the weird is, for all local users, this works fine, but as soon as > > > > asterisk is involved, the reply doesn't get triggered (not > > seeing the > > > "reply received" either, only when disconnecting the call). The > call > > > get's established fine, asterisk is sending media to the > > mediaproxy, > > > but the SDP towards the calling phone is not modified (since the > > > onreply isn't triggered) > > > > > > am I missing something here? > > > > > > thanks > > > Arne > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org <mailto:Users@openser.org> > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

1 0

[Users] How do I see who is registered?
by Ronald Wiplinger 25 Sep '06

25 Sep '06

I am watching openserctl moni to see what happens, but more than that 5 people are registered I cannot see. How do I see who is registered? bye Ronald

2 1

[serusers] Openser Admin Login Access
by John Shellhouse 24 Sep '06

24 Sep '06

Hi All, I am sure I must be missing something just not sure what it is. I granted the permission in mysql: GRANT ALL ON openser.* TO 'openser'@'dbhost' IDENTIFIED BY 'openserrw'; as stated in the Install guide and then launched the server from terminal with ./script/server and in the browser typed: http://myomain:3000 which displayed the login page. I then typed into the login:password area openser:openserrw But I was unsuccessful each time. Does anyone have any idea what I am missing or doing incorrectly? Thank you. jwshel

2 2

[serusers] Openser Admin Login Access
by John Shellhouse 23 Sep '06

23 Sep '06

Hi All, I accessed the login/password area but I am erroring but don't know how to fix. I granted all: GRANT ALL ON openser.* TO 'openser'@'dbhost' IDENTIFIED BY 'openserrw'; and changed the database.yml to dbhost as well which allowed me to login when I left it the same as localhost I was not able to login at all with default password. Now that I am in the login zone, I get this error: Mysql::Error in LoginController#login Unknown MySQL server host 'dbhost' (1) RAILS_ROOT: script/../config/.. Application Trace | Framework Trace | Full Trace /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:330:in `real_connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:330:in `connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:87:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:36:in `mysql_connection' Plus more, but I did not want to paste it all. Any help appreciated. jwshel

1 0

[Users] Trusted IP examples
by ram 23 Sep '06

23 Sep '06

Hi iam trying to test trusted IP kind of setup I have openser server connect to my ITSP another Office i have OpenSER, i want to connect to main Server and send the calls out could some one give me examples or docs where i can find similar kind of real world examples Ram

2 1

[Users] The correct hardware request
by Earley Scripts 23 Sep '06

23 Sep '06

Being very interested in SIP Routers and SIP Softphones is is an obvious step to SIP Servers etc. Having found OpenSER on the Web I have decided that I should at least get a system that I can run locally to use for testing kit and to demonstart to whomever the potential of SIP communication. I realise that I need a "server" and a Linux OS to run on I would be obliged if someone could advise the most reasonable lump of Hardware to use by giving a indication of say type of Processor and Memory requirements. I have to hand a Chassis that has a AMD K6-2 350 Mhz processor and 106 Meg of memory. I do also have a another chassis with a AMD K6-2 550 Mhz Processor and 256 Meg of Memory but that currently runs a test version of XP . I realise that Linus is required can anybody suggest which "version" of Linux has had the best success with OpenSER as I would like to avoid as much as possible the need to interact to much with the OS so I can concentrate on the Application more. In anticipation .................thanks Paul Coenraats Data Communications Consultant

2 1

[Users] which web GUI interface good for openser 1.1.x?
by Steve Xu 23 Sep '06

23 Sep '06

Hi All, I want a web GUI interface to do configuration, adding users. etc. Which addon is good for openser 1.1.x ? Thanks, Steve

3 2

RE: [Users] compiling OpenSER on Solaris 10 x86
by peter.3.edwards＠bt.com 23 Sep '06

23 Sep '06

> I'm having a lot of difficulty compiling OpenSER on Solaris > 10. Firstly > I was missing a lot of packages which I needed to download from > sunfreeware.com (gcc, ld) the /usr/bin/ccs version of make > does not work > on the makefile > so I'm kindda stuck, it's apparently been compiled on the x68 s10 > platform before so there must be some sort of makefile modifications I > must make everywhere ? > > Anyone know what may be going on? I was hoping there would be > a package > OpenSER version for solaris built somewhere :S > > Thanks! > Nick Hi, Nick. All the pre-requisites to build the default OpenSER 1.1.0 under Solaris are available on the Sun Software Companion CD which is supplied with Solaris. These packages are also available for download from Sun's website: http://www.sun.com/software/solaris/freeware/ Off the top of my head, I think I installed these packages from the Solaris 9 Software Companion CD: Application/Editors - GNU sed v3.02 Application/Utilities - GNU text file processing utilities v2.0 Development/Languages - gcc-3.3.2 - GNU bison v1.35 - GNU flex v2.54 Development/Tools - GNU make utility v3.80 Although there may be a few others I've forgotten or already had installed, for use with other software. The only pieces of non-Sun software I had to install were OpenSSL 0.7 and MySQL's libmysqlclient (which requires you to compile MySQL from scratch - as it's not included with the Solaris pkg supplied by MySQL), both of which are optional when it comes to compiling OpenSER. Everything should work OK, if you're using the requirements as specified on the install page though you may also want to look at a trio of bugs I raised in OpenSER's Tracker system regarding some Solaris-compability issues with a couple of OpenSER's shell scripts, in case they haven't been fixed in the release you're using: [1544112] Solaris openser.init script error [1540132] Solaris openserctl.base script error [1540127] Solaris openserctl.fifo script error I believe the last two at least have been fixed in CVS, but they might not have made it into the latest released package yet. HTH, Peter. -- This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

1 0

[Serusers] forward service with avpops
by Rosario Pingaro 23 Sep '06

23 Sep '06

I am starting to implement the forward capability using avpops module. Before to start I would like to know if it is possible to bill the forwarded call directly to the forwader (the destination of the first call). We account our call using radius, Wiil be possible to see the forwarded call like originated from the forwarder from accounting point of view? If I am wrong what is the good method to bill the forwarded call to the forwarder? Regards Rosario

1 0

[Users] ngrep explanation please
by Ronald Wiplinger 23 Sep '06

23 Sep '06

I have a softphone (eyebeam at 7211) and I try to call a hard phone at 7112. Eyebeam shows after the phone rang: Call failed: Media negotiation failed. I have set-up both for voice: G711u and video: h264. Can you please brief explain me what I see with: ngrep -d any -qt port 5060 (output is below)? (brief) and where do I get a detail information for it? and what could be the problem, what could be a possible solution? thanks for your help! bye Ronald U 2006/09/18 09:52:11.253462 192.168.250.108:43854 -> 192.168.250.41:5060 INVITE sip:7112@192.168.250.41 SIP/2.0..Via: SIP/2.0/UDP 192.168.250.108:43854;bran ch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543-;rport..Max-Forwards: 70..Contact: <si p:7211@192.168.250.108:43854>..To: "7112"<sip:7112@192.168.250.41>..From: "Ronald-t aipei @ video "<sip:7211@192.168.250.41>;tag=85349c3f..Call-ID: Zjk4YjVjZDczYzQyNmN kMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agen t: eyeBeam release 1004p stamp 31962..Content-Length: 785....v=0..o=- 2 2 IN IP4 19 2.168.250.108..s=CounterPath eyeBeam 1.5..c=IN IP4 192.168.250.108..t=0 0..m=audio 11588 RTP/AVP 107 100 106 6 0 105 18 3 5 101..a=alt:1 2 : 1HNRF2W7 lFCxbz3O 5.32.86 .166 11588..a=alt:2 1 : wmC2UUv5 SDteeaPr 192.168.250.108 11588..a=fmtp:18 annexb=n o..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap:105 SPEEX-FEC/8000..a=rtpmap:101 telephone-event/8000..a =sendrecv..a=x-rtp-session-id:D08BC9B4AF734681A0D89918C35B13C3..m=video 62096 RTP/A VP 125..a=alt:1 2 : JXP4adeI lQwRBdzZ 5.32.86.166 62096..a=alt:2 1 : hh2NmIOI MdCTT UEJ 192.168.250.108 62096..a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbp s=19800..a=rtpmap:125 H264/90000..a=sendrecv..a=x-rtp-session-id:8178D81F66DF4316AA D51ABF209E1157.. U 2006/09/18 09:52:11.271928 192.168.250.41:5060 -> 192.168.250.108:43854 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.250.10 8:43854;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543-;rport=43854..To: "7112"<s ip:7112@192.168.250.41>..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41>;ta g=85349c3f..Call-ID: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE.. Server: OpenSer (1.1.0-notls (x86_64/linux))..Content-Length: 0..Warning: 392 192.1 68.250.41:5060 "Noisy feedback tells: pid=31767 req_src_ip=192.168.250.108 req_src _port=43854 in_uri=sip:7112@192.168.250.41 out_uri=sip:7112@192.168.250.116:5060 vi a_cnt==1".... U 2006/09/18 09:52:11.271969 192.168.250.41:5060 -> 192.168.250.116:5060 INVITE sip:7112@192.168.250.116:5060 SIP/2.0..Record-Route: <sip:192.168.250.41;lr= on;ftag=85349c3f>..Via: SIP/2.0/UDP 192.168.250.41;branch=z9hG4bK2a6b.b6ab15e2.0..V ia: SIP/2.0/UDP 192.168.250.108:43854;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87 543-;rport=43854..Max-Forwards: 69..Contact: <sip:7211@192.168.250.108:43854>..To: "7112"<sip:7112@192.168.250.41>..From: "Ronald-taipei @ video "<sip:7211@192.168.25 0.41>;tag=85349c3f..Call-ID: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE , INFO..Content-Type: application/sdp..User-Agent: eyeBeam release 1004p stamp 3196 2..Content-Length: 785..P-hint: usrloc applied....v=0..o=- 2 2 IN IP4 192.168.250.1 08..s=CounterPath eyeBeam 1.5..c=IN IP4 192.168.250.108..t=0 0..m=audio 11588 RTP/A VP 107 100 106 6 0 105 18 3 5 101..a=alt:1 2 : 1HNRF2W7 lFCxbz3O 5.32.86.166 11588. .a=alt:2 1 : wmC2UUv5 SDteeaPr 192.168.250.108 11588..a=fmtp:18 annexb=no..a=fmtp:1 01 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/ 16000..a=rtpmap:105 SPEEX-FEC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. a=x-rtp-session-id:D08BC9B4AF734681A0D89918C35B13C3..m=video 62096 RTP/AVP 125..a=a lt:1 2 : JXP4adeI lQwRBdzZ 5.32.86.166 62096..a=alt:2 1 : hh2NmIOI MdCTTUEJ 192.168 .250.108 62096..a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800..a= rtpmap:125 H264/90000..a=sendrecv..a=x-rtp-session-id:8178D81 U 2006/09/18 09:52:11.281432 192.168.250.116:5060 -> 192.168.250.41:5060 SIP/2.0 100 Trying..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=85 349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-ID : Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 19 2.168.250.41 ;received=192.168.250.41 ;branch=z9hG4bK2a6b.b6ab15e2.0..Via: SIP/2.0/ UDP 192.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rport =43854..Contact: <sip:7112@192.168.250.116:5060>..Record-Route: <sip:192.168.250.41 ;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:11.785445 192.168.250.116:5060 -> 192.168.250.41:5060 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.41 ;received=192.168.250.41 ;branch=z9hG4bK2a6b.b6ab15e2.0..Via: SIP/2.0 /UDP 192.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rpor t=43854..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Reco rd-Route: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:11.785662 192.168.250.41:5060 -> 192.168.250.108:43854 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rport=4385 4..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Record-Rou te: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:12.287448 192.168.250.116:5060 -> 192.168.250.41:5060 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.41 ;received=192.168.250.41 ;branch=z9hG4bK2a6b.b6ab15e2.0..Via: SIP/2.0 /UDP 192.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rpor t=43854..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Reco rd-Route: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:12.287785 192.168.250.41:5060 -> 192.168.250.108:43854 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rport=4385 4..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Record-Rou te: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:13.287502 192.168.250.116:5060 -> 192.168.250.41:5060 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.41 ;received=192.168.250.41 ;branch=z9hG4bK2a6b.b6ab15e2.0..Via: SIP/2.0 /UDP 192.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rpor t=43854..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Reco rd-Route: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:13.287742 192.168.250.41:5060 -> 192.168.250.108:43854 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rport=4385 4..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Record-Rou te: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:15.287590 192.168.250.116:5060 -> 192.168.250.41:5060 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.41 ;received=192.168.250.41 ;branch=z9hG4bK2a6b.b6ab15e2.0..Via: SIP/2.0 /UDP 192.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rpor t=43854..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Reco rd-Route: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:15.287803 192.168.250.41:5060 -> 192.168.250.108:43854 SIP/2.0 180 Ringing..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=8 5349c3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-I D: Zjk4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 1 92.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rport=4385 4..Require: 100rel..RSeq: 441..Contact: <sip:7112@192.168.250.116:5060>..Record-Rou te: <sip:192.168.250.41;ftag=85349c3f>..Content-Length:0.... U 2006/09/18 09:52:17.801707 192.168.250.116:5060 -> 192.168.250.41:5060 SIP/2.0 200 OK..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=85349c 3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-ID: Zj k4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.16 8.250.41 ;received=192.168.250.41 ;branch=z9hG4bK2a6b.b6ab15e2.0..Via: SIP/2.0/UDP 192.168.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rport=438 54..Contact: <sip:7112@192.168.250.116:5060>..Record-Route: <sip:192.168.250.41;fta g=85349c3f>..Content-Type: application/SDP..Content-Length:95....v=0..o=wintech 506 1 5071 IN IP4 192.168.250.116..s=ssss-name..c=IN IP4 192.168.250.116..t=0 0.. U 2006/09/18 09:52:17.801935 192.168.250.41:5060 -> 192.168.250.108:43854 SIP/2.0 200 OK..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=85349c 3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-ID: Zj k4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.16 8.250.108:43854 ;branch=z9hG4bK-d87543-aa3e232283193c6d-1--d87543- ;rport=43854..Co ntact: <sip:7112@192.168.250.116:5060>..Record-Route: <sip:192.168.250.41;ftag=8534 9c3f>..Content-Type: application/SDP..Content-Length:95....v=0..o=wintech 5061 5071 IN IP4 192.168.250.116..s=ssss-name..c=IN IP4 192.168.250.116..t=0 0.. U 2006/09/18 09:52:18.006466 192.168.250.108:43854 -> 192.168.250.41:5060 ACK sip:192.168.250.41;ftag=85349c3f SIP/2.0..Via: SIP/2.0/UDP 192.168.250.108:4385 4;branch=z9hG4bK-d87543-fc36ee32f92f8656-1--d87543-;rport..Max-Forwards: 70..Route: <sip:7112@192.168.250.116:5060>..Contact: <sip:7211@192.168.250.108:43854>..To: "7 112"<sip:7112@192.168.250.41>;tag=0-13c4-9122d-236f034a-80b..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41>;tag=85349c3f..Call-ID: Zjk4YjVjZDczYzQyNmNkMzgxOD c3MDVmNjJlOWIyYTc...CSeq: 1 ACK..User-Agent: eyeBeam release 1004p stamp 31962..Con tent-Length: 0.... U 2006/09/18 09:52:18.006478 192.168.250.108:43854 -> 192.168.250.41:5060 BYE sip:192.168.250.41;ftag=85349c3f SIP/2.0..Via: SIP/2.0/UDP 192.168.250.108:4385 4;branch=z9hG4bK-d87543-d52beb5db212b556-1--d87543-;rport..Max-Forwards: 70..Route: <sip:7112@192.168.250.116:5060>..Contact: <sip:7211@192.168.250.108:43854>..To: "7 112"<sip:7112@192.168.250.41>;tag=0-13c4-9122d-236f034a-80b..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41>;tag=85349c3f..Call-ID: Zjk4YjVjZDczYzQyNmNkMzgxOD c3MDVmNjJlOWIyYTc...CSeq: 2 BYE..User-Agent: eyeBeam release 1004p stamp 31962..Rea son: SIP;description="Application Rejected Sdp(usually no common codec)"..Content-L ength: 0.... U 2006/09/18 09:52:18.006887 192.168.250.41:5060 -> 192.168.250.116:5060 ACK sip:7112@192.168.250.116:5060 SIP/2.0..Record-Route: <sip:192.168.250.41;lr=on; ftag=85349c3f>..Via: SIP/2.0/UDP 192.168.250.41;branch=z9hG4bK2a6b.b6ab15e2.2..Via: SIP/2.0/UDP 192.168.250.108:43854;branch=z9hG4bK-d87543-fc36ee32f92f8656-1--d87543 -;rport=43854..Max-Forwards: 69..Contact: <sip:7211@192.168.250.108:43854>..To: "71 12"<sip:7112@192.168.250.41>;tag=0-13c4-9122d-236f034a-80b..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41>;tag=85349c3f..Call-ID: Zjk4YjVjZDczYzQyNmNkMzgxODc 3MDVmNjJlOWIyYTc...CSeq: 1 ACK..User-Agent: eyeBeam release 1004p stamp 31962..Cont ent-Length: 0..P-hint: rr-enforced.... U 2006/09/18 09:52:18.007147 192.168.250.41:5060 -> 192.168.250.116:5060 BYE sip:7112@192.168.250.116:5060 SIP/2.0..Record-Route: <sip:192.168.250.41;lr=on; ftag=85349c3f>..Via: SIP/2.0/UDP 192.168.250.41;branch=z9hG4bKf96b.03d4bda5.0..Via: SIP/2.0/UDP 192.168.250.108:43854;branch=z9hG4bK-d87543-d52beb5db212b556-1--d87543 -;rport=43854..Max-Forwards: 69..Contact: <sip:7211@192.168.250.108:43854>..To: "71 12"<sip:7112@192.168.250.41>;tag=0-13c4-9122d-236f034a-80b..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41>;tag=85349c3f..Call-ID: Zjk4YjVjZDczYzQyNmNkMzgxODc 3MDVmNjJlOWIyYTc...CSeq: 2 BYE..User-Agent: eyeBeam release 1004p stamp 31962..Reas on: SIP;description="Application Rejected Sdp(usually no common codec)"..Content-Le ngth: 0..P-hint: rr-enforced.... U 2006/09/18 09:52:18.039998 192.168.250.116:5060 -> 192.168.250.41:5060 SIP/2.0 200 OK..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=85349c 3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-ID: Zj k4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 2 BYE..Via: SIP/2.0/UDP 192.168.2 50.41 ;received=192.168.250.41 ;branch=z9hG4bKf96b.03d4bda5.0..Via: SIP/2.0/UDP 192 .168.250.108:43854 ;branch=z9hG4bK-d87543-d52beb5db212b556-1--d87543- ;rport=43854. .Content-Length:0.... U 2006/09/18 09:52:18.040200 192.168.250.41:5060 -> 192.168.250.108:43854 SIP/2.0 200 OK..From: "Ronald-taipei @ video "<sip:7211@192.168.250.41> ;tag=85349c 3f..To: "7112"<sip:7112@192.168.250.41> ;tag=0-13c4-9122d-236f034a-80b..Call-ID: Zj k4YjVjZDczYzQyNmNkMzgxODc3MDVmNjJlOWIyYTc...CSeq: 2 BYE..Via: SIP/2.0/UDP 192.168.2 50.108:43854 ;branch=z9hG4bK-d87543-d52beb5db212b556-1--d87543- ;rport=43854..Conte nt-Length:0.... U 2006/09/18 09:52:20.758052 192.168.250.108:43854 -> 192.168.250.41:5060 .... U 2006/09/18 09:52:30.491226 192.168.250.116:5060 -> 192.168.250.41:5060 REGISTER sip:192.168.250.41:5060 SIP/2.0..From: <sip:7112@testdomain.com> ;tag=0-13 c4-91241-236f4e52-101..To: <sip:7112@testdomain.com>..Call-ID: 0-13c4-4-1316-362a.. CSeq: 3106 REGISTER..Via: SIP/2.0/UDP 192.168.250.116:5060 ;branch=z9hG4bK-91241-23 6f4e52-6ba6..Max-Forwards: 70..Contact: <sip:7112@192.168.250.116:5060>;expires=720 0..Content-Length:0.... U 2006/09/18 09:52:30.491441 192.168.250.41:5060 -> 192.168.250.116:5060 SIP/2.0 200 OK..From: <sip:7112@testdomain.com> ;tag=0-13c4-91241-236f4e52-101..To: <sip:7112@testdomain.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.d9ec..Call-ID: 0-13 c4-4-1316-362a..CSeq: 3106 REGISTER..Via: SIP/2.0/UDP 192.168.250.116:5060 ;branch= z9hG4bK-91241-236f4e52-6ba6..Contact: <sip:7112@192.168.250.116:5060>;expires=7200. .Server: OpenSer (1.1.0-notls (x86_64/linux))..Content-Length: 0..Warning: 392 192. 168.250.41:5060 "Noisy feedback tells: pid=31768 req_src_ip=192.168.250.116 req_sr c_port=5060 in_uri=sip:192.168.250.41:5060 out_uri=sip:192.168.250.41:5060 via_cnt= =1"....

4 3

[Users] Problem in TLS testing with minisip. Please help..Please...
by Ferianto siregar 23 Sep '06

23 Sep '06

Dear all, Thank you very much for this chance. Thank you. I am sorry, I have some questions. Can I ask you, Please? 1. All, I have build openser server with TLS support. I use openser-1.1.0-tls Now, I tried to test it by making a call by using softphone that supported TLS. Based on the suggestion from this forum, for testing it, I can use minisip. I have configured minisip like the installation note that I got when download it. But, I have a problem. I have error when running minisip and make a call in Windows XP. This is the error message : Starting MiniSIP ... welcome! libmutil: setupDefaultSignalHandling: No stack trace signal handler available Initializing NetUtil Creating SipSoftPhoneConfiguration init 1/9: Creating timeout provider init 2/9: Creating GUI Creating GTK GUI Setting contact db Thread 2 running - doing initParseConfig init 3/9: Parsing configuration file () WARNING: Could not determine home directory Config file version checked ok! SipIdentity::SipIdentity : cretated identity id=1 SipIdentity::setSipUri: sipUsername=<aldi> sipDomain=<pcr.ac.id> SipIdentity::setSipProxy: autodetect is false; userUri=aldi(a)pcr.ac.id; transport = UDP; proxyAddr=sip.pcr.ac.id; proxyPort=5060 SipProxy:setProxy(str) : addr = sip.pcr.ac.id SipIdentity::setProxy: manual sipproxy success ... SipIdentity::setProxy: else ... Identities: identity=1; username=aldi; domain=pcr.ac.id proxy=[proxyString=sip.pcr.a c.id; proxyString=sip.pcr.ac.id; port=5060; transport=UDP; autodetect=no; user=a ldi; password=aldi; expires=1000]; isRegistered=0 init 4/9: Creating IP provider Adapter Name: {DAF8C45E-7B2A-4D08-98AA-5E5091C3464D} Adapter Name: {6FFFD1E7-4BC6-426E-929D-E8292B7FC0DB} Adapter Name: {B926ACA5-8120-42A4-BAD7-AE95AA6708F0} SimpleIPProvider: localIp = SimpleIPProvider: preferred network interface = {DAF8C45E-7B2A-4D08-98AA-5E5091C 3464D} SimpleIPProvider: preferred interface found Minisip is using IP = 202.95.149.5 init 5/9: Creating MediaHandler Sound I/O: using Spatial Audio Mixer Adding audio codec: G.711 init 6/9: Creating MSip SIP stack init 7/9: Connecting GUI to SIP logic init 8.2/9: Starting TCP transport worker thread init 9/9: Registering Identities to registrar server Registering user aldi(a)pcr.ac.id to proxy sip.pcr.ac.id, requesting domain pcr.ac .id IP4Address(string): sip.pcr.ac.id (202.95.149.251) IP4Address(string): sip.pcr.ac.id (202.95.149.251) IP4Address(string): sip.pcr.ac.id (202.95.149.251) Before new mediaSession RtpReceiver:: final trying port = 34694 After new mediaSession Before addDialog After addDialog Before handleCommand After handleCommand Session::getSdpOffer: v=0 o=- 3344 3344 IN IP4 202.95.149.5 s=Minisip Session c=IN IP4 202.95.149.5 t=0 0 m=audio 34694 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 IP4Address(string): sip.pcr.ac.id (202.95.149.251) ERROR:(in IP6Address) Unknown host: <pcr.ac.id> What`s wrong? I have tried to analyze it, but I still do not understand about the error message. I do hope anybody can give me a suggestion. Please help me..Please. 2. In order minisip can use TLS, I have loaded a certificate from "certificate settings" menu. What kind of certificate that I have to load? Where I can load the certificate? Does it means the certificate that I use in openser.cfg? 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" I mean, Do I have to copy user-cert.pem, user-privkey.pem, user-calist.pem to Client computer (using Windows XP), and load it by using "certificate setting" menu in minisip? Maybe my question is too much. I am sorry for that. I do hope anybody can give me a suggestion. Please help...this newbie one..Please... Thank you very much, Regards, Ferianto Note : Here is my openser.cfg file: debug=4 fork=no log_stderror=yes listen=200.95.149.251 # put your openserver IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" fifo_mode=0666 alias="pcr.ac.id" tls_port_no=5061 # uncomment the following lines for TLS support disable_tls = 0 listen = tls:200.95.149.251:5061 tls_verify_client = on tls_require_client_certificate = on tls_verify_server=on 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" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "s:inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); return; }; if (uri==myself) { if (method=="ACK") { route(6); return; } else if (method=="CANCEL") { route(3); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "Uopenser Not Found"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } #if (!allow_trusted() && nat_uac_test("19")) { # setflag(6); #} lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); return; }; if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); return; }; sl_send_reply("404", "Uopenser Not Found"); return; }; if (method=="CANCEL") { route(1); return; } if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("200.95.149.254:5060"); # INOPENSERT YOUR PSTN GATEWAY IP ADDRESS # avp_write("i:45", "inv_timeout"); #if (!allow_trusted() && isflagset(6)) { if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

1 0

[Users] Problem in TLS testing with minisip. Please help..Please...
by Ferianto siregar 23 Sep '06

23 Sep '06

Dear all, Thank you very much for this chance. Thank you. I am sorry, I have some questions. Can I ask you, Please? 1. All, I have build openser server with TLS support. I use openser-1.1.0-tls Now, I tried to test it by making a call by using softphone that supported TLS. Based on the suggestion from this forum, for testing it, I can use minisip. I have configured minisip like the installation note that I got when download it. But, I have a problem. I have error when running minisip and make a call in Windows XP. This is the error message : Starting MiniSIP ... welcome! libmutil: setupDefaultSignalHandling: No stack trace signal handler available Initializing NetUtil Creating SipSoftPhoneConfiguration init 1/9: Creating timeout provider init 2/9: Creating GUI Creating GTK GUI Setting contact db Thread 2 running - doing initParseConfig init 3/9: Parsing configuration file () WARNING: Could not determine home directory Config file version checked ok! SipIdentity::SipIdentity : cretated identity id=1 SipIdentity::setSipUri: sipUsername=<aldi> sipDomain=<pcr.ac.id> SipIdentity::setSipProxy: autodetect is false; userUri=aldi(a)pcr.ac.id; transport = UDP; proxyAddr=sip.pcr.ac.id; proxyPort=5060 SipProxy:setProxy(str) : addr = sip.pcr.ac.id SipIdentity::setProxy: manual sipproxy success ... SipIdentity::setProxy: else ... Identities: identity=1; username=aldi; domain=pcr.ac.id proxy=[proxyString=sip.pcr.a c.id; proxyString=sip.pcr.ac.id; port=5060; transport=UDP; autodetect=no; user=a ldi; password=aldi; expires=1000]; isRegistered=0 init 4/9: Creating IP provider Adapter Name: {DAF8C45E-7B2A-4D08-98AA-5E5091C3464D} Adapter Name: {6FFFD1E7-4BC6-426E-929D-E8292B7FC0DB} Adapter Name: {B926ACA5-8120-42A4-BAD7-AE95AA6708F0} SimpleIPProvider: localIp = SimpleIPProvider: preferred network interface = {DAF8C45E-7B2A-4D08-98AA-5E5091C 3464D} SimpleIPProvider: preferred interface found Minisip is using IP = 202.95.149.5 init 5/9: Creating MediaHandler Sound I/O: using Spatial Audio Mixer Adding audio codec: G.711 init 6/9: Creating MSip SIP stack init 7/9: Connecting GUI to SIP logic init 8.2/9: Starting TCP transport worker thread init 9/9: Registering Identities to registrar server Registering user aldi(a)pcr.ac.id to proxy sip.pcr.ac.id, requesting domain pcr.ac .id IP4Address(string): sip.pcr.ac.id (202.95.149.251) IP4Address(string): sip.pcr.ac.id (202.95.149.251) IP4Address(string): sip.pcr.ac.id (202.95.149.251) Before new mediaSession RtpReceiver:: final trying port = 34694 After new mediaSession Before addDialog After addDialog Before handleCommand After handleCommand Session::getSdpOffer: v=0 o=- 3344 3344 IN IP4 202.95.149.5 s=Minisip Session c=IN IP4 202.95.149.5 t=0 0 m=audio 34694 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 IP4Address(string): sip.pcr.ac.id (202.95.149.251) ERROR:(in IP6Address) Unknown host: <pcr.ac.id> What`s wrong? I have tried to analyze it, but I still do not understand about the error message. I do hope anybody can give me a suggestion. Please help me..Please. 2. In order minisip can use TLS, I have loaded a certificate from "certificate settings" menu. What kind of certificate that I have to load? Where I can load the certificate? Does it means the certificate that I use in openser.cfg? 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" I mean, Do I have to copy user-cert.pem, user-privkey.pem, user-calist.pem to Client computer (using Windows XP), and load it by using "certificate setting" menu in minisip? Maybe my question is too much. I am sorry for that. I do hope anybody can give me a suggestion. Please help...this newbie one..Please... Thank you very much, Regards, Ferianto Note : Here is my openser.cfg file: debug=4 fork=no log_stderror=yes listen=200.95.149.251 # put your openserver IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" fifo_mode=0666 alias="pcr.ac.id" tls_port_no=5061 # uncomment the following lines for TLS support disable_tls = 0 listen = tls:200.95.149.251:5061 tls_verify_client = on tls_require_client_certificate = on tls_verify_server=on 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" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "s:inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); return; }; if (uri==myself) { if (method=="ACK") { route(6); return; } else if (method=="CANCEL") { route(3); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "Uopenser Not Found"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } #if (!allow_trusted() && nat_uac_test("19")) { # setflag(6); #} lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); return; }; if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); return; }; sl_send_reply("404", "Uopenser Not Found"); return; }; if (method=="CANCEL") { route(1); return; } if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("200.95.149.254:5060"); # INOPENSERT YOUR PSTN GATEWAY IP ADDRESS # avp_write("i:45", "inv_timeout"); #if (!allow_trusted() && isflagset(6)) { if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. --------------------------------- All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.

1 0

[Users] how to change password of user openser in openser1.1.0
by Steve Xu 23 Sep '06

23 Sep '06

Hi Folks, I am need to openser and download the openser1.1.0 to try on. How to change the default password 'openserrw' for user 'openser' when openser 1.10 connected to MySQL 5 plus? /usr/local/sbin/openser_mysql.sh has the default password 'openserrw' however openselctl will give me error "cannot connecte to database server" if I changed the default password. Is it hard coded? Where should I change it after the default installation. Any help will be appreciated. Steve

2 1

[Serusers] Documentation of What's New in 0.10
by Greger V. Teigre 23 Sep '06

23 Sep '06

Hi guys! We have started a documentation effort for 0.10 to document what has been changed and what is now in the upcoming release. http://www.iptel.org/ser/doc/010whatsnew There is already quite some content there, but it is by no means finished. If you have experiences with 0.10 (current CVS head), please feel free to add your experiences where appropriate (if you create a new page, use Book as type and find What's new in the Parent dialog). All: Add comments if there are some things you would like to have documented. Regardless of which version of SER you are using or planning to use, I believe this document will be of interest to you: http://www.iptel.org/ser/howtos/optimizing_the_use_of_rtp_proxy g-)

4 4

[Serusers] RE: Back to Back INVITE?
by xwang＠cascotec.com 23 Sep '06

23 Sep '06

Hello I have SER and RTPPROXY installed. Everything works fine till we debug a softphone. I use SPA3000 to dial, the receiver is a softphone. The softphone registers correctly. When dialing from SPA, here is the handshaking sequence SPA3000 -- INVITE --><-- TRYING SER -- INVITE -- INVITE --><--TRYING RING softphone. There is a back-to-back INVITE sent out immediately by SER to the softphone. I don't see those sequence to call other UAs. Does anyone know why SER sends two INVITE? Thanks! Steven

2 1

[Users] ACC module - important changes
by Bogdan-Andrei Iancu 22 Sep '06

22 Sep '06

Hi everybody, I just commited on the CVS a grand ACC restructuring and clean-up. Here is form commit log: - simplify internal logic for preparing and message selection - more efficient data collection - eliminate the FMT-based accounting; instead of it, a default set of basic values are all the time accounted; for more, extra accounting should be used -> this leads to a more selective / flexible / performant accounting - pre-builds sets of names / columns / RADIUS AVPs / DIAMETER AVPs instead of doing it each time a request is accounted - "n/a" string for not found/available values replaced with empty string "" - eliminated a lot of un-necessary functions - heritage from the all TM callback design (pre pre-transaction transaction) - DB accounting table redesigned - accounting logic separated in a new file for a better modularity. - implemented an accounting environment for storing data; helps in reducing the number of function's parameters and of internal functions; more important, useless conversions from str to int and to int again are eliminated The most important change was the drop of FMTs for a more flexible system: have a basic set of values used of call matching, plus the usage of extra accounting for anything more. The basic set is defined to : method name From URI From TAG To URI To TAG CallID Cseqno Reply code Reply reason Time stamp having a timestamp generated by the proxy instead by the backend (syslog, RADIUS server, SQL server) is important as it is more accurate - the delays due transport to the backend server are eliminated. Also the time is logged in UNIX timestamp format to avoid confusions about timezones. From how to use it (enabling acc via flags), nothing changed :)..only some module parameters were lost. For more details, please see: http://openser.org/docs/modules/1.2.x/acc.html any feedback is welcomed regards, bogdan

1 0

Re: [Users] Checking for Digit Length
by Mark Kent 22 Sep '06

22 Sep '06

> Basically I am trying to find out a way of checking for minimum and > maximum digit length dialled by our users, in order to filter some > of the traffic through our OpenSER server. Dialling rules aside, is > there a way? Unless I missed it, no one told the guy what he needs to know. Instead, he is pointed at a lengthy specification for posix regular expressions. I know there is value in teaching a guy to fish, but at some point someone has to just answer the question... This if (uri=~"sip:[0-9]{10,24}") { ... } will match a sequence of digits between 10 and 24 in length (inclusive). Now, the above is not a great test in practice. I don't know how long the longest number can be outside the North American Numbering plan (where it is relatively easy: 1-NXX-NXX-XXXX, see nanpa.com). And there are restrictions on the first few digits that you might want to check, depending on where you are in the world. I'm sure that someone reading this list has already got the ultimate, global, number checker coded for openser... it's just a question of whether they'll cough it up in public. -mark

2 1

[Serusers] Active Session Count in RTPProxy
by Gaurav Kulshreshtha 22 Sep '06

22 Sep '06

Is it possible in someway to extract the actual number of active sessions in RTP Proxy. There is sessions_created variable but its a cumulative count of sessions created. Gaurav

1 0

[Users] compiling OpenSER on Solaris 10 x86
by Nick De Cristofaro 22 Sep '06

22 Sep '06

I'm having a lot of difficulty compiling OpenSER on Solaris 10. Firstly I was missing a lot of packages which I needed to download from sunfreeware.com (gcc, ld) the /usr/bin/ccs version of make does not work on the makefile here is the error: bash-3.00# make make: Fatal error in reader: Makefile.sources, line 18: Unexpected end of line seen I looked in the file but I don't see how it would complaing of some unexpected end of line. I then followed to try GNU's make I downloaded. and then I get an error on the /usr/bin/ccs/lex and yacc's (it mentioned to use yacc and not bison) the error was: bash-3.00# /usr/local/bin/make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce lex cfg.lex "cfg.lex":line 102: Error: missing translation value make: *** [lex.yy.c] Error 1 Then I decided to try flex instead of lex. bash-3.00# /usr/local/bin/make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce flex cfg.lex flex: fatal internal error, exec failed make: *** [lex.yy.c] Broken Pipe make: *** Deleting file `lex.yy.c' so I'm kindda stuck, it's apparently been compiled on the x68 s10 platform before so there must be some sort of makefile modifications I must make everywhere ? Anyone know what may be going on? I was hoping there would be a package OpenSER version for solaris built somewhere :S Thanks! Nick

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[Users] Local authentication
by Artur Hayne 22 Sep '06

22 Sep '06

Hello, Can I authenticate the openser locally? I want the user that do local login (in the computer of the server) can be authenticate in openser server. Its possible? Thanks. --------------------------------- Yahoo! Search Música para ver e ouvir: You're Beautiful, do James Blunt

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[Users] rtpproxy bridge mode with multiple handsets per extension
by David Carter 22 Sep '06

22 Sep '06

Good afternoon, I have a small openser test rig with rtpproxy running as a bridge between two networks: rtpproxy -l <public IP>/<private IP> -s /tmp/rtpproxy.sock This works well apart from one issue. I have set up two handsets for an extension, one on the public network and one on the private network. The idea is to provide one phone in the office and one at home, so I can pick up at either location when someone calls. If I call the extension both handsets ring correctly, but audio is only set up correctly for one of the two handsets when I pick up. Everything works fine if both phones are on the public or the private network. I infer that this is because primitives working on the ruri and friends: avp_write("$ruri/domain", "i:20"); only work on the address of the first registered handset that they find, even if lookup("location") has created multiple branches. Does anyone have any ideas or workarounds which would allow me to set up multiple handsets for a given extension when bridge mode is involved? I tried abusing the aliases table to redirect an incoming call onto multiple SIP URIs. It looks like rtpproxy indexes by Call-ID, and hence struggles with multiple branches which share a single Call-ID. Thanks, -- David Carter Email: David.Carter(a)ucs.cam.ac.uk University Computing Service, Phone: (01223) 334502 New Museums Site, Pembroke Street, Fax: (01223) 334679 Cambridge UK. CB2 3QH.

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[Serusers] Re: Re: Using IP address as domain
by mukesh 22 Sep '06

22 Sep '06

Hi andrey, I have made suggested changes in ser.cfg as follows listen=202.87.x.x alias=202.87.x.x and did #export SIP_DOMAIN=202.87.x.x I get following log when try to start SER #/etc/rc.id/init.d/ser start ------------log starts here-------------------- Starting ser: 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 Listening on udp: 202.87.x.x [202.87.x.x]:5060 tcp: 202.87.x.x [202.87.x.x]:5060 Aliases: *: 202.87.x.x:* 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) ERROR: udp_init: bind(6, 0x810237c, 16) on 202.87.x.x: Cannot assign requested address [FAILED] ------------log ends here---------------------- please advice. Regards Mukesh From: "Andrey Kuprianov" <andrey.kouprianov(a)gmail.com> Subject: Re: [Serusers] Using IP address as domain To: serusers(a)iptel.org Message-ID: <ab5f5bda0609210935y71edb46ci6ea9966b170e52de(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, AFAIK, SIP_DOMAIN you need to export anyway.. Just put your IP instead of the name. Then this SIP_DOMAIN will be used to properly add contacts to your subscriber table. So, if your domain is an IP address, newly added ppl will have an IP instead of domain for their host part of URI. Try also to specify alias as you IP address, i.e. listen="202.x.x.x" alias="202.x.x.x" You may also try to add a domain entry to the domain table with serctl (that is if you use functions in your config such as "is_uri_host_local()"), i.e. serctl domain add 202.x.x.x Hope it will work for you. Andrey. On 9/20/06, mukesh <mukesh(a)hungama.com> wrote: > Hi friends, > I have SER (with database support) running on my linux box. But I do > not have a domain name registered for the IP of this machine but it does > have a public IP (202.87.xx.xxx) address which allows me access it from > outside. > > If I try to access (using sipsak client) a uri from local system it > fails. > example URIs I am trying are ... > sip:mukesh@<public ip> > sip:mukesh@<public ip>:5060 > > ...with no result :( > > I referred http://mit.edu/sip/sip.edu/ser.shtml to configure my > ser.cfg. > > So the question is How can I connect to SER using IP and do I need > to use "# export SIP_DOMAIN=<public ip>"? > > Regards > Mukesh

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[Serusers] Re: Re: Using IP address as domain
by mukesh 22 Sep '06

22 Sep '06

Hi andrey, I have made follwing changes in ser.cfg listen="202.87.x.x" alias="202.87.x.x" #export SIP_DOMAIN = 202.87.x.x when I run # /etc/rc.d/init.d/ser restart I get this log ----------------------log starts here ------------------------- Starting ser: 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 Listening on udp: 202.87.x.x [202.87.x.x]:5060 tcp: 202.87.x.x [202.87.x.x]:5060 Aliases: *: 202.87.x.x:* 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) ERROR: udp_init: bind(6, 0x810237c, 16) on 202.87.x.x: Cannot assign requested address [FAILED] [root@oradb1 /]# ----------------------log ends here ------------------------- Regards Mukesh From: "Andrey Kuprianov" <andrey.kouprianov(a)gmail.com> Subject: Re: [Serusers] Using IP address as domain To: serusers(a)iptel.org Message-ID: <ab5f5bda0609210935y71edb46ci6ea9966b170e52de(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, AFAIK, SIP_DOMAIN you need to export anyway.. Just put your IP instead of the name. Then this SIP_DOMAIN will be used to properly add contacts to your subscriber table. So, if your domain is an IP address, newly added ppl will have an IP instead of domain for their host part of URI. Try also to specify alias as you IP address, i.e. listen="202.x.x.x" alias="202.x.x.x" You may also try to add a domain entry to the domain table with serctl (that is if you use functions in your config such as "is_uri_host_local()"), i.e. serctl domain add 202.x.x.x Hope it will work for you. Andrey. On 9/20/06, mukesh <mukesh(a)hungama.com> wrote: > Hi friends, > I have SER (with database support) running on my linux box. But I do > not have a domain name registered for the IP of this machine but it does > have a public IP (202.87.xx.xxx) address which allows me access it from > outside. > > If I try to access (using sipsak client) a uri from local system it > fails. > example URIs I am trying are ... > sip:mukesh@<public ip> > sip:mukesh@<public ip>:5060 > > ...with no result :( > > I referred http://mit.edu/sip/sip.edu/ser.shtml to configure my > ser.cfg. > > So the question is How can I connect to SER using IP and do I need > to use "# export SIP_DOMAIN=<public ip>"? > > Regards > Mukesh

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RE: [Serusers] SIPP with SER
by Verbeiren, David 22 Sep '06

22 Sep '06

> Hello , > > has anyone any experience on testing SER with SIPP ?? > > I've sent the following mail to sipp-users mailing list, I'm sending it > also here, in case anyone has any idea or experience. This may not be exactly what you want to test but I've done tests with SIPp and various SIP proxies, SER being one of them, by targeting the UAS itself on the SIPp command line but with a "remote sending address" parameter specifying the proxy to use. This works great and does not require the 'service' user to be known to SER as it simply looks at the hostname (or ip address) part of the URI to proxy the call. This gives the following modified command lines: UAC : ./sipp -sn uac 2.2.2.2 -r 1 -m 1 -d 5000 -i 1.1.1.1 -rsa 3.3.3.3 UAS : ./sipp -sn uas -rsa 3.3.3.3 The remote sending address is similar to the "outbound proxy" parameter found on many SIP phones. This way of running SIPp gives the call flow you describe but shortcuts most of the ser.cfg routing logic as uri!=myself. Regards, -David

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[Serusers] problem connecting two asterisk servers through SER
by Sameer Ahmed 22 Sep '06

22 Sep '06

Hi We have two asterisks registered as UA's. SER is acting as a proxy server. We want to make calls between two asterisk users through SER. One asterisk server has extensions starting with 2, and the other one has extensions starting with 4. We route the calls based on the first digit. We are unable to call from an extension on one asterisk server to the other asterisk server. "rewritehostport" doesn't seem to be doing the trick for us. Does anyone have a solution? Below is our ser.cfg file: =====================SER.CFG===================== # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines for debugging */ #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@192.168.1.60 /ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if(method == "INVITE") { if (uri =~ "sip:2[0-9]{3}@*"){ log(1, "Forwarding to PakVista BlueFlash\n"); route(2); break; } else if (uri =~ "sip:4[0-9]{3}@*"){ log(1, "Forwarding to MyRoom BlueFlash\n"); route(3); break; } } log(1, "After INVITE\n"); # 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(); log(1, "After REGISTER\n"); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { log(1,"Inside NOT uri==myself \n"); # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { log(1, "Inside uri==myself\n"); if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.1.60", "subscriber")) { www_challenge(" 192.168.1.60", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log(1, "location Not Found\n"); sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { log(1, "Inside route 1\n"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { log(1, "Inside route 2\n"); rewritehostport("192.168.1.61:5060"); if (!t_relay()) { log(1, "t_relay error\n"); sl_reply_error(); }; log(1, "Leaving route 2\n"); } route[3] { log(1, "Inside route 3\n"); rewritehostport("192.168.1.62"); if (!t_relay()) { log(1, "t_relay error\n"); sl_reply_error(); }; #forward(uri:host, uri:port); log(1, "Leaving route 3\n"); ===============================EOF========================

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[Serusers] some error messages
by jorain 22 Sep '06

22 Sep '06

Hi all, Did anyone have encountered these error codes? Error: str2i(AmUtils.cpp:149):str2i:unexpected char R in RTP/AVP 97409 Error: parse_sdp_media(AmSdp.cpp:549)while parsing 'm=audio' line Error: 400 session description parsing failed. pthread_detach failed with code EINVAL:thread already in detached state Error: reply(AmRequest.cpp:335):AmRequestUAS::reply:400 fifo_t_reply:to-ta expected Error: writeToBuffer(IsdnCapiConnection.cpp:544):Warning:dropping data, try increase AUDIO_BUFFER_LEN... Error: changeSIPState(IsdnCall.cpp:339):unhandled SIP state transition(inbound) from (DISCONNECTED|DISCONNECTED) to (CONNECTED|DISCONNECTED) Thanks Regards, jorain

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Re: [Users] serweb: DB.php is missing on my system
by Ferianto siregar 22 Sep '06

22 Sep '06

Dear Roland, I think that DB.php is missing, So, you need DB.php packet. You can download it from http://pear.php.net For the installation, you can type # pear <name of the packet> There is also many tutorials about it in that web. I do hope this can help you. :) Thank you Regards, Ferianto Message: 4 Date: Fri, 22 Sep 2006 07:43:16 +0800 From: Ronald Wiplinger Subject: [Users] serweb: DB.php is missing on my system To: Openser Message-ID: <45132394.8040309(a)elmit.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed [Fri Sep 22 06:54:42 2006] [error] [client 192.168.250.108] PHP Warning: require_once(DB.php) [href='function.require-once'>function.require-once]: failed to open stream: No such file or directory in /srv/www/serweb/html/main_prepend.php on line 38 [Fri Sep 22 06:54:42 2006] [error] [client 192.168.250.108] PHP Fatal error: require_once() [href='function.require'>function.require]: Failed opening required 'DB.php' (include_path='.:/usr/share/php5:/usr/share/php5/PEAR') in /srv/www/serweb/html/main_prepend.php on line 38 # locate DB.php /usr/share/php5/PEAR/PEAR/DependencyDB.php How can I get it? In the meantime I have tried to turn off loging in config.php: $config->enable_loging = false; $config->log_file = "/var/log/serweb"; but get still the same error. SuSE 10.1 installed pear5 (there is no 'pear') # pear5 list Installed packages, channel pear.php.net: ========================================= Package Version State Archive_Tar 1.3.1 stable Console_Getopt 1.2 stable PEAR 1.4.6 stable # pear5 list-all PHP Fatal error: Allowed memory size of 8388608 bytes exhausted (tried to allocate 8208 bytes) in /usr/share/php5/PEAR/PEAR/Registry.php on line 1048 What do I need to do? bye Ronald ------------------------------ _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 16, Issue 54 ************************************* --------------------------------- Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.

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[Users] TLS problem : May I ask you again..Please....
by Ferianto siregar 22 Sep '06

22 Sep '06

Dear all, First I would like to say thanks to every body in this forum. Thank you very much.. All, I am sorry for my message again. Sorry for the long message. I am in rush and panic. I have read the instruction in tls.htm about TLS configuration many times. But, Untill now, I can not build it. Maybe because of my lack understanding and skill. When I run "netstat -anp|grep 5061", I got this output message : [root@localhost root]# netstat -anp|grep 5061 tcp 0 0 202.95.149.251:5061 0.0.0.0:* LISTEN 5099/openser [root@localhost root]# netstat -anp|grep 5060 tcp 0 0 202.95.149.251:5060 0.0.0.0:* LISTEN 5099/openser udp 0 0 202.95.149.251:5060 0.0.0.0:* 5099/openser From the output, I can see that the port 5061 (for TLS) can be loaded. But, why when I run "ngrep -d any...", there is no port 5061? Here is the output U 202.95.149.3:5060 -> 202.95.149.251:5060 .. #### T 127.0.0.1:32804 -> 127.0.0.1:631 [AP] POST / HTTP/1.1.. ## T 127.0.0.1:32804 -> 127.0.0.1:631 [AP] Content-Length: 220.. ## T 127.0.0.1:32804 -> 127.0.0.1:631 [AP] Content-Type: application/ipp.. ## T 127.0.0.1:32804 -> 127.0.0.1:631 [AP] Host: localhost.. ## T 127.0.0.1:32804 -> 127.0.0.1:631 [AP] .. ## T 127.0.0.1:32804 -> 127.0.0.1:631 [AP] ........ ## T 127.0.0.1:631 -> 127.0.0.1:32804 [AP] HTTP/1.1 200 OK.. # T 127.0.0.1:631 -> 127.0.0.1:32804 [AP] Date: Fri, 22 Sep 2006 08:10:21 GMT.. ## T 127.0.0.1:631 -> 127.0.0.1:32804 [AP] Server: CUPS/1.1..Connection: Keep-Alive..Keep-Alive: timeout=60..Content-Language: C..Content-Type: application/ipp; ch arset=iso-8859-1..Content-Length: 81.............G..attributes-charset..utf-8H..attributes-natural-language..en-us.utf-8 . What should I do? Please help me..Please give me a suggestion. These are the tasks that I have done before : 1. I have added in DNS Records : pcr.ac.id IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.pcr.ac.id. pcr.ac.id IN NAPTR 60 40 "s" "SIP+D2T" "" _sip._tcp.pcr.ac.id. pcr.ac.id IN NAPTR 60 60 "s" "SIP+D2U" "" _sip._udp.pcr.ac.id 2. Here is my openser.cfg configuration : debug=9 fork=no log_stderror=yes listen=202.95.149.251 # put your openserver IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" fifo_mode=0666 alias="pcr.ac.id" tls_port_no=5061 # uncomment the following lines for TLS support disable_tls = 0 listen = tls:202.95.149.251:5061 #listen = udp:202.95.149.251:5060 tls_verify_client = on tls_require_client_certificate = on tls_verify_server=on 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" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "s:inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); return; }; if (uri==myself) { if (method=="ACK") { route(6); return; } else if (method=="CANCEL") { route(3); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "Uopenser Not Found"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } #if (!allow_trusted() && nat_uac_test("19")) { # setflag(6); #} lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); return; }; if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); return; }; sl_send_reply("404", "Uopenser Not Found"); return; }; if (method=="CANCEL") { route(1); return; } if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.254:5060"); # INOPENSERT YOUR PSTN GATEWAY IP ADDRESS # avp_write("i:45", "inv_timeout"); #if (!allow_trusted() && isflagset(6)) { if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } What should I add in openser.cfg? Please help..Please.. Thank you very much, Regards, Ferianto --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

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[Serusers] Call btn two SIP Servers -Thanks
by Sharon Abraham 22 Sep '06

22 Sep '06

Hi guys, Could final get the call thru both the SER poxies. Thanks for all the tips. Sharon Abraham Network Engineer Afila Tech India (p) Ltd www.afila.com

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[Serusers] How to configure multi-domain?
by Ricardo Carvalho 22 Sep '06

22 Sep '06

I've added the following lines to my ser.cfg file: ################### modparam("usrloc", "use_domain", 1) #If the domain part of the user should be also saved and used for identifing the user (along with the username part). Useful in multi domain scenarios. Non 0 value means true. modparam("registrar", "use_domain", 1) # If set to 1 then the registrar will use username@domain as address of record. If the variable is set to 0 then only username will be used as the address of record. Default value is 0. ################### I've added 2 diferent domains to the domain table, and got different phones registered with those different domains. Although, after editing that way ser.cfg file, my phones still register in Ser, but none of them can call any other, even if I call someone only by username or by username@domain. Am I in the right path to configure a multi-domain Ser server? What else should I do to make things work fine? Regards, Ricardo.

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Re: [Serusers] RTP being sent to wrong port
by Jiri Kuthan 22 Sep '06

22 Sep '06

The client should be then fixed or exchanged. It is not yet a BCP RFC or so, but it should be soon. In the meantime it is just an informal but important practice to build clients in a symmetric way. I'm actually aware of few that are not build this way. -jiri At 18:12 20/09/2006, Andrey Kuprianov wrote: >Just looked up the code to make sure... :) It's definitely assymetric! >Thanks Vamsi!!! > >On 9/20/06, Vamsi Pottangi <vamsipottangi(a)gmail.com> wrote: >> >>Symmetric client uses the same port to send and receive messages. >> >> >>On 9/20/06, Andrey Kuprianov <andrey.kouprianov(a)gmail.com > wrote: >>> Forgive my ignorance. I know the term "symmetric client", but i dont >>> really know what that means. Can u please explain or just give me some >>> doc/rfc to read? :) >>> >>> Andrey. >>> >>> On 9/20/06, Vamsi Pottangi < vamsipottangi(a)gmail.com> wrote: >>> > >>> > Is your application symmetric? >>> > >>> > >>> > On 9/20/06, Andrey Kuprianov < andrey.kouprianov(a)gmail.com> wrote: >>> > > >>> > Hi, >>> > >>> > Im facing this problem for a few days already. Hope someone might have >>> > an idea or two. It's a long email too, btw. >>> > >>> > Im testing my app behind NAT's with SER-0.9.6 + mediaproxy + Asterisk >>> > (conf server). While X-Lite works *fine* and I get 2 way audio always, >>> > my app seems to have some kind of weird bug and I end up with 1 way >>> > audio stream. >>> > >>> > Here's the thing. Application starts receiver and transmitter on >>> > completely different ports. Here's SDP for remote and local machines >>> > (public IP's been x'ed by myself, of course): >>> > >>> > This one's sent in INVITE message to SER ==> >>> > 69563 DEBUG media.MediaManager - Local SDP: (this one's from local >>machine) >>> > v=0 >>> > o=xps 1158739216750 1158739216757 IN IP4 192.168.1.3 >>> > s=MC >>> > c=IN IP4 192.168.1.3 >>> > t=0 0 >>> > m=audio 25000 RTP/AVP 0 8 >>> > a=rtpmap:0 PCMU/8000 >>> > a=rtpmap:8 PCMA/8000 >>> > a=sendrecv >>> > >>> > This one's sent with 200 OK from SER ==> >>> > 69563 DEBUG media.MediaManager - Remote SDP: >>> > v=0 >>> > o=root 2500 2500 IN IP4 203.159.x.x >>> > s=session >>> > c=IN IP4 203.159.x.x >>> > t=0 0 >>> > m=audio 30004 RTP/AVP 0 8 >>> > a=rtpmap:0 PCMU/8000 >>> > a=rtpmap:8 PCMA/8000 >>> > a=silenceSupp:off - - - - >>> > >>> > The further log shows that transmitters and receivers have started on >>> > the proper IP/port ==> >>> > >>> > 69594 DEBUG media.AVReceiver - Start listening for RTP @ addr: >>> > 192.168.1.3 port: 25000 ttl: 1 >>> > 69657 DEBUG media.AVTransmitter - Created transmitter for: >>> > [203.159.x.x] at ports: [30004] encoded as: [[ULAW/rtp, alaw]] >>> > 69657 DEBUG media.MediaManager - Starting transmission >>> > 69672 DEBUG media.AVTransmitter - Track 0 is set to transmit as: >>> > ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits >>> > 69860 INFO media.AVTransmitter - Binded to port 30004 >>> > 69875 DEBUG media.AVTransmitter - Started transmitting track 0 encoded >>> > as ULAW/rtp @ [203.159.x.x]:30004 >>> > >>> > Nevertheless, the Ethereal capture shows that mediaproxy sends the >>> > stream NOT to port 25000 (local port), but to port 30004 (remote >>> > port)!! Why? This is not the case with X-Lite, however. Xlite captures >>> > show that streams are sent to proper ports always. Btw, Im attaching >>> > SIP+RTP Ethereal captures with this mail. Please, take a look. >>> > >>> > I also want to mention, that this doesnt happen with my app. for all >>> > the NATed nets. In some nets it works fine, but not in this one. Just >>> > weird. >>> > >>> > Please, let me know if anyone has ideas or hints regarding this nonsence >>:) >>> > >>> > Bests, >>> > >>> > Andrey. >>> > >>> > >>> > _______________________________________________ >>> > Serusers mailing list >>> > Serusers(a)lists.iptel.org >>> > http://lists.iptel.org/mailman/listinfo/serusers >>> > >>> > >>> > >>> > >>> > >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >> >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 1

[Serusers] List of Routing functions/commands
by Sameer Ahmed 22 Sep '06

22 Sep '06

Hi Where can I get a list (with description) of all the routing functions/commands that can be used in ser.cfg? Thank you. Regards, Sameer

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[Serusers] Scaling SER
by Jon Farmer 22 Sep '06

22 Sep '06

Hi I currently have a single SER server using MySQL for data storage. Due to the number of users I now have and to help redundancy I want to setup another SER server. I have search the web and have not been able to find a tutorial on this so I hope someone on this list can help. I need the SER servers to at least know about the location of clients registered with each server. Therefore can I point both servers at the same MySQL and for them to share the data? Are there any issues around this strategy? Is there another way to accomplish what I want? Regards Jon -- Jon Farmer Telford, Shropshire, UK

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[Serusers] STUN characteristics
by sip 22 Sep '06

22 Sep '06

I'm doing an experiment with our userbase at the moment, making some modifications to allow more users to register than who'd been allowed to register before (to minimise the amount of bandwidth we need to use for our services, we are trying to absolutely minimise the number of users that require the use of an RTPproxy, since, in a userbase of 100,000 users, this can be rather pricey if there are too many of them which require it). We used to not allow anyone with a contact header that was NOT a public IP address to register onto the system, with the assumption that they could use STUN and it SHOULD rewrite the contact header. This assumption has proven to be... optimistic. Several of the UAs we see show radically different characteristics during this 'open' trial, and I'm not sure if they're just differing STUN versions or if our users simply can't follow the install/configure instructions. SO... my question becomes: what are the characterstics of functioning STUN implementations and how they should overwrite contact/via/sdp headers for the UA's messages? Most every one of the STUN implementations I've seen rewrites the Contact header and the data inside the O and C lines in the SDP header. Most of them seem to overwrite the Via headers with valid information as well, although not all of them do. What are some other things I should be looking for? I'm trying to essentially test for a particular variant of STUN, allowing users who've at least followed the instructions to the letter to have working and usable clients (perhaps using RTP proxy where needed) while not allowing users who're still attempting to use private IPs for everything to waste our bandwidth because they're not following the instructions or they're using a UA that is, in effect, broken in its implementation. This is a lot more complex than it sounds, I know, but I think if I have more data to work with, I might be able to come up with a more acceptable scenario than we currently employ. N.

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[Serusers] SIPP with SER
by Kostas Marneris 21 Sep '06

21 Sep '06

Hello , has anyone any experience on testing SER with SIPP ?? I've sent the following mail to sipp-users mailing list, I'm sending it also here, in case anyone has any idea or experience. ---------------------------------------------------------------------------------------------------- Hello all, I run the basic UAC senario in Host A (1.1.1.1) and the basic UAS senario in Host B (2.2.2.2). I have a SER SIP Proxy in the middle (3.3.3.3). UAC : ./sipp -sn uac -r 1 -m 1 -d 5000 -i 1.1.1.1 3.3.3.3 UAS : ./sipp -sn uas SER.cfg : I use the following hack in ser.cfg, because the default user 'service' is not actually registered in Location DB. # In INVITE handler .. ... # --- SIPP hack --- if (uri =~ "^sip:service@3.3.3.3:5060") { rewritehostport("2.2.2.2:5060"); route(1); break; }; # ----------------- # In MAIN Route Block # --------------------- Nothing changed from 'default SER behaviour' : if ((method != "REGISTER") && (method != "OPTIONS")) { log(1, " Add RR"); record_route(); }; if (loose_route()) { route(1); break; }; Where is the problem : It seems that SER doesn't recognize (in the dialog) the ACK sent by the UAC (Host A), do does not relay it to the UAS. The UAS continues to send '200 OK'... so I can't make it to have a 'full call setup simulation senario'. Call Flow : HOST A (UAC) SER HOST B (UAS) -----------> (1) INVITE <----------- (2) 100 Trying --------------> (3) INVITE <------------- (4) 180 Ringing <----------- (5) 180 Ringing <------------- (6) 200 OK <----------- (7) 200 OK -----------> (8) ACK <------------- (9) 200 OK -----------> (10) ACK .... .... The first ACK (8) from UAC to SER doesn't relayed to UAS, so UAS keeps sending '200 OK'... This ACK doesn't go through the 'loose_route()' block because SER thinks that this ACK is not in the dialog. Even if I add the following code in MAIN route block, SER insists to absorb this ACK request. ... } else if (method == "ACK") { # ACK mesgs --> route[1] log(1," * ACK -- not in the DIALOG --"); route(1); break; } else if ... Do you find something wrong in SER conf ? or do you have any idea why this happening ? ---------------------------------------------------------------------------------------------------- thanks in advance for any hint. Kostas

2 1

[Users] redundancy
by Ramin Dousti 21 Sep '06

21 Sep '06

Hi, A SIP INVITE to sip.mydoamin.com with sip.mydomain.com having two A-RR's, must be forwarded to the first IP and if that IP is unreachable then to the second IP. How can I achieve this? My first problem is that I seem to not have the list of the two IP's available to me via a configuration key-word. So I need to hardcode the IP's in my conf file. My second problem is I don't know how to iterate over these two IP's. Maybe I can use t_on_failure() and have the second IP processed in the failure_route[]. But then again, how can I distinguish between a timeout and a legitimate failure? Thank you in advance, -- Ramin

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[Users] AcctSessionTime (table radacct)
by Alberto Pacifico 21 Sep '06

21 Sep '06

Hi all, I'd like to get AcctSessionTime (db radius /table radacct) with setflag(). I´ve already got the AcctStartTime and AcctStopTime. The values of AcctSessionTime that I´ve got are zero in the calls. I know that I can calculate the duration of the call using the values of AcctStartTime and AcctStopTime, but I'm using an software (CDRTool) that uses the values of AcctSessionTime to calculate the duration of the call. How can I get the value of AcctSessionTime (diferent of zero)? Thanks Alberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] Q:ABOUT:Use the source IP address
by Alejandro Sanchez 21 Sep '06

21 Sep '06

Hello. I need tell to OPENSER that use the source IP address of the NAT/Firewall as the return address instead of the contact information in the SIP message. Someone knos How? Thank's. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] Using IP address as domain
by mukesh 21 Sep '06

21 Sep '06

Hi friends, I have SER (with database support) running on my linux box. But I do not have a domain name registered for the IP of this machine but it does have a public IP (202.87.xx.xxx) address which allows me access it from outside. If I try to access (using sipsak client) a uri from local system it fails. example URIs I am trying are ... sip:mukesh@<public ip> sip:mukesh@<public ip>:5060 ...with no result :( I referred http://mit.edu/sip/sip.edu/ser.shtml to configure my ser.cfg. So the question is How can I connect to SER using IP and do I need to use "# export SIP_DOMAIN=<public ip>"? Regards Mukesh

2 1

[Users] Configurable nathelper nortpproxy string
by Jeremie Le Hen 21 Sep '06

21 Sep '06

Hi list, I have made a small patch to the nathelper module, that allows to configure the "a=nortpproxy:yes\r\n" string. Basically, as noticed in user documentation, you can add this to your configuration file, once the patch is applied and OpenSER recompiled : % modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes") I needed to do this because I have the following setup: SoftPhone1_ \____ ____NATing Proxy_______Registrar Proxy_______(outside) _/ SoftPhone2 For some reason, the registrar also uses nathelper/rtpproxy. I first used the "f" flag for force_rtp_proxy() in my NATing proxy, in order to circumvent the nortpproxy flag that the registrar proxy has set. This worked perfectly except if softphone1 tries to call softphone2. In this case my NATing proxy would replace the SDP twice. With my patch, I simply turn the nortpproxy string to something else that only my NATing proxy cares about. Please, commit it if you find it worth enough. Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 2

[Serusers] Call btn two SIP Servers
by Sharon Abraham 21 Sep '06

21 Sep '06

The prob is pretty basic. I have installed 2 SERs and have a set of users registered to either of the proxies. How can i place a call from user1 at SER1 to user2 @ SER2? This is what i've done 1. In my "domain" table (i.e. MySQL) in both SER's I have added domain entries. 2. Then into" trusted" tables of either proxies , i've added the value of the other proxy 3. Then made the entries in the ser.cfg. But when i restart ser i get this as the error [root@localhost sbin]# service ser restart Stopping ser: [ OK ] Starting ser: ERROR: bad config file (3 errors) [FAILED] My ser.cfg of one of the Proxies is as follows... can anyone help??? pls? # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/xlog.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; if (method=="INVITE") { if( !allow_trusted() && !is_uri_host_local() ) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }

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