- sr-users - kamailio.org

[Users] openser TLS error problem
by Lindsey Hans Joseph Lao 05 Oct '06

05 Oct '06

Hello, I have an Openser server with TLS support running ok. However, when a UA client tries to register with it, it gives off this error: 7(63031) tls_accept: Error in SSL: 7(63031) tls_error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number I ran SSLDUMP and got these logs: 8 1 2.4157 (2.4157) C>S SSLv2 compatible client hello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_3DES_EDE_CBC_SHA SSL2_CK_RC4 SSL2_CK_3DES SSL2_CK_RC2 TLS_RSA_WITH_DES_CBC_SHA SSL2_CK_DES TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_RSA_EXPORT_WITH_RC4_40_MD5 TLS_RSA_EXPORT_WITH_RC2_CBC_40_MD5 SSL2_CK_RC4_EXPORT40 SSL2_CK_RC2_EXPORT40 TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA TLS_DHE_DSS_WITH_DES_CBC_SHA TLS_DHE_DSS_EXPORT1024_WITH_DES_CBC_SHA 8 2.4163 (0.0005) S>C TCP FIN 8 2.4165 (0.0002) C>S TCP FIN Any idea of what might the problem be? Help would be greatly appreciated. Thanks!

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[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

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[Users] OpenSER Summit @ VoN Europe
by Daniel-Constantin Mierla 05 Oct '06

05 Oct '06

Hello Everybody, we are pleased to announce that first OpenSER Summit will take place in Berlin, on the 8th November. The summit is hosted inside the VoN Europe Exhibition. The participants registered via OpenSER will have free access to summit and exhibition. A registration form will be available very soon on the web site, as an alternative you can send email to team(a)openser.org with your details. We are looking to network people around OpenSER, to share experiences and build business relations, among invited persons are executives from major communication companies. The board members and main developers will be there as well. There is no need for other words about VoN (http://www.von.com), is the leading conference in VoIP and we are glad that we can tie up together the OpenSER Summit. To be able to cover organizing expenses, we are looking for sponsorships. The sponsors will get the opportunity to held a presentation and do advertising inside the event. Banners will be published on the OpenSER web site and event's page. We will be able to offer to sponsors free passes to full conference (all presentations at VoN, lunches, dinners and parties) and other discounts. If you are interested to be a sponsor of the event, please contact us at team(a)openser.org and we will give you more details about the conditions and offering you will get. More details will be published on http://www.openser.org site as soon as updates come. Looking forward to meet you there, Daniel

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[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.

2 1

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

4 3

[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.

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[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

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[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

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[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

2 1

[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.

1 0

[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

2 2

[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

3 6

[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

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[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

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[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!

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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? >

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[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

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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 >> >> > >> >> >> >> >> >>

1 0

[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

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[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

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